Computer Networking A Top-Down Approach 6th Edition
James F. Kurose
University of Massachusetts, Amherst
Keith W. Ross
Polytechnic Institute of NYU
COMPUTER
NETWORKING
A Top-Down Approach
SIXTH EDITION
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Library of Congress Cataloging-in-Publication Data
Kurose, James F.
Computer networking : a top-down approach / James F. Kurose, Keith W. Ross.6th ed.
p. cm.
Includes bibliographical references and index.
ISBN-13: 978-0-13-285620-1
ISBN-10: 0-13-285620-4
1. Internet. 2. Computer networks. I. Ross, Keith W., 1956- II. Title.
TK5105.875.I57K88 2012
004.6dc23
2011048215
10987654321
ISBN-13: 978-0-13-285620-1
ISBN-10: 0-13-285620-4
This book was composed in Quark. Basal font is Times. Display font is Berkeley.
iii
About the Authors
Jim Kurose
Jim Kurose is a Distinguished University Professor of Computer Science at the
University of Massachusetts, Amherst.
Dr. Kurose has received a number of recognitions for his educational
activities including Outstanding Teacher Awards from the National
Technological University (eight times), the University of Massachusetts, and
the Northeast Association of Graduate Schools. He received the IEEE Taylor
Booth Education Medal and was recognized for his leadership of
Massachusetts Commonwealth Information Technology Initiative. He has
been the recipient of a GE Fellowship, an IBM Faculty Development Award,
and a Lilly Teaching Fellowship.
Dr. Kurose is a former Editor-in-Chief of IEEE Transactions on
Communications and of IEEE/ACM Transactions on Networking. He has
been active in the program committees for IEEE Infocom, ACM SIGCOMM,
ACM Internet Measurement Conference, and ACM SIGMETRICS for a
number of years and has served as Technical Program Co-Chair for those
conferences. He is a Fellow of the IEEE and the ACM. His research interests
include network protocols and architecture, network measurement, sensor
networks, multimedia communication, and modeling and performance
evaluation. He holds a PhD in Computer Science from Columbia University.
Keith Ross
Keith Ross is the Leonard J. Shustek Chair Professor and Head of the Computer
Science Department at Polytechnic Institute of NYU. Before joining NYU-Poly in
2003, he was a professor at the University of Pennsylvania (13 years) and a
professor at Eurecom Institute (5 years). He received a B.S.E.E from Tufts
University, a M.S.E.E. from Columbia University, and a Ph.D. in Computer and
Control Engineering from The University of Michigan. Keith Ross is also the
founder and original CEO of Wimba, which develops online multimedia
applications for e-learning and was acquired by Blackboard in 2010.
Professor Rosss research interests are in security and privacy, social networks,
peer-to-peer networking, Internet measurement, video streaming, content distribution
networks, and stochastic modeling. He is an IEEE Fellow, recipient of the Infocom
2009 Best Paper Award, and recipient of 2011 and 2008 Best Paper Awards
for Multimedia Communications (awarded by IEEE Communications Society). He
has served on numerous journal editorial boards and conference program commit-
tees, including IEEE/ACM Transactions on Networking, ACM SIGCOMM, ACM
CoNext, and ACM Internet Measurement Conference. He also has served as an
advisor to the Federal Trade Commission on P2P file sharing.
This page intentionally left blank
To Julie and our three precious
onesChris, Charlie, and Nina
JFK
A big THANKS to my professors, colleagues,
and students all over the world.
KWR
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Preface
Welcome to the sixth edition of Computer Networking: A Top-Down Approach. Since
the publication of the first edition 12 years ago, our book has been adopted for use at
many hundreds of colleges and universities, translated into 14 languages, and used
by over one hundred thousand students and practitioners worldwide. Weve heard
from many of these readers and have been overwhelmed by the positive response.
Whats New in the Sixth Edition?
We think one important reason for this success has been that our book continues to offer
a fresh and timely approach to computer networking instruction. Weve made changes
in this sixth edition, but weve also kept unchanged what we believe (and the instruc-
tors and students who have used our book have confirmed) to be the most important
aspects of this book: its top-down approach, its focus on the Internet and a modern
treatment of computer networking, its attention to both principles and practice, and its
accessible style and approach toward learning about computer networking. Neverthe-
less, the sixth edition has been revised and updated substantially:
The Companion Web site has been significantly expanded and enriched to
include VideoNotes and interactive exercises, as discussed later in this Preface.
In Chapter 1, the treatment of access networks has been modernized, and the
description of the Internet ISP ecosystem has been substantially revised, account-
ing for the recent emergence of content provider networks, such as Googles. The
presentation of packet switching and circuit switching has also been reorganized,
providing a more topical rather than historical orientation.
In Chapter 2, Python has replaced Java for the presentation of socket program-
ming. While still explicitly exposing the key ideas behind the socket API, Python
code is easier to understand for the novice programmer. Moreover, unlike Java,
Python provides access to raw sockets, enabling students to build a larger variety
of network applications. Java-based socket programming labs have been
replaced with corresponding Python labs, and a new Python-based ICMP Ping
lab has been added. As always, when material is retired from the book, such as
Java-based socket programming material, it remains available on the books
Companion Web site (see following text).
In Chapter 3, the presentation of one of the reliable data transfer protocols has
been simplified and a new sidebar on TCP splitting, commonly used to optimize
the performance of cloud services, has been added.
In Chapter 4, the section on router architectures has been significantly updated,
reflecting recent developments and practices in the field. Several new integrative
sidebars involving DNS, BGP, and OSPF are included.
Chapter 5 has been reorganized and streamlined, accounting for the ubiquity of
switched Ethernet in local area networks and the consequent increased use of
Ethernet in point-to-point scenarios. Also, a new section on data center network-
ing has been added.
Chapter 6 has been updated to reflect recent advances in wireless networks, par-
ticularly cellular data networks and 4G services and architecture.
Chapter 7, which focuses on multimedia networking, has gone through a major
revision. The chapter now includes an in-depth discussion of streaming video,
including adaptive streaming, and an entirely new and modernized discussion of
CDNs. A newly added section describes the Netflix, YouTube, and Kankan video
streaming systems. The material that has been removed to make way for these
new topics is still available on the Companion Web site.
Chapter 8 now contains an expanded discussion on endpoint authentication.
Significant new material involving end-of-chapter problems has been added. As
with all previous editions, homework problems have been revised, added, and
removed.
Audience
This textbook is for a first course on computer networking. It can be used in both
computer science and electrical engineering departments. In terms of programming
languages, the book assumes only that the student has experience with C, C++, Java,
or Python (and even then only in a few places). Although this book is more precise
and analytical than many other introductory computer networking texts, it rarely
uses any mathematical concepts that are not taught in high school. We have made a
deliberate effort to avoid using any advanced calculus, probability, or stochastic
process concepts (although weve included some homework problems for students
with this advanced background). The book is therefore appropriate for undergradu-
ate courses and for first-year graduate courses. It should also be useful to practition-
ers in the telecommunications industry.
What Is Unique about This Textbook?
The subject of computer networking is enormously complex, involving many
concepts, protocols, and technologies that are woven together in an intricate
manner. To cope with this scope and complexity, many computer networking texts
are often organized around the layers of a network architecture. With a layered
organization, students can see through the complexity of computer networking
they learn about the distinct concepts and protocols in one part of the architecture
while seeing the big picture of how all parts fit together. From a pedagogical
perspective, our personal experience has been that such a layered approach
viii Preface
Preface ix
indeed works well. Nevertheless, we have found that the traditional approach of
teachingbottom up; that is, from the physical layer towards the application
layeris not the best approach for a modern course on computer networking.
A Top-Down Approach
Our book broke new ground 12 years ago by treating networking in a top-down
mannerthat is, by beginning at the application layer and working its way down
toward the physical layer. The feedback we received from teachers and students
alike have confirmed that this top-down approach has many advantages and does
indeed work well pedagogically. First, it places emphasis on the application layer
(a high growth area in networking). Indeed, many of the recent revolutions in
computer networkingincluding the Web, peer-to-peer file sharing, and media
streaminghave taken place at the application layer. An early emphasis on application-
layer issues differs from the approaches taken in most other texts, which have only a
small amount of material on network applications, their requirements, application-layer
paradigms (e.g., client-server and peer-to-peer), and application programming inter-
faces. Second, our experience as instructors (and that of many instructors who have
used this text) has been that teaching networking applications near the beginning of
the course is a powerful motivational tool. Students are thrilled to learn about how
networking applications workapplications such as e-mail and the Web, which most
students use on a daily basis. Once a student understands the applications, the student
can then understand the network services needed to support these applications. The
student can then, in turn, examine the various ways in which such services might be
provided and implemented in the lower layers. Covering applications early thus pro-
vides motivation for the remainder of the text.
Third, a top-down approach enables instructors to introduce network appli-
cation development at an early stage. Students not only see how popular applica-
tions and protocols work, but also learn how easy it is to create their own
network applications and application-level protocols. With the top-down
approach, students get early exposure to the notions of socket programming, serv-
ice models, and protocolsimportant concepts that resurface in all subsequent
layers. By providing socket programming examples in Python, we highlight the
central ideas without confusing students with complex code. Undergraduates in
electrical engineering and computer science should not have difficulty following
the Python code.
An Internet Focus
Although we dropped the phrase Featuring the Internet from the title of this book
with the fourth edition, this doesnt mean that we dropped our focus on the Internet!
Indeed, nothing could be further from the case! Instead, since the Internet has
become so pervasive, we felt that any networking textbook must have a significant
focus on the Internet, and thus this phrase was somewhat unnecessary. We continue
to use the Internets architecture and protocols as primary vehicles for studying fun-
damental computer networking concepts. Of course, we also include concepts and
protocols from other network architectures. But the spotlight is clearly on the Inter-
net, a fact reflected in our organizing the book around the Internets five-layer archi-
tecture: the application, transport, network, link, and physical layers.
Another benefit of spotlighting the Internet is that most computer science and
electrical engineering students are eager to learn about the Internet and its protocols.
They know that the Internet has been a revolutionary and disruptive technology and
can see that it is profoundly changing our world. Given the enormous relevance of
the Internet, students are naturally curious about what is under the hood. Thus, it
is easy for an instructor to get students excited about basic principles when using the
Internet as the guiding focus.
Teaching Networking Principles
Two of the unique features of the bookits top-down approach and its focus on the
Internethave appeared in the titles of our book. If we could have squeezed a third
phrase into the subtitle, it would have contained the word principles. The field of
networking is now mature enough that a number of fundamentally important issues
can be identified. For example, in the transport layer, the fundamental issues include
reliable communication over an unreliable network layer, connection establishment/
teardown and handshaking, congestion and flow control, and multiplexing. Two fun-
damentally important network-layer issues are determining good paths between
two routers and interconnecting a large number of heterogeneous networks. In the
link layer, a fundamental problem is sharing a multiple access channel. In network
security, techniques for providing confidentiality, authentication, and message
integrity are all based on cryptographic fundamentals. This text identifies fundamen-
tal networking issues and studies approaches towards addressing these issues. The
student learning these principles will gain knowledge with a long shelf lifelong
after todays network standards and protocols have become obsolete, the principles
they embody will remain important and relevant. We believe that the combination of
using the Internet to get the students foot in the door and then emphasizing funda-
mental issues and solution approaches will allow the student to quickly understand
just about any networking technology.
The Web Site
Each new copy of this textbook includes six months of access to a Companion Web site
for all book readers at http://www.pearsonhighered.com/kurose-ross, which includes:
Interactive learning material. An important new component of the sixth edition
is the significantly expanded online and interactive learning material. The
books Companion Web site now contains VideoNotesvideo presentations of
x Preface
http://www.pearsonhighered.com/kurose-ross
Preface xi
important topics thoughout the book done by the authors, as well as walk-
throughs of solutions to problems similar to those at the end of the chapter.
Weve also added Interactive Exercises that can create (and present solutions
for) problems similar to selected end-of-chapter problems. Since students can
generate (and view solutions for) an unlimited number of similar problem
instances, they can work until the material is truly mastered. Weve seeded the
Web site with VideoNotes and online problems for chapters 1 through 5 and will
continue to actively add and update this material over time. As in earlier edi-
tions, the Web site contains the interactive Java applets that animate many key
networking concepts. The site also has interactive quizzes that permit students
to check their basic understanding of the subject matter. Professors can integrate
these interactive features into their lectures or use them as mini labs.
Additional technical material. As we have added new material in each edition of
our book, weve had to remove coverage of some existing topics to keep the
book at manageable length. For example, to make room for the new material in
this edition, weve removed material on ATM networks and the RTSP protocol
for multimedia. Material that appeared in earlier editions of the text is still of
interest, and can be found on the books Web site.
Programming assignments. The Web site also provides a number of detailed
programming assignments, which include building a multithreaded Web
server, building an e-mail client with a GUI interface, programming the sender
and receiver sides of a reliable data transport protocol, programming a distrib-
uted routing algorithm, and more.
Wireshark labs. Ones understanding of network protocols can be greatly deep-
ened by seeing them in action. The Web site provides numerous Wireshark
assignments that enable students to actually observe the sequence of messages
exchanged between two protocol entities. The Web site includes separate Wire-
shark labs on HTTP, DNS, TCP, UDP, IP, ICMP, Ethernet, ARP, WiFi, SSL, and
on tracing all protocols involved in satisfying a request to fetch a web page.
Well continue to add new labs over time.
Pedagogical Features
We have each been teaching computer networking for more than 20 years.
Together, we bring more than 50 years of teaching experience to this text, during
which time we have taught many thousands of students. We have also been active
researchers in computer networking during this time. (In fact, Jim and Keith first
met each other as masters students in a computer networking course taught by
Mischa Schwartz in 1979 at Columbia University.) We think all this gives us a
good perspective on where networking has been and where it is likely to go in the
future. Nevertheless, we have resisted temptations to bias the material in this book
towards our own pet research projects. We figure you can visit our personal Web
sites if you are interested in our research. Thus, this book is about modern com-
puter networkingit is about contemporary protocols and technologies as well as
the underlying principles behind these protocols and technologies. We also believe
that learning (and teaching!) about networking can be fun. A sense of humor, use
of analogies, and real-world examples in this book will hopefully make this mate-
rial more fun.
Supplements for Instructors
We provide a complete supplements package to aid instructors in teaching this course.
This material can be accessed from Pearsons Instructor Resource Center
(http://www.pearsonhighered.com/irc). Visit the Instructor Resource Center or send
e-mail to [email protected] for information about accessing these instructors
supplements.
PowerPoint slides. We provide PowerPoint slides for all nine chapters. The
slides have been completely updated with this sixth edition. The slides cover
each chapter in detail. They use graphics and animations (rather than relying
only on monotonous text bullets) to make the slides interesting and visually
appealing. We provide the original PowerPoint slides so you can customize them
to best suit your own teaching needs. Some of these slides have been contributed
by other instructors who have taught from our book.
Homework solutions. We provide a solutions manual for the homework problems
in the text, programming assignments, and Wireshark labs. As noted earlier, weve
introduced many new homework problems in the first five chapters of the book.
Chapter Dependencies
The first chapter of this text presents a self-contained overview of computer net-
working. Introducing many key concepts and terminology, this chapter sets the stage
for the rest of the book. All of the other chapters directly depend on this first chap-
ter. After completing Chapter 1, we recommend instructors cover Chapters 2
through 5 in sequence, following our top-down philosophy. Each of these five chap-
ters leverages material from the preceding chapters. After completing the first five
chapters, the instructor has quite a bit of flexibility. There are no interdependencies
among the last four chapters, so they can be taught in any order. However, each of
the last four chapters depends on the material in the first five chapters. Many
instructors first teach the first five chapters and then teach one of the last four chap-
ters for dessert.
xii Preface
http://www.pearsonhighered.com/irc
Preface xiii
One Final Note: Wed Love to Hear from You
We encourage students and instructors to e-mail us with any comments they might
have about our book. Its been wonderful for us to hear from so many instructors
and students from around the world about our first four editions. Weve incorporated
many of these suggestions into later editions of the book. We also encourage instructors
to send us new homework problems (and solutions) that would complement the
current homework problems. Well post these on the instructor-only portion of the
Web site. We also encourage instructors and students to create new Java applets that
illustrate the concepts and protocols in this book. If you have an applet that you
think would be appropriate for this text, please submit it to us. If the applet (including
notation and terminology) is appropriate, well be happy to include it on the texts
Web site, with an appropriate reference to the applets authors.
So, as the saying goes, Keep those cards and letters coming! Seriously,
please do continue to send us interesting URLs, point out typos, disagree with
any of our claims, and tell us what works and what doesnt work.Tell us what
you think should or shouldnt be included in the next edition. Send your e-mail
to [email protected] and [email protected].
Acknowledgments
Since we began writing this book in 1996, many people have given us invaluable
help and have been influential in shaping our thoughts on how to best organize and
teach a networking course. We want to say A BIG THANKS to everyone who has
helped us from the earliest first drafts of this book, up to this fifth edition. We are also
very thankful to the many hundreds of readers from around the worldstudents, fac-
ulty, practitionerswho have sent us thoughts and comments on earlier editions of
the book and suggestions for future editions of the book. Special thanks go out to:
Al Aho (Columbia University)
Hisham Al-Mubaid (University of Houston-Clear Lake)
Pratima Akkunoor (Arizona State University)
Paul Amer (University of Delaware)
Shamiul Azom (Arizona State University)
Lichun Bao (University of California at Irvine)
Paul Barford (University of Wisconsin)
Bobby Bhattacharjee (University of Maryland)
Steven Bellovin (Columbia University)
Pravin Bhagwat (Wibhu)
Supratik Bhattacharyya (previously at Sprint)
Ernst Biersack (Eurcom Institute)
Shahid Bokhari (University of Engineering & Technology, Lahore)
Jean Bolot (Technicolor Research)
Daniel Brushteyn (former University of Pennsylvania student)
Ken Calvert (University of Kentucky)
Evandro Cantu (Federal University of Santa Catarina)
Jeff Case (SNMP Research International)
Jeff Chaltas (Sprint)
Vinton Cerf (Google)
Byung Kyu Choi (Michigan Technological University)
Bram Cohen (BitTorrent, Inc.)
Constantine Coutras (Pace University)
John Daigle (University of Mississippi)
Edmundo A. de Souza e Silva (Federal University of Rio de Janeiro)
Philippe Decuetos (Eurcom Institute)
Christophe Diot (Technicolor Research)
Prithula Dhunghel (Akamai)
Deborah Estrin (University of California, Los Angeles)
Michalis Faloutsos (University of California at Riverside)
Wu-chi Feng (Oregon Graduate Institute)
Sally Floyd (ICIR, University of California at Berkeley)
Paul Francis (Max Planck Institute)
Lixin Gao (University of Massachusetts)
JJ Garcia-Luna-Aceves (University of California at Santa Cruz)
Mario Gerla (University of California at Los Angeles)
David Goodman (NYU-Poly)
Yang Guo (Alcatel/Lucent Bell Labs)
Tim Griffin (Cambridge University)
Max Hailperin (Gustavus Adolphus College)
Bruce Harvey (Florida A&M University, Florida State University)
Carl Hauser (Washington State University)
Rachelle Heller (George Washington University)
Phillipp Hoschka (INRIA/W3C)
Wen Hsin (Park University)
Albert Huang (former University of Pennsylvania student)
Cheng Huang (Microsoft Research)
Esther A. Hughes (Virginia Commonwealth University)
Van Jacobson (Xerox PARC)
Pinak Jain (former NYU-Poly student)
Jobin James (University of California at Riverside)
Sugih Jamin (University of Michigan)
Shivkumar Kalyanaraman (IBM Research, India)
Jussi Kangasharju (University of Helsinki)
Sneha Kasera (University of Utah)
Parviz Kermani (formerly of IBM Research)
xiv Preface
Preface xv
Hyojin Kim (former University of Pennsylvania student)
Leonard Kleinrock (University of California at Los Angeles)
David Kotz (Dartmouth College)
Beshan Kulapala (Arizona State University)
Rakesh Kumar (Bloomberg)
Miguel A. Labrador (University of South Florida)
Simon Lam (University of Texas)
Steve Lai (Ohio State University)
Tom LaPorta (Penn State University)
Tim-Berners Lee (World Wide Web Consortium)
Arnaud Legout (INRIA)
Lee Leitner (Drexel University)
Brian Levine (University of Massachusetts)
Chunchun Li (former NYU-Poly student)
Yong Liu (NYU-Poly)
William Liang (former University of Pennsylvania student)
Willis Marti (Texas A&M University)
Nick McKeown (Stanford University)
Josh McKinzie (Park University)
Deep Medhi (University of Missouri, Kansas City)
Bob Metcalfe (International Data Group)
Sue Moon (KAIST)
Jenni Moyer (Comcast)
Erich Nahum (IBM Research)
Christos Papadopoulos (Colorado Sate University)
Craig Partridge (BBN Technologies)
Radia Perlman (Intel)
Jitendra Padhye (Microsoft Research)
Vern Paxson (University of California at Berkeley)
Kevin Phillips (Sprint)
George Polyzos (Athens University of Economics and Business)
Sriram Rajagopalan (Arizona State University)
Ramachandran Ramjee (Microsoft Research)
Ken Reek (Rochester Institute of Technology)
Martin Reisslein (Arizona State University)
Jennifer Rexford (Princeton University)
Leon Reznik (Rochester Institute of Technology)
Pablo Rodrigez (Telefonica)
Sumit Roy (University of Washington)
Avi Rubin (Johns Hopkins University)
Dan Rubenstein (Columbia University)
Douglas Salane (John Jay College)
Despina Saparilla (Cisco Systems)
John Schanz (Comcast)
Henning Schulzrinne (Columbia University)
Mischa Schwartz (Columbia University)
Ardash Sethi (University of Delaware)
Harish Sethu (Drexel University)
K. Sam Shanmugan (University of Kansas)
Prashant Shenoy (University of Massachusetts)
Clay Shields (Georgetown University)
Subin Shrestra (University of Pennsylvania)
Bojie Shu (former NYU-Poly student)
Mihail L. Sichitiu (NC State University)
Peter Steenkiste (Carnegie Mellon University)
Tatsuya Suda (University of California at Irvine)
Kin Sun Tam (State University of New York at Albany)
Don Towsley (University of Massachusetts)
David Turner (California State University, San Bernardino)
Nitin Vaidya (University of Illinois)
Michele Weigle (Clemson University)
David Wetherall (University of Washington)
Ira Winston (University of Pennsylvania)
Di Wu (Sun Yat-sen University)
Shirley Wynn (NYU-Poly)
Raj Yavatkar (Intel)
Yechiam Yemini (Columbia University)
Ming Yu (State University of New York at Binghamton)
Ellen Zegura (Georgia Institute of Technology)
Honggang Zhang (Suffolk University)
Hui Zhang (Carnegie Mellon University)
Lixia Zhang (University of California at Los Angeles)
Meng Zhang (former NYU-Poly student)
Shuchun Zhang (former University of Pennsylvania student)
Xiaodong Zhang (Ohio State University)
ZhiLi Zhang (University of Minnesota)
Phil Zimmermann (independent consultant)
Cliff C. Zou (University of Central Florida)
We also want to thank the entire Addison-Wesley teamin particular, Michael Hirsch,
Marilyn Lloyd, and Emma Sniderwho have done an absolutely outstanding job on
this sixth edition (and who have put up with two very finicky authors who seem con-
genitally unable to meet deadlines!). Thanks also to our artists, Janet Theurer and
Patrice Rossi Calkin, for their work on the beautiful figures in this book, and to Andrea
Stefanowicz and her team at PreMediaGlobal for their wonderful production work on
this edition. Finally, a most special thanks go to Michael Hirsch, our editor at Addison-
Wesley, and Susan Hartman, our former editor at Addison-Wesley. This book would
not be what it is (and may well not have been at all) without their graceful manage-
ment, constant encouragement, nearly infinite patience, good humor, and perseverance.
xvi Preface
xvii
Table of Contents
Chapter 1 Computer Networks and the Internet 1
1.1 What Is the Internet? 2
1.1.1 A Nuts-and-Bolts Description 2
1.1.2 A Services Description 5
1.1.3 What Is a Protocol? 7
1.2 The Network Edge 9
1.2.1 Access Networks 12
1.2.2 Physical Media 18
1.3 The Network Core 22
1.3.1 Packet Switching 22
1.3.2 Circuit Switching 27
1.3.3 A Network of Networks 32
1.4 Delay, Loss, and Throughput in Packet-Switched Networks 35
1.4.1 Overview of Delay in Packet-Switched Networks 35
1.4.2 Queuing Delay and Packet Loss 39
1.4.3 End-to-End Delay 42
1.4.4 Throughput in Computer Networks 44
1.5 Protocol Layers and Their Service Models 47
1.5.1 Layered Architecture 47
1.5.2 Encapsulation 53
1.6 Networks Under Attack 55
1.7 History of Computer Networking and the Internet 60
1.7.1 The Development of Packet Switching: 19611972 60
1.7.2 Proprietary Networks and Internetworking: 19721980 62
1.7.3 A Proliferation of Networks: 19801990 63
1.7.4 The Internet Explosion: The 1990s 64
1.7.5 The New Millennium 65
1.8 Summary 66
Homework Problems and Questions 68
Wireshark Lab 78
Interview: Leonard Kleinrock 80
Chapter 2 Application Layer 83
2.1 Principles of Network Applications 84
2.1.1 Network Application Architectures 86
2.1.2 Processes Communicating 88
2.1.3 Transport Services Available to Applications 91
2.1.4 Transport Services Provided by the Internet 93
2.1.5 Application-Layer Protocols 96
2.1.6 Network Applications Covered in This Book 97
2.2 The Web and HTTP 98
2.2.1 Overview of HTTP 98
2.2.2 Non-Persistent and Persistent Connections 100
2.2.3 HTTP Message Format 103
2.2.4 User-Server Interaction: Cookies 108
2.2.5 Web Caching 110
2.2.6 The Conditional GET 114
2.3 File Transfer: FTP 116
2.3.1 FTP Commands and Replies 118
2.4 Electronic Mail in the Internet 118
2.4.1 SMTP 121
2.4.2 Comparison with HTTP 124
2.4.3 Mail Message Format 125
2.4.4 Mail Access Protocols 125
2.5 DNSThe Internets Directory Service 130
2.5.1 Services Provided by DNS 131
2.5.2 Overview of How DNS Works 133
2.5.3 DNS Records and Messages 139
2.6 Peer-to-Peer Applications 144
2.6.1 P2P File Distribution 145
2.6.2 Distributed Hash Tables (DHTs) 151
2.7 Socket Programming: Creating Network Applications 156
2.7.1 Socket Programming with UDP 157
2.7.2 Socket Programming with TCP 163
2.8 Summary 168
Homework Problems and Questions 169
Socket Programming Assignments 179
Wireshark Labs: HTTP, DNS 181
Interview: Marc Andreessen 182
xviii Table of Contents
Table of Contents xix
Chapter 3 Transport Layer 185
3.1 Introduction and Transport-Layer Services 186
3.1.1 Relationship Between Transport and Network Layers 186
3.1.2 Overview of the Transport Layer in the Internet 189
3.2 Multiplexing and Demultiplexing 191
3.3 Connectionless Transport: UDP 198
3.3.1 UDP Segment Structure 202
3.3.2 UDP Checksum 202
3.4 Principles of Reliable Data Transfer 204
3.4.1 Building a Reliable Data Transfer Protocol 206
3.4.2 Pipelined Reliable Data Transfer Protocols 215
3.4.3 Go-Back-N (GBN) 218
3.4.4 Selective Repeat (SR) 223
3.5 Connection-Oriented Transport: TCP 230
3.5.1 The TCP Connection 231
3.5.2 TCP Segment Structure 233
3.5.3 Round-Trip Time Estimation and Timeout 238
3.5.4 Reliable Data Transfer 242
3.5.5 Flow Control 250
3.5.6 TCP Connection Management 252
3.6 Principles of Congestion Control 259
3.6.1 The Causes and the Costs of Congestion 259
3.6.2 Approaches to Congestion Control 265
3.6.3 Network-Assisted Congestion-Control Example:
ATM ABR Congestion Control 266
3.7 TCP Congestion Control 269
3.7.1 Fairness 279
3.8 Summary 283
Homework Problems and Questions 285
Programming Assignments 300
Wireshark Labs: TCP, UDP 301
Interview: Van Jacobson 302
Chapter 4 The Network Layer 305
4.1 Introduction 306
4.1.1 Forwarding and Routing 308
4.1.2 Network Service Models 310
4.2 Virtual Circuit and Datagram Networks 313
4.2.1 Virtual-Circuit Networks 314
4.2.2 Datagram Networks 317
4.2.3 Origins of VC and Datagram Networks 319
4.3 Whats Inside a Router? 320
4.3.1 Input Processing 322
4.3.2 Switching 324
4.3.3 Output Processing 326
4.3.4 Where Does Queuing Occur? 327
4.3.5 The Routing Control Plane 331
4.4 The Internet Protocol (IP): Forwarding and Addressing in the Internet 331
4.4.1 Datagram Format 332
4.4.2 IPv4 Addressing 338
4.4.3 Internet Control Message Protocol (ICMP) 353
4.4.4 IPv6 356
4.4.5 A Brief Foray into IP Security 362
4.5 Routing Algorithms 363
4.5.1 The Link-State (LS) Routing Algorithm 366
4.5.2 The Distance-Vector (DV) Routing Algorithm 371
4.5.3 Hierarchical Routing 379
4.6 Routing in the Internet 383
4.6.1 Intra-AS Routing in the Internet: RIP 384
4.6.2 Intra-AS Routing in the Internet: OSPF 388
4.6.3 Inter-AS Routing: BGP 390
4.7 Broadcast and Multicast Routing 399
4.7.1 Broadcast Routing Algorithms 400
4.7.2 Multicast 405
4.8 Summary 412
Homework Problems and Questions 413
Programming Assignments 429
Wireshark Labs: IP, ICMP 430
Interview: Vinton G. Cerf 431
Chapter 5 The Link Layer: Links, Access Networks, and LANs 433
5.1 Introduction to the Link Layer 434
5.1.1 The Services Provided by the Link Layer 436
5.1.2 Where Is the Link Layer Implemented? 437
5.2 Error-Detection and -Correction Techniques 438
5.2.1 Parity Checks 440
5.2.2 Checksumming Methods 442
5.2.3 Cyclic Redundancy Check (CRC) 443
5.3 Multiple Access Links and Protocols 445
5.3.1 Channel Partitioning Protocols 448
5.3.2 Random Access Protocols 449
5.3.3 Taking-Turns Protocols 459
5.3.4 DOCSIS: The Link-Layer Protocol for Cable Internet Access 460
xx Table of Contents
Table of Contents xxi
5.4 Switched Local Area Networks 461
5.4.1 Link-Layer Addressing and ARP 462
5.4.2 Ethernet 469
5.4.3 Link-Layer Switches 476
5.4.4 Virtual Local Area Networks (VLANs) 482
5.5 Link Virtualization: A Network as a Link Layer 486
5.5.1 Multiprotocol Label Switching (MPLS) 487
5.6 Data Center Networking 490
5.7 Retrospective: A Day in the Life of a Web Page Request 495
5.7.1 Getting Started: DHCP, UDP, IP, and Ethernet 495
5.7.2 Still Getting Started: DNS and ARP 497
5.7.3 Still Getting Started: Intra-Domain Routing to the DNS Server 498
5.7.4 Web Client-Server Interaction: TCP and HTTP 499
5.8 Summary 500
Homework Problems and Questions 502
Wireshark Labs: Ethernet and ARP, DHCP 510
Interview: Simon S. Lam 511
Chapter 6 Wireless and Mobile Networks 513
6.1 Introduction 514
6.2 Wireless Links and Network Characteristics 519
6.2.1 CDMA 522
6.3 WiFi: 802.11 Wireless LANs 526
6.3.1 The 802.11 Architecture 527
6.3.2 The 802.11 MAC Protocol 531
6.3.3 The IEEE 802.11 Frame 537
6.3.4 Mobility in the Same IP Subnet 541
6.3.5 Advanced Features in 802.11 542
6.3.6 Personal Area Networks: Bluetooth and Zigbee 544
6.4 Cellular Internet Access 546
6.4.1 An Overview of Cellular Network Architecture 547
6.4.2 3G Cellular Data Networks: Extending the Internet to Cellular
Subscribers 550
6.4.3 On to 4G: LTE 553
6.5 Mobility Management: Principles 555
6.5.1 Addressing 557
6.5.2 Routing to a Mobile Node 559
6.6 Mobile IP 564
6.7 Managing Mobility in Cellular Networks 570
6.7.1 Routing Calls to a Mobile User 571
6.7.2 Handoffs in GSM 572
6.8 Wireless and Mobility: Impact on Higher-Layer Protocols 575
6.9 Summary 578
Homework Problems and Questions 578
Wireshark Lab: IEEE 802.11 (WiFi) 583
Interview: Deborah Estrin 584
Chapter 7 Multimedia Networking 587
7.1 Multimedia Networking Applications 588
7.1.1 Properties of Video 588
7.1.2 Properties of Audio 590
7.1.3 Types of Multimedia Network Applications 591
7.2 Streaming Stored Video 593
7.2.1 UDP Streaming 595
7.2.2 HTTP Streaming 596
7.2.3 Adaptive Streaming and DASH 600
7.2.4 Content Distribution Networks 602
7.2.5 Case Studies: Netflix, YouTube, and Kankan 608
7.3 Voice-over-IP 612
7.3.1 Limitations of the Best-Effort IP Service 612
7.3.2 Removing Jitter at the Receiver for Audio 614
7.3.3 Recovering from Packet Loss 617
7.3.4 Case Study: VoIP with Skype 620
7.4 Protocols for Real-Time Conversational Applications 623
7.4.1 RTP 624
7.4.2 SIP 627
7.5 Network Support for Multimedia 632
7.5.1 Dimensioning Best-Effort Networks 634
7.5.2 Providing Multiple Classes of Service 636
7.5.3 Diffserv 648
7.5.4 Per-Connection Quality-of-Service (QoS) Guarantees:
Resource Reservation and Call Admission 652
7.6 Summary 655
Homework Problems and Questions 656
Programming Assignment 666
Interview: Henning Schulzrinne 668
Chapter 8 Security in Computer Networks 671
8.1 What Is Network Security? 672
8.2 Principles of Cryptography 675
8.2.1 Symmetric Key Cryptography 676
8.2.2 Public Key Encryption 683
xxii Table of Contents
Table of Contents xxiii
8.3 Message Integrity and Digital Signatures 688
8.3.1 Cryptographic Hash Functions 689
8.3.2 Message Authentication Code 691
8.3.3 Digital Signatures 693
8.4 End-Point Authentication 700
8.4.1 Authentication Protocol ap1.0 700
8.4.2 Authentication Protocol ap2.0 701
8.4.3 Authentication Protocol ap3.0 702
8.4.4 Authentication Protocol ap3.1 703
8.4.5 Authentication Protocol ap4.0 703
8.5 Securing E-Mail 705
8.5.1 Secure E-Mail 706
8.5.2 PGP 710
8.6 Securing TCP Connections: SSL 711
8.6.1 The Big Picture 713
8.6.2 A More Complete Picture 716
8.7 Network-Layer Security: IPsec and Virtual Private Networks 718
8.7.1 IPsec and Virtual Private Networks (VPNs) 718
8.7.2 The AH and ESP Protocols 720
8.7.3 Security Associations 720
8.7.4 The IPsec Datagram 721
8.7.5 IKE: Key Management in IPsec 725
8.8 Securing Wireless LANs 726
8.8.1 Wired Equivalent Privacy (WEP) 726
8.8.2 IEEE 802.11i 728
8.9 Operational Security: Firewalls and Intrusion Detection Systems 731
8.9.1 Firewalls 731
8.9.2 Intrusion Detection Systems 739
8.10 Summary 742
Homework Problems and Questions 744
Wireshark Lab: SSL 752
IPsec Lab 752
Interview: Steven M. Bellovin 753
Chapter 9 Network Management 755
9.1 What Is Network Management? 756
9.2 The Infrastructure for Network Management 760
9.3 The Internet-Standard Management Framework 764
9.3.1 Structure of Management Information: SMI 766
9.3.2 Management Information Base: MIB 770
9.3.3 SNMP Protocol Operations and Transport Mappings 772
9.3.4 Security and Administration 775
9.4 ASN.1 778
9.5 Conclusion 783
Homework Problems and Questions 783
Interview: Jennifer Rexford 786
References 789
Index 823
xxiv Table of Contents
COMPUTER
NETWORKING
A Top-Down Approach
SIXTH EDITION
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CHAPTER 1
Computer
Networks and
the Internet
1
Todays Internet is arguably the largest engineered system ever created by mankind,
with hundreds of millions of connected computers, communication links, and
switches; with billions of users who connect via laptops, tablets, and smartphones;
and with an array of new Internet-connected devices such as sensors, Web cams,
game consoles, picture frames, and even washing machines. Given that the Internet
is so large and has so many diverse components and uses, is there any hope of
understanding how it works? Are there guiding principles and structure that can pro-
vide a foundation for understanding such an amazingly large and complex system?
And if so, is it possible that it actually could be both interesting and fun to learn
about computer networks? Fortunately, the answers to all of these questions is a
resounding YES! Indeed, its our aim in this book to provide you with a modern
introduction to the dynamic field of computer networking, giving you the principles
and practical insights youll need to understand not only todays networks, but
tomorrows as well.
This first chapter presents a broad overview of computer networking and the
Internet. Our goal here is to paint a broad picture and set the context for the rest of
this book, to see the forest through the trees. Well cover a lot of ground in this intro-
ductory chapter and discuss a lot of the pieces of a computer network, without los-
ing sight of the big picture.
Well structure our overview of computer networks in this chapter as follows.
After introducing some basic terminology and concepts, well first examine the
basic hardware and software components that make up a network. Well begin at
the networks edge and look at the end systems and network applications running
in the network. Well then explore the core of a computer network, examining the
links and the switches that transport data, as well as the access networks and phys-
ical media that connect end systems to the network core. Well learn that the Inter-
net is a network of networks, and well learn how these networks connect with
each other.
After having completed this overview of the edge and core of a computer net-
work, well take the broader and more abstract view in the second half of this chap-
ter. Well examine delay, loss, and throughput of data in a computer network and
provide simple quantitative models for end-to-end throughput and delay: models
that take into account transmission, propagation, and queuing delays. Well then
introduce some of the key architectural principles in computer networking, namely,
protocol layering and service models. Well also learn that computer networks are
vulnerable to many different types of attacks; well survey some of these attacks and
consider how computer networks can be made more secure. Finally, well close this
chapter with a brief history of computer networking.
1.1 What Is the Internet?
In this book, well use the public Internet, a specific computer network, as our prin-
cipal vehicle for discussing computer networks and their protocols. But what is the
Internet? There are a couple of ways to answer this question. First, we can describe
the nuts and bolts of the Internet, that is, the basic hardware and software components
that make up the Internet. Second, we can describe the Internet in terms of a net-
working infrastructure that provides services to distributed applications. Lets begin
with the nuts-and-bolts description, using Figure 1.1 to illustrate our discussion.
1.1.1 A Nuts-and-Bolts Description
The Internet is a computer network that interconnects hundreds of millions of com-
puting devices throughout the world. Not too long ago, these computing devices were
primarily traditional desktop PCs, Linux workstations, and so-called servers that store
and transmit information such as Web pages and e-mail messages. Increasingly,
however, nontraditional Internet end systems such as laptops, smartphones, tablets,
TVs, gaming consoles, Web cams, automobiles, environmental sensing devices,
picture frames, and home electrical and security systems are being connected to the
Internet. Indeed, the term computer network is beginning to sound a bit dated, given
the many nontraditional devices that are being hooked up to the Internet. In Internet jar-
gon, all of these devices are called hosts or end systems. As of July 2011, there were
2 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
1.1 WHAT IS THE INTERNET? 3
Figure 1.1 Some pieces of the Internet
Key:
Host
(= end system)
Server Mobile Router Link-Layer
switch
Modem Base
station
Smartphone Cell phone
tower
National or
Global ISP
Mobile Network
Local or
Regional ISP
Enterprise Network
Home Network
nearly 850 million end systems attached to the Internet [ISC 2012], not counting
smartphones, laptops, and other devices that are only intermittently connected to the
Internet. Overall, more there are an estimated 2 billion Internet users [ITU 2011].
End systems are connected together by a network of communication links and
packet switches. Well see in Section 1.2 that there are many types of communica-
tion links, which are made up of different types of physical media, including coaxial
cable, copper wire, optical fiber, and radio spectrum. Different links can transmit
data at different rates, with the transmission rate of a link measured in bits/second.
When one end system has data to send to another end system, the sending end sys-
tem segments the data and adds header bytes to each segment. The resulting pack-
ages of information, known as packets in the jargon of computer networks, are then
sent through the network to the destination end system, where they are reassembled
into the original data.
A packet switch takes a packet arriving on one of its incoming communication
links and forwards that packet on one of its outgoing communication links. Packet
switches come in many shapes and flavors, but the two most prominent types in
todays Internet are routers and link-layer switches. Both types of switches for-
ward packets toward their ultimate destinations. Link-layer switches are typically
used in access networks, while routers are typically used in the network core. The
sequence of communication links and packet switches traversed by a packet from
the sending end system to the receiving end system is known as a route or path
through the network. The exact amount of traffic being carried in the Internet is
difficult to estimate but Cisco [Cisco VNI 2011] estimates global Internet traffic will
be nearly 40 exabytes per month in 2012.
Packet-switched networks (which transport packets) are in many ways simi-
lar to transportation networks of highways, roads, and intersections (which trans-
port vehicles). Consider, for example, a factory that needs to move a large
amount of cargo to some destination warehouse located thousands of kilometers
away. At the factory, the cargo is segmented and loaded into a fleet of trucks.
Each of the trucks then independently travels through the network of highways,
roads, and intersections to the destination warehouse. At the destination ware-
house, the cargo is unloaded and grouped with the rest of the cargo arriving from
the same shipment. Thus, in many ways, packets are analogous to trucks, com-
munication links are analogous to highways and roads, packet switches are anal-
ogous to intersections, and end systems are analogous to buildings. Just as a truck
takes a path through the transportation network, a packet takes a path through a
computer network.
End systems access the Internet through Internet Service Providers (ISPs),
including residential ISPs such as local cable or telephone companies; corporate
ISPs; university ISPs; and ISPs that provide WiFi access in airports, hotels, coffee
shops, and other public places. Each ISP is in itself a network of packet switches
and communication links. ISPs provide a variety of types of network access to the
end systems, including residential broadband access such as cable modem or DSL,
4 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
high-speed local area network access, wireless access, and 56 kbps dial-up modem
access. ISPs also provide Internet access to content providers, connecting Web
sites directly to the Internet. The Internet is all about connecting end systems to
each other, so the ISPs that provide access to end systems must also be intercon-
nected. These lower-tier ISPs are interconnected through national and interna-
tional upper-tier ISPs such as Level 3 Communications, AT&T, Sprint, and NTT.
An upper-tier ISP consists of high-speed routers interconnected with high-speed
fiber-optic links. Each ISP network, whether upper-tier or lower-tier, is managed
independently, runs the IP protocol (see below), and conforms to certain naming
and address conventions. Well examine ISPs and their interconnection more
closely in Section 1.3.
End systems, packet switches, and other pieces of the Internet run protocols
that control the sending and receiving of information within the Internet. The
Transmission Control Protocol (TCP) and the Internet Protocol (IP) are two of
the most important protocols in the Internet. The IP protocol specifies the format of
the packets that are sent and received among routers and end systems. The Internets
principal protocols are collectively known as TCP/IP. Well begin looking into pro-
tocols in this introductory chapter. But thats just a startmuch of this book is con-
cerned with computer network protocols!
Given the importance of protocols to the Internet, its important that everyone
agree on what each and every protocol does, so that people can create systems and
products that interoperate. This is where standards come into play. Internet stan-
dards are developed by the Internet Engineering Task Force (IETF)[IETF 2012].
The IETF standards documents are called requests for comments (RFCs). RFCs
started out as general requests for comments (hence the name) to resolve network
and protocol design problems that faced the precursor to the Internet [Allman 2011].
RFCs tend to be quite technical and detailed. They define protocols such as TCP, IP,
HTTP (for the Web), and SMTP (for e-mail). There are currently more than 6,000
RFCs. Other bodies also specify standards for network components, most notably
for network links. The IEEE 802 LAN/MAN Standards Committee [IEEE 802
2012], for example, specifies the Ethernet and wireless WiFi standards.
1.1.2 A Services Description
Our discussion above has identified many of the pieces that make up the Internet.
But we can also describe the Internet from an entirely different anglenamely, as
an infrastructure that provides services to applications. These applications
include electronic mail, Web surfing, social networks, instant messaging, Voice-
over-IP (VoIP), video streaming, distributed games, peer-to-peer (P2P) file shar-
ing, television over the Internet, remote login, and much, much more. The
applications are said to be distributed applications, since they involve multiple
end systems that exchange data with each other. Importantly, Internet applications
1.1 WHAT IS THE INTERNET? 5
run on end systemsthey do not run in the packet switches in the network core.
Although packet switches facilitate the exchange of data among end systems, they
are not concerned with the application that is the source or sink of data.
Lets explore a little more what we mean by an infrastructure that provides
services to applications. To this end, suppose you have an exciting new idea for a
distributed Internet application, one that may greatly benefit humanity or one that
may simply make you rich and famous. How might you go about transforming
this idea into an actual Internet application? Because applications run on end sys-
tems, you are going to need to write programs that run on the end systems. You
might, for example, write your programs in Java, C, or Python. Now, because you
are developing a distributed Internet application, the programs running on the
different end systems will need to send data to each other. And here we get to a
central issueone that leads to the alternative way of describing the Internet as a
platform for applications. How does one program running on one end system
instruct the Internet to deliver data to another program running on another end
system?
End systems attached to the Internet provide an Application Programming
Interface (API) that specifies how a program running on one end system asks
the Internet infrastructure to deliver data to a specific destination program run-
ning on another end system. This Internet API is a set of rules that the sending
program must follow so that the Internet can deliver the data to the destination
program. Well discuss the Internet API in detail in Chapter 2. For now, lets
draw upon a simple analogy, one that we will frequently use in this book. Sup-
pose Alice wants to send a letter to Bob using the postal service. Alice, of course,
cant just write the letter (the data) and drop the letter out her window. Instead,
the postal service requires that Alice put the letter in an envelope; write Bobs
full name, address, and zip code in the center of the envelope; seal the envelope;
put a stamp in the upper-right-hand corner of the envelope; and finally, drop the
envelope into an official postal service mailbox. Thus, the postal service has its
own postal service API, or set of rules, that Alice must follow to have the
postal service deliver her letter to Bob. In a similar manner, the Internet has an
API that the program sending data must follow to have the Internet deliver the
data to the program that will receive the data.
The postal service, of course, provides more than one service to its customers.
It provides express delivery, reception confirmation, ordinary use, and many more
services. In a similar manner, the Internet provides multiple services to its applica-
tions. When you develop an Internet application, you too must choose one of the
Internets services for your application. Well describe the Internets services in
Chapter 2.
We have just given two descriptions of the Internet; one in terms of its hardware
and software components, the other in terms of an infrastructure for providing
services to distributed applications. But perhaps you are still confused as to what the
6 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Internet is. What are packet switching and TCP/IP? What are routers? What kinds of
communication links are present in the Internet? What is a distributed application?
How can a toaster or a weather sensor be attached to the Internet? If you feel a bit
overwhelmed by all of this now, dont worrythe purpose of this book is to intro-
duce you to both the nuts and bolts of the Internet and the principles that govern how
and why it works. Well explain these important terms and questions in the follow-
ing sections and chapters.
1.1.3 What Is a Protocol?
Now that weve got a bit of a feel for what the Internet is, lets consider another
important buzzword in computer networking: protocol. What is a protocol? What
does a protocol do?
A Human Analogy
It is probably easiest to understand the notion of a computer network protocol by
first considering some human analogies, since we humans execute protocols all of
the time. Consider what you do when you want to ask someone for the time of day.
A typical exchange is shown in Figure 1.2. Human protocol (or good manners, at
least) dictates that one first offer a greeting (the first Hi in Figure 1.2) to initiate
communication with someone else. The typical response to a Hi is a returned
Hi message. Implicitly, one then takes a cordial Hi response as an indication
that one can proceed and ask for the time of day. A different response to the initial
Hi (such as Dont bother me! or I dont speak English, or some unprintable
reply) might indicate an unwillingness or inability to communicate. In this case,
the human protocol would be not to ask for the time of day. Sometimes one gets no
response at all to a question, in which case one typically gives up asking that per-
son for the time. Note that in our human protocol, there are specific messages we
send, and specific actions we take in response to the received reply messages or
other events (such as no reply within some given amount of time). Clearly, trans-
mitted and received messages, and actions taken when these messages are sent or
received or other events occur, play a central role in a human protocol. If people
run different protocols (for example, if one person has manners but the other does
not, or if one understands the concept of time and the other does not) the protocols
do not interoperate and no useful work can be accomplished. The same is true in
networkingit takes two (or more) communicating entities running the same pro-
tocol in order to accomplish a task.
Lets consider a second human analogy. Suppose youre in a college class (a
computer networking class, for example!). The teacher is droning on about proto-
cols and youre confused. The teacher stops to ask, Are there any questions? (a
1.1 WHAT IS THE INTERNET? 7
message that is transmitted to, and received by, all students who are not sleeping).
You raise your hand (transmitting an implicit message to the teacher). Your teacher
acknowledges you with a smile, saying Yes . . . (a transmitted message encourag-
ing you to ask your questionteachers love to be asked questions), and you then ask
your question (that is, transmit your message to your teacher). Your teacher hears
your question (receives your question message) and answers (transmits a reply to
you). Once again, we see that the transmission and receipt of messages, and a set of
conventional actions taken when these messages are sent and received, are at the
heart of this question-and-answer protocol.
Network Protocols
A network protocol is similar to a human protocol, except that the entities exchang-
ing messages and taking actions are hardware or software components of some
device (for example, computer, smartphone, tablet, router, or other network-capable
8 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
GET http://www.awl.com/kurose-ross
TCP connection request
Time Time
TCP
conn
ectio
n re
ply
Hi
Got the time?
Time Time
Hi
2:00
Figure 1.2 A human protocol and a computer network protocol
device). All activity in the Internet that involves two or more communicating remote
entities is governed by a protocol. For example, hardware-implemented protocols in
two physically connected computers control the flow of bits on the wire between
the two network interface cards; congestion-control protocols in end systems con-
trol the rate at which packets are transmitted between sender and receiver; protocols
in routers determine a packets path from source to destination. Protocols are run-
ning everywhere in the Internet, and consequently much of this book is about com-
puter network protocols.
As an example of a computer network protocol with which you are probably
familiar, consider what happens when you make a request to a Web server, that is,
when you type the URL of a Web page into your Web browser. The scenario is illus-
trated in the right half of Figure 1.2. First, your computer will send a connection
request message to the Web server and wait for a reply. The Web server will eventu-
ally receive your connection request message and return a connection reply mes-
sage. Knowing that it is now OK to request the Web document, your computer then
sends the name of the Web page it wants to fetch from that Web server in a GET
message. Finally, the Web server returns the Web page (file) to your computer.
Given the human and networking examples above, the exchange of messages
and the actions taken when these messages are sent and received are the key defin-
ing elements of a protocol:
A protocol defines the format and the order of messages exchanged between
two or more communicating entities, as well as the actions taken on the trans-
mission and/or receipt of a message or other event.
The Internet, and computer networks in general, make extensive use of proto-
cols. Different protocols are used to accomplish different communication tasks. As
you read through this book, you will learn that some protocols are simple and
straightforward, while others are complex and intellectually deep. Mastering the
field of computer networking is equivalent to understanding the what, why, and how
of networking protocols.
1.2 The Network Edge
In the previous section we presented a high-level overview of the Internet and net-
working protocols. We are now going to delve a bit more deeply into the compo-
nents of a computer network (and the Internet, in particular). We begin in this
section at the edge of a network and look at the components with which we are most
familiarnamely, the computers, smartphones and other devices that we use on a
daily basis. In the next section well move from the network edge to the network
core and examine switching and routing in computer networks.
1.2 THE NETWORK EDGE 9
Recall from the previous section that in computer networking jargon, the com-
puters and other devices connected to the Internet are often referred to as end sys-
tems. They are referred to as end systems because they sit at the edge of the Internet,
as shown in Figure 1.3. The Internets end systems include desktop computers (e.g.,
desktop PCs, Macs, and Linux boxes), servers (e.g., Web and e-mail servers), and
mobile computers (e.g., laptops, smartphones, and tablets). Furthermore, an increas-
ing number of non-traditional devices are being attached to the Internet as end sys-
tems (see sidebar).
End systems are also referred to as hosts because they host (that is, run) appli-
cation programs such as a Web browser program, a Web server program, an e-mail
client program, or an e-mail server program. Throughout this book we will use the
terms hosts and end systems interchangeably; that is, host = end system. Hosts are
sometimes further divided into two categories: clients and servers. Informally,
clients tend to be desktop and mobile PCs, smartphones, and so on, whereas servers
tend to be more powerful machines that store and distribute Web pages, stream
video, relay e-mail, and so on. Today, most of the servers from which we receive
10 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
A DIZZYING ARRAY OF INTERNET END SYSTEMS
Not too long ago, the end-system devices connected to the Internet were primarily
traditional computers such as desktop machines and powerful servers. Beginning in
the late 1990s and continuing today, a wide range of interesting devices are being
connected to the Internet, leveraging their ability to send and receive digital data.
Given the Internets ubiquity, its well-defined (standardized) protocols, and the
availability of Internet-ready commodity hardware, its natural to use Internet tech-
nology to network these devices together and to Internet-connected servers.
Many of these devices are based in the homevideo game consoles (e.g.,
Microsofts Xbox), Internet-ready televisions, digital picture frames that download
and display digital pictures, washing machines, refrigerators, and even a toaster
that downloads meteorological information and burns an image of the days fore-
cast (e.g., mixed clouds and sun) on your morning toast [BBC 2001]. IP-enabled
phones with GPS capabilities put location-dependent services (maps, information
about nearby services or people) at your fingertips. Networked sensors embedded
into the physical environment allow monitoring of buildings, bridges, seismic activi-
ty, wildlife habitats, river estuaries, and the weather. Biomedical devices can be
embedded and networked in a body-area network. With so many diverse devices
being networked together, the Internet is indeed becoming an Internet of things
[ITU 2005b].
CASE HISTORY
search results, e-mail, Web pages, and videos reside in large data centers. For
example, Google has 3050 data centers, with many having more than one hundred
thousand servers.
Mobile Network
National or
Global ISP
Local or
Regional ISP
Enterprise Network
Home Network
Figure 1.3 End-system interaction
1.2 THE NETWORK EDGE 11
12 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
National or
Global ISP
Mobile Network
Local or
Regional ISP
Enterprise Network
Home Network
Figure 1.4 Access networks
1.2.1 Access Networks
Having considered the applications and end systems at the edge of the network,
lets next consider the access networkthe network that physically connects an end
system to the first router (also known as the edge router) on a path from the end
system to any other distant end system. Figure 1.4 shows several types of access
networks with thick, shaded lines, and the settings (home, enterprise, and wide-area
mobile wireless) in which they are used.
Home Access: DSL, Cable, FTTH, Dial-Up, and Satellite
In developed countries today, more than 65 percent of the households have Internet
access, with Korea, Netherlands, Finland, and Sweden leading the way with more than
80 percent of households having Internet access, almost all via a high-speed broadband
connection [ITU 2011]. Finland and Spain have recently declared high-speed Internet
access to be a legal right. Given this intense interest in home access, lets begin our
overview of access networks by considering how homes connect to the Internet.
Today, the two most prevalent types of broadband residential access are digital
subscriber line (DSL) and cable. A residence typically obtains DSL Internet access
from the same local telephone company (telco) that provides its wired local phone
access. Thus, when DSL is used, a customers telco is also its ISP. As shown in
Figure 1.5, each customers DSL modem uses the existing telephone line (twisted-
pair copper wire, which well discuss in Section 1.2.2) to exchange data with a digi-
tal subscriber line access multiplexer (DSLAM) located in the telcos local central
office (CO). The homes DSL modem takes digital data and translates it to high-
frequency tones for transmission over telephone wires to the CO; the analog signals
from many such houses are translated back into digital format at the DSLAM.
The residential telephone line carries both data and traditional telephone sig-
nals simultaneously, which are encoded at different frequencies:
A high-speed downstream channel, in the 50 kHz to 1 MHz band
A medium-speed upstream channel, in the 4 kHz to 50 kHz band
An ordinary two-way telephone channel, in the 0 to 4 kHz band
This approach makes the single DSL link appear as if there were three separate
links, so that a telephone call and an Internet connection can share the DSL link at
the same time. (Well describe this technique of frequency-division multiplexing in
Home PC
Home
phone
DSL
modem
Internet
Telephone
network
Splitter
Existing phone line:
0-4KHz phone; 4-50KHz
upstream data; 50KHz
1MHz downstream data
Central
office
DSLAM
Figure 1.5 DSL Internet access
1.2 THE NETWORK EDGE 13
14 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Section 1.3.1). On the customer side, a splitter separates the data and telephone sig-
nals arriving to the home and forwards the data signal to the DSL modem. On the
telco side, in the CO, the DSLAM separates the data and phone signals and sends
the data into the Internet. Hundreds or even thousands of households connect to a
single DSLAM [Dischinger 2007].
The DSL standards define transmission rates of 12 Mbps downstream and
1.8 Mbps upstream [ITU 1999], and 24 Mbps downstream and 2.5 Mbps upstream
[ITU 2003]. Because the downstream and upstream rates are different, the access is
said to be asymmetric. The actual downstream and upstream transmission rates
achieved may be less than the rates noted above, as the DSL provider may purpose-
fully limit a residential rate when tiered service (different rates, available at differ-
ent prices) are offered, or because the maximum rate can be limited by the distance
between the home and the CO, the gauge of the twisted-pair line and the degree of
electrical interference. Engineers have expressly designed DSL for short distances
between the home and the CO; generally, if the residence is not located within 5 to 10
miles of the CO, the residence must resort to an alternative form of Internet access.
While DSL makes use of the telcos existing local telephone infrastructure,
cable Internet access makes use of the cable television companys existing cable
television infrastructure. A residence obtains cable Internet access from the same
company that provides its cable television. As illustrated in Figure 1.6, fiber optics
connect the cable head end to neighborhood-level junctions, from which tradi-
tional coaxial cable is then used to reach individual houses and apartments. Each
neighborhood junction typically supports 500 to 5,000 homes. Because both fiber
and coaxial cable are employed in this system, it is often referred to as hybrid
fiber coax (HFC).
Fiber
cable
Coaxial cable
Hundreds
of homes
Cable head end
Hundreds
of homes
Fiber
node
Fiber
node
Internet
CMTS
Figure 1.6 A hybrid fiber-coaxial access network
Cable internet access requires special modems, called cable modems. As with a
DSL modem, the cable modem is typically an external device and connects to the
home PC through an Ethernet port. (We will discuss Ethernet in great detail in
Chapter 5.) At the cable head end, the cable modem termination system (CMTS)
serves a similar function as the DSL networks DSLAMturning the analog signal
sent from the cable modems in many downstream homes back into digital format.
Cable modems divide the HFC network into two channels, a downstream and an
upstream channel. As with DSL, access is typically asymmetric, with the down-
stream channel typically allocated a higher transmission rate than the upstream
channel. The DOCSIS 2.0 standard defines downstream rates up to 42.8 Mbps and
upstream rates of up to 30.7 Mbps. As in the case of DSL networks, the maximum
achievable rate may not be realized due to lower contracted data rates or media
impairments.
One important characteristic of cable Internet access is that it is a shared
broadcast medium. In particular, every packet sent by the head end travels down-
stream on every link to every home and every packet sent by a home travels on the
upstream channel to the head end. For this reason, if several users are simultane-
ously downloading a video file on the downstream channel, the actual rate at which
each user receives its video file will be significantly lower than the aggregate cable
downstream rate. On the other hand, if there are only a few active users and they
are all Web surfing, then each of the users may actually receive Web pages at the
full cable downstream rate, because the users will rarely request a Web page at
exactly the same time. Because the upstream channel is also shared, a distributed
multiple access protocol is needed to coordinate transmissions and avoid collisions.
(Well discuss this collision issue in some detail in Chapter 5.)
Although DSL and cable networks currently represent more than 90 percent of
residential broadband access in the United States, an up-and-coming technology that
promises even higher speeds is the deployment of fiber to the home (FTTH)
[FTTH Council 2011a]. As the name suggests, the FTTH concept is simple
provide an optical fiber path from the CO directly to the home. In the United States,
Verizon has been particularly aggressive with FTTH with its FIOS service [Verizon
FIOS 2012].
There are several competing technologies for optical distribution from the
CO to the homes. The simplest optical distribution network is called direct fiber,
with one fiber leaving the CO for each home. More commonly, each fiber leav-
ing the central office is actually shared by many homes; it is not until the fiber
gets relatively close to the homes that it is split into individual customer-specific
fibers. There are two competing optical-distribution network architectures that
perform this splitting: active optical networks (AONs) and passive optical net-
works (PONs). AON is essentially switched Ethernet, which is discussed in
Chapter 5.
Here, we briefly discuss PON, which is used in Verizons FIOS service.
Figure 1.7 shows FTTH using the PON distribution architecture. Each home has
1.2 THE NETWORK EDGE 15
an optical network terminator (ONT), which is connected by dedicated optical
fiber to a neighborhood splitter. The splitter combines a number of homes (typically
less than 100) onto a single, shared optical fiber, which connects to an optical line
terminator (OLT) in the telcos CO. The OLT, providing conversion between optical
and electrical signals, connects to the Internet via a telco router. In the home, users
connect a home router (typically a wireless router) to the ONT and access the Inter-
net via this home router. In the PON architecture, all packets sent from OLT to the
splitter are replicated at the splitter (similar to a cable head end).
FTTH can potentially provide Internet access rates in the gigabits per second
range. However, most FTTH ISPs provide different rate offerings, with the higher
rates naturally costing more money. The average downstream speed of US FTTH
customers was approximately 20 Mbps in 2011 (compared with 13 Mbps for cable
access networks and less than 5 Mbps for DSL) [FTTH Council 2011b].
Two other access network technologies are also used to provide Internet access
to the home. In locations where DSL, cable, and FTTH are not available (e.g., in
some rural settings), a satellite link can be used to connect a residence to the Inter-
net at speeds of more than 1 Mbps; StarBand and HughesNet are two such satellite
access providers. Dial-up access over traditional phone lines is based on the same
model as DSLa home modem connects over a phone line to a modem in the ISP.
Compared with DSL and other broadband access networks, dial-up access is excru-
ciatingly slow at 56 kbps.
Access in the Enterprise (and the Home): Ethernet and WiFi
On corporate and university campuses, and increasingly in home settings, a local area
network (LAN) is used to connect an end system to the edge router. Although there
are many types of LAN technologies, Ethernet is by far the most prevalent access
technology in corporate, university, and home networks. As shown in Figure 1.8,
Ethernet users use twisted-pair copper wire to connect to an Ethernet switch, a
16 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Internet
Central office
Optical
splitter
ONT
ONT
ONT
OLT
Optical
fibers
Figure 1.7 FTTH Internet access
technology discussed in detail in Chapter 5. The Ethernet switch, or a network of
such interconnected switches, is then in turn connected into the larger Internet. With
Ethernet access, users typically have 100 Mbps access to the Ethernet switch,
whereas servers may have 1 Gbps or even 10 Gbps access.
Increasingly, however, people are accessing the Internet wirelessly from lap-
tops, smartphones, tablets, and other devices (see earlier sidebar on A Dizzying
Array of Devices). In a wireless LAN setting, wireless users transmit/receive pack-
ets to/from an access point that is connected into the enterprises network (most
likely including wired Ethernet), which in turn is connected to the wired Internet. A
wireless LAN user must typically be within a few tens of meters of the access point.
Wireless LAN access based on IEEE 802.11 technology, more colloquially known
as WiFi, is now just about everywhereuniversities, business offices, cafes, air-
ports, homes, and even in airplanes. In many cities, one can stand on a street corner
and be within range of ten or twenty base stations (for a browseable global map of
802.11 base stations that have been discovered and logged on a Web site by people
who take great enjoyment in doing such things, see [wigle.net 2012]). As discussed
in detail in Chapter 6, 802.11 today provides a shared transmission rate of up to
54 Mbps.
Even though Ethernet and WiFi access networks were initially deployed in enter-
prise (corporate, university) settings, they have recently become relatively common
components of home networks. Many homes combine broadband residential access
(that is, cable modems or DSL) with these inexpensive wireless LAN technologies to
create powerful home networks [Edwards 2011]. Figure 1.9 shows a typical home
network. This home network consists of a roaming laptop as well as a wired PC; a
base station (the wireless access point), which communicates with the wireless PC; a
cable modem, providing broadband access to the Internet; and a router, which inter-
connects the base station and the stationary PC with the cable modem. This network
allows household members to have broadband access to the Internet with one mem-
ber roaming from the kitchen to the backyard to the bedrooms.
Ethernet
switch
Institutional
router
100 Mbps
100 Mbps
100 Mbps
Server
To Institutions
ISP
Figure 1.8 Ethernet Internet access
1.2 THE NETWORK EDGE 17
Wide-Area Wireless Access: 3G and LTE
Increasingly, devices such as iPhones, BlackBerrys, and Android devices are being
used to send email, surf the Web, Tweet, and download music while on the run.
These devices employ the same wireless infrastructure used for cellular telephony
to send/receive packets through a base station that is operated by the cellular net-
work provider. Unlike WiFi, a user need only be within a few tens of kilometers (as
opposed to a few tens of meters) of the base station.
Telecommunications companies have made enormous investments in so-called
third-generation (3G) wireless, which provides packet-switched wide-area wireless
Internet access at speeds in excess of 1 Mbps. But even higher-speed wide-area
access technologiesa fourth-generation (4G) of wide-area wireless networksare
already being deployed. LTE ( for Long-Term Evolutiona candidate for Bad
Acronym of the Year Award) has its roots in 3G technology, and can potentially
achieve rates in excess of 10 Mbps. LTE downstream rates of many tens of Mbps
have been reported in commercial deployments. Well cover the basic principles of
wireless networks and mobility, as well as WiFi, 3G, and LTE technologies (and
more!) in Chapter 6.
1.2.2 Physical Media
In the previous subsection, we gave an overview of some of the most important
network access technologies in the Internet. As we described these technologies,
we also indicated the physical media used. For example, we said that HFC uses a
combination of fiber cable and coaxial cable. We said that DSL and Ethernet use
copper wire. And we said that mobile access networks use the radio spectrum.
18 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Cable
head end
House
Internet
Figure 1.9 A typical home network
In this subsection we provide a brief overview of these and other transmission
media that are commonly used in the Internet.
In order to define what is meant by a physical medium, let us reflect on the
brief life of a bit. Consider a bit traveling from one end system, through a series of
links and routers, to another end system. This poor bit gets kicked around and
transmitted many, many times! The source end system first transmits the bit, and
shortly thereafter the first router in the series receives the bit; the first router then
transmits the bit, and shortly thereafter the second router receives the bit; and so
on. Thus our bit, when traveling from source to destination, passes through a series
of transmitter-receiver pairs. For each transmitter-receiver pair, the bit is sent by
propagating electromagnetic waves or optical pulses across a physical medium.
The physical medium can take many shapes and forms and does not have to be of
the same type for each transmitter-receiver pair along the path. Examples of physi-
cal media include twisted-pair copper wire, coaxial cable, multimode fiber-optic
cable, terrestrial radio spectrum, and satellite radio spectrum. Physical media fall
into two categories: guided media and unguided media. With guided media, the
waves are guided along a solid medium, such as a fiber-optic cable, a twisted-pair
copper wire, or a coaxial cable. With unguided media, the waves propagate in the
atmosphere and in outer space, such as in a wireless LAN or a digital satellite
channel.
But before we get into the characteristics of the various media types, let us say
a few words about their costs. The actual cost of the physical link (copper wire,
fiber-optic cable, and so on) is often relatively minor compared with other network-
ing costs. In particular, the labor cost associated with the installation of the physical
link can be orders of magnitude higher than the cost of the material. For this reason,
many builders install twisted pair, optical fiber, and coaxial cable in every room in a
building. Even if only one medium is initially used, there is a good chance that
another medium could be used in the near future, and so money is saved by not hav-
ing to lay additional wires in the future.
Twisted-Pair Copper Wire
The least expensive and most commonly used guided transmission medium is
twisted-pair copper wire. For over a hundred years it has been used by telephone
networks. In fact, more than 99 percent of the wired connections from the tele-
phone handset to the local telephone switch use twisted-pair copper wire. Most of
us have seen twisted pair in our homes and work environments. Twisted pair con-
sists of two insulated copper wires, each about 1 mm thick, arranged in a regular
spiral pattern. The wires are twisted together to reduce the electrical interference
from similar pairs close by. Typically, a number of pairs are bundled together in a
cable by wrapping the pairs in a protective shield. A wire pair constitutes a single
communication link. Unshielded twisted pair (UTP) is commonly used for
1.2 THE NETWORK EDGE 19
computer networks within a building, that is, for LANs. Data rates for LANs
using twisted pair today range from 10 Mbps to 10 Gbps. The data rates that can
be achieved depend on the thickness of the wire and the distance between trans-
mitter and receiver.
When fiber-optic technology emerged in the 1980s, many people disparaged
twisted pair because of its relatively low bit rates. Some people even felt that fiber-
optic technology would completely replace twisted pair. But twisted pair did not
give up so easily. Modern twisted-pair technology, such as category 6a cable, can
achieve data rates of 10 Gbps for distances up to a hundred meters. In the end,
twisted pair has emerged as the dominant solution for high-speed LAN networking.
As discussed earlier, twisted pair is also commonly used for residential Internet
access. We saw that dial-up modem technology enables access at rates of up to 56
kbps over twisted pair. We also saw that DSL (digital subscriber line) technology
has enabled residential users to access the Internet at tens of Mbps over twisted pair
(when users live close to the ISPs modem).
Coaxial Cable
Like twisted pair, coaxial cable consists of two copper conductors, but the two con-
ductors are concentric rather than parallel. With this construction and special insula-
tion and shielding, coaxial cable can achieve high data transmission rates. Coaxial
cable is quite common in cable television systems. As we saw earlier, cable televi-
sion systems have recently been coupled with cable modems to provide residential
users with Internet access at rates of tens of Mbps. In cable television and cable
Internet access, the transmitter shifts the digital signal to a specific frequency band,
and the resulting analog signal is sent from the transmitter to one or more receivers.
Coaxial cable can be used as a guided shared medium. Specifically, a number of
end systems can be connected directly to the cable, with each of the end systems
receiving whatever is sent by the other end systems.
Fiber Optics
An optical fiber is a thin, flexible medium that conducts pulses of light, with each
pulse representing a bit. A single optical fiber can support tremendous bit rates, up
to tens or even hundreds of gigabits per second. They are immune to electromag-
netic interference, have very low signal attenuation up to 100 kilometers, and are
very hard to tap. These characteristics have made fiber optics the preferred long-
haul guided transmission media, particularly for overseas links. Many of the long-
distance telephone networks in the United States and elsewhere now use fiber optics
exclusively. Fiber optics is also prevalent in the backbone of the Internet. However,
the high cost of optical devicessuch as transmitters, receivers, and switcheshas
hindered their deployment for short-haul transport, such as in a LAN or into the
20 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
home in a residential access network. The Optical Carrier (OC) standard link speeds
range from 51.8 Mbps to 39.8 Gbps; these specifications are often referred to as OC-
n, where the link speed equals n 51.8 Mbps. Standards in use today include OC-1,
OC-3, OC-12, OC-24, OC-48, OC-96, OC-192, OC-768. [Mukherjee 2006,
Ramaswamy 2010] provide coverage of various aspects of optical networking.
Terrestrial Radio Channels
Radio channels carry signals in the electromagnetic spectrum. They are an attractive
medium because they require no physical wire to be installed, can penetrate walls,
provide connectivity to a mobile user, and can potentially carry a signal for long dis-
tances. The characteristics of a radio channel depend significantly on the propagation
environment and the distance over which a signal is to be carried. Environmental con-
siderations determine path loss and shadow fading (which decrease the signal strength
as the signal travels over a distance and around/through obstructing objects), multi-
path fading (due to signal reflection off of interfering objects), and interference (due
to other transmissions and electromagnetic signals).
Terrestrial radio channels can be broadly classified into three groups: those that
operate over very short distance (e.g., with one or two meters); those that operate in
local areas, typically spanning from ten to a few hundred meters; and those that
operate in the wide area, spanning tens of kilometers. Personal devices such as wire-
less headsets, keyboards, and medical devices operate over short distances; the
wireless LAN technologies described in Section 1.2.1 use local-area radio channels;
the cellular access technologies use wide-area radio channels. Well discuss radio
channels in detail in Chapter 6.
Satellite Radio Channels
A communication satellite links two or more Earth-based microwave transmitter/
receivers, known as ground stations. The satellite receives transmissions on one fre-
quency band, regenerates the signal using a repeater (discussed below), and transmits
the signal on another frequency. Two types of satellites are used in communications:
geostationary satellites and low-earth orbiting (LEO) satellites.
Geostationary satellites permanently remain above the same spot on Earth. This
stationary presence is achieved by placing the satellite in orbit at 36,000 kilometers
above Earths surface. This huge distance from ground station through satellite back
to ground station introduces a substantial signal propagation delay of 280 millisec-
onds. Nevertheless, satellite links, which can operate at speeds of hundreds of Mbps,
are often used in areas without access to DSL or cable-based Internet access.
LEO satellites are placed much closer to Earth and do not remain permanently
above one spot on Earth. They rotate around Earth (just as the Moon does) and may
communicate with each other, as well as with ground stations. To provide continuous
1.2 THE NETWORK EDGE 21
22 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
coverage to an area, many satellites need to be placed in orbit. There are currently
many low-altitude communication systems in development. Lloyds satellite con-
stellations Web page [Wood 2012] provides and collects information on satellite
constellation systems for communications. LEO satellite technology may be used
for Internet access sometime in the future.
1.3 The Network Core
Having examined the Internets edge, let us now delve more deeply inside the net-
work corethe mesh of packet switches and links that interconnects the Internets
end systems. Figure 1.10 highlights the network core with thick, shaded lines.
1.3.1 Packet Switching
In a network application, end systems exchange messages with each other. Mes-
sages can contain anything the application designer wants. Messages may perform a
control function (for example, the Hi messages in our handshaking example in
Figure 1.2) or can contain data, such as an email message, a JPEG image, or an MP3
audio file. To send a message from a source end system to a destination end system,
the source breaks long messages into smaller chunks of data known as packets.
Between source and destination, each packet travels through communication links
and packet switches (for which there are two predominant types, routers and link-
layer switches). Packets are transmitted over each communication link at a rate
equal to the full transmission rate of the link. So, if a source end system or a packet
switch is sending a packet of L bits over a link with transmission rate R bits/sec, then
the time to transmit the packet is L/R seconds.
Store-and-Forward Transmission
Most packet switches use store-and-forward transmission at the inputs to the
links. Store-and-forward transmission means that the packet switch must receive
the entire packet before it can begin to transmit the first bit of the packet onto the
outbound link. To explore store-and-forward transmission in more detail, consider
a simple network consisting of two end systems connected by a single router, as
shown in Figure 1.11. A router will typically have many incident links, since its job
is to switch an incoming packet onto an outgoing link; in this simple example, the
router has the rather simple task of transferring a packet from one (input) link to
the only other attached link. In this example, the source has three packets, each
consisting of L bits, to send to the destination. At the snapshot of time shown in
Figure 1.11, the source has transmitted some of packet 1, and the front of packet 1
has already arrived at the router. Because the router employs store-and-forwarding,
at this instant of time, the router cannot transmit the bits it has received; instead it
National or
Global ISP
Local or
Regional ISP
Enterprise Network
Home Network
Mobile Network
Figure 1.10 The network core
1.3 THE NETWORK CORE 23
24 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
must first buffer (i.e., store) the packets bits. Only after the router has received
all of the packets bits can it begin to transmit (i.e., forward) the packet onto the
outbound link. To gain some insight into store-and-forward transmission, lets now
calculate the amount of time that elapses from when the source begins to send the
packet until the destination has received the entire packet. (Here we will ignore
propagation delaythe time it takes for the bits to travel across the wire at near the
speed of lightwhich will be discussed in Section 1.4.) The source begins to trans-
mit at time 0; at time L/R seconds, the source has transmitted the entire packet, and
the entire packet has been received and stored at the router (since there is no propa-
gation delay). At time L/R seconds, since the router has just received the entire
packet, it can begin to transmit the packet onto the outbound link towards the desti-
nation; at time 2L/R, the router has transmitted the entire packet, and the entire
packet has been received by the destination. Thus, the total delay is 2L/R. If
the switch instead forwarded bits as soon as they arrive (without first receiving the
entire packet), then the total delay would be L/R since bits are not held up at
the router. But, as we will discuss in Section 1.4, routers need to receive, store, and
process the entire packet before forwarding.
Now lets calculate the amount of time that elapses from when the source
begins to send the first packet until the destination has received all three packets.
As before, at time L/R, the router begins to forward the first packet. But also at time
L/R the source will begin to send the second packet, since it has just finished send-
ing the entire first packet. Thus, at time 2L/R, the destination has received the first
packet and the router has received the second packet. Similarly, at time 3L/R, the
destination has received the first two packets and the router has received the third
packet. Finally, at time 4L/R the destination has received all three packets!
Lets now consider the general case of sending one packet from source to desti-
nation over a path consisting of N links each of rate R (thus, there are N-1 routers
between source and destination). Applying the same logic as above, we see that the
end-to-end delay is:
(1.1)
You may now want to try to determine what the delay would be for P packets sent
over a series of N links.
dend@to@end = N
L
R
Source
R bps
12
DestinationFront of packet 1
stored in router,
awaiting remaining
bits before forwarding
3
Figure 1.11 Store-and-forward packet switching
Queuing Delays and Packet Loss
Each packet switch has multiple links attached to it. For each attached link, the
packet switch has an output buffer (also called an output queue), which stores
packets that the router is about to send into that link. The output buffers play a key
role in packet switching. If an arriving packet needs to be transmitted onto a link but
finds the link busy with the transmission of another packet, the arriving packet must
wait in the output buffer. Thus, in addition to the store-and-forward delays, packets
suffer output buffer queuing delays. These delays are variable and depend on the
level of congestion in the network. Since the amount of buffer space is finite, an
arriving packet may find that the buffer is completely full with other packets wait-
ing for transmission. In this case, packet loss will occureither the arriving packet
or one of the already-queued packets will be dropped.
Figure 1.12 illustrates a simple packet-switched network. As in Figure 1.11,
packets are represented by three-dimensional slabs. The width of a slab represents
the number of bits in the packet. In this figure, all packets have the same width and
hence the same length. Suppose Hosts A and B are sending packets to Host E. Hosts
A and B first send their packets along 10 Mbps Ethernet links to the first router. The
router then directs these packets to the 1.5 Mbps link. If, during a short interval of
time, the arrival rate of packets to the router (when converted to bits per second)
exceeds 1.5 Mbps, congestion will occur at the router as packets queue in the links
output buffer before being transmitted onto the link. For example, if Host A and B
each send a burst of five packets back-to-back at the same time, then most of these
packets will spend some time waiting in the queue. The situation is, in fact, entirely
analogous to many common-day situationsfor example, when we wait in line for
a bank teller or wait in front of a tollbooth. Well examine this queuing delay in
more detail in Section 1.4.
1.3 THE NETWORK CORE 25
10 Mbps Ethernet
Key:
Packets
A
B
C
D E
1.5 Mbps
Queue of
packets waiting
for output link
Figure 1.12 Packet switching
26 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Forwarding Tables and Routing Protocols
Earlier, we said that a router takes a packet arriving on one of its attached
communication links and forwards that packet onto another one of its attached com-
munication links. But how does the router determine which link it should forward
the packet onto? Packet forwarding is actually done in different ways in different
types of computer networks. Here, we briefly describe how it is done in the
Internet.
In the Internet, every end system has an address called an IP address. When a
source end system wants to send a packet to a destination end system, the source
includes the destinations IP address in the packets header. As with postal addresses,
this address has a hierarchical structure. When a packet arrives at a router in the
network, the router examines a portion of the packets destination address and for-
wards the packet to an adjacent router. More specifically, each router has a
forwarding table that maps destination addresses (or portions of the destination
addresses) to that routers outbound links. When a packet arrives at a router, the
router examines the address and searches its forwarding table, using this destination
address, to find the appropriate outbound link. The router then directs the packet to
this outbound link.
The end-to-end routing process is analogous to a car driver who does not use
maps but instead prefers to ask for directions. For example, suppose Joe is driving
from Philadelphia to 156 Lakeside Drive in Orlando, Florida. Joe first drives to his
neighborhood gas station and asks how to get to 156 Lakeside Drive in Orlando,
Florida. The gas station attendant extracts the Florida portion of the address and
tells Joe that he needs to get onto the interstate highway I-95 South, which has an
entrance just next to the gas station. He also tells Joe that once he enters Florida,
he should ask someone else there. Joe then takes I-95 South until he gets to Jack-
sonville, Florida, at which point he asks another gas station attendant for directions.
The attendant extracts the Orlando portion of the address and tells Joe that he
should continue on I-95 to Daytona Beach and then ask someone else. In Daytona
Beach, another gas station attendant also extracts the Orlando portion of the
address and tells Joe that he should take I-4 directly to Orlando. Joe takes I-4 and
gets off at the Orlando exit. Joe goes to another gas station attendant, and this time
the attendant extracts the Lakeside Drive portion of the address and tells Joe the
road he must follow to get to Lakeside Drive. Once Joe reaches Lakeside Drive, he
asks a kid on a bicycle how to get to his destination. The kid extracts the 156 por-
tion of the address and points to the house. Joe finally reaches his ultimate destina-
tion. In the above analogy, the gas station attendants and kids on bicycles are
analogous to routers.
We just learned that a router uses a packets destination address to index a for-
warding table and determine the appropriate outbound link. But this statement begs
yet another question: How do forwarding tables get set? Are they configured by
hand in each and every router, or does the Internet use a more automated procedure?
This issue will be studied in depth in Chapter 4. But to whet your appetite here,
well note now that the Internet has a number of special routing protocols that are
used to automatically set the forwarding tables. A routing protocol may, for exam-
ple, determine the shortest path from each router to each destination and use the
shortest path results to configure the forwarding tables in the routers.
How would you actually like to see the end-to-end route that packets take in the
Internet? We now invite you to get your hands dirty by interacting with the Trace-
route program. Simply visit the site www.traceroute.org, choose a source in a particu-
lar country, and trace the route from that source to your computer. (For a discussion of
Traceroute, see Section 1.4.)
1.3.2 Circuit Switching
There are two fundamental approaches to moving data through a network of links
and switches: circuit switching and packet switching. Having covered packet-
switched networks in the previous subsection, we now turn our attention to circuit-
switched networks.
In circuit-switched networks, the resources needed along a path (buffers, link
transmission rate) to provide for communication between the end systems are
reserved for the duration of the communication session between the end systems. In
packet-switched networks, these resources are not reserved; a sessions messages
use the resources on demand, and as a consequence, may have to wait (that is,
queue) for access to a communication link. As a simple analogy, consider two
restaurants, one that requires reservations and another that neither requires reserva-
tions nor accepts them. For the restaurant that requires reservations, we have to go
through the hassle of calling before we leave home. But when we arrive at the
restaurant we can, in principle, immediately be seated and order our meal. For the
restaurant that does not require reservations, we dont need to bother to reserve a
table. But when we arrive at the restaurant, we may have to wait for a table before
we can be seated.
Traditional telephone networks are examples of circuit-switched networks.
Consider what happens when one person wants to send information (voice or fac-
simile) to another over a telephone network. Before the sender can send the infor-
mation, the network must establish a connection between the sender and the
receiver. This is a bona fide connection for which the switches on the path
between the sender and receiver maintain connection state for that connection. In
the jargon of telephony, this connection is called a circuit. When the network
establishes the circuit, it also reserves a constant transmission rate in the net-
works links (representing a fraction of each links transmission capacity) for the
duration of the connection. Since a given transmission rate has been reserved for
this sender-to-receiver connection, the sender can transfer the data to the receiver
at the guaranteed constant rate.
1.3 THE NETWORK CORE 27
www.traceroute.org
28 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Figure 1.13 A simple circuit-switched network consisting of
four switches and four links
Figure 1.13 illustrates a circuit-switched network. In this network, the four cir-
cuit switches are interconnected by four links. Each of these links has four circuits,
so that each link can support four simultaneous connections. The hosts (for exam-
ple, PCs and workstations) are each directly connected to one of the switches. When
two hosts want to communicate, the network establishes a dedicated end-to-end
connection between the two hosts. Thus, in order for Host A to communicate with
Host B, the network must first reserve one circuit on each of two links. In this exam-
ple, the dedicated end-to-end connection uses the second circuit in the first link and
the fourth circuit in the second link. Because each link has four circuits, for each
link used by the end-to-end connection, the connection gets one fourth of the links
total transmission capacity for the duration of the connection. Thus, for example, if
each link between adjacent switches has a transmission rate of 1 Mbps, then each
end-to-end circuit-switch connection gets 250 kbps of dedicated transmission rate.
In contrast, consider what happens when one host wants to send a packet to
another host over a packet-switched network, such as the Internet. As with circuit
switching, the packet is transmitted over a series of communication links. But different
from circuit switching, the packet is sent into the network without reserving any link
resources whatsoever. If one of the links is congested because other packets need to be
transmitted over the link at the same time, then the packet will have to wait in a buffer
at the sending side of the transmission link and suffer a delay. The Internet makes its
best effort to deliver packets in a timely manner, but it does not make any guarantees.
Multiplexing in Circuit-Switched Networks
A circuit in a link is implemented with either frequency-division multiplexing
(FDM) or time-division multiplexing (TDM). With FDM, the frequency spec-
trum of a link is divided up among the connections established across the link.
Specifically, the link dedicates a frequency band to each connection for the
duration of the connection. In telephone networks, this frequency band typically
has a width of 4 kHz (that is, 4,000 hertz or 4,000 cycles per second). The width
of the band is called, not surprisingly, the bandwidth. FM radio stations also use
FDM to share the frequency spectrum between 88 MHz and 108 MHz, with each
station being allocated a specific frequency band.
For a TDM link, time is divided into frames of fixed duration, and each frame
is divided into a fixed number of time slots. When the network establishes a connec-
tion across a link, the network dedicates one time slot in every frame to this connec-
tion. These slots are dedicated for the sole use of that connection, with one time slot
available for use (in every frame) to transmit the connections data.
Figure 1.14 illustrates FDM and TDM for a specific network link supporting up
to four circuits. For FDM, the frequency domain is segmented into four bands, each
of bandwidth 4 kHz. For TDM, the time domain is segmented into frames, with four
time slots in each frame; each circuit is assigned the same dedicated slot in the
revolving TDM frames. For TDM, the transmission rate of a circuit is equal to the
frame rate multiplied by the number of bits in a slot. For example, if the link trans-
mits 8,000 frames per second and each slot consists of 8 bits, then the transmission
rate of a circuit is 64 kbps.
Proponents of packet switching have always argued that circuit switching is
wasteful because the dedicated circuits are idle during silent periods. For example,
1.3 THE NETWORK CORE 29
4KHz
TDM
FDM
Link Frequency
4KHz
Slot
Key:
All slots labeled 2 are dedicated
to a specific sender-receiver pair.
Frame
1
2
2 3 4 1 2 3 4 1 2 3 4 1 2 3 4
Time
Figure 1.14 With FDM, each circuit continuously gets a fraction of the
bandwidth. With TDM, each circuit gets all of the bandwidth
periodically during brief intervals of time (that is, during slots)
30 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
when one person in a telephone call stops talking, the idle network resources (fre-
quency bands or time slots in the links along the connections route) cannot be used
by other ongoing connections. As another example of how these resources can be
underutilized, consider a radiologist who uses a circuit-switched network to
remotely access a series of x-rays. The radiologist sets up a connection, requests an
image, contemplates the image, and then requests a new image. Network resources
are allocated to the connection but are not used (i.e., are wasted) during the radiolo-
gists contemplation periods. Proponents of packet switching also enjoy pointing out
that establishing end-to-end circuits and reserving end-to-end transmission capacity
is complicated and requires complex signaling software to coordinate the operation
of the switches along the end-to-end path.
Before we finish our discussion of circuit switching, lets work through a numeri-
cal example that should shed further insight on the topic. Let us consider how long it
takes to send a file of 640,000 bits from Host A to Host B over a circuit-switched net-
work. Suppose that all links in the network use TDM with 24 slots and have a bit rate
of 1.536 Mbps. Also suppose that it takes 500 msec to establish an end-to-end circuit
before Host A can begin to transmit the file. How long does it take to send the file?
Each circuit has a transmission rate of (1.536 Mbps)/24 = 64 kbps, so it takes (640,000
bits)/(64 kbps) = 10 seconds to transmit the file. To this 10 seconds we add the circuit
establishment time, giving 10.5 seconds to send the file. Note that the transmission
time is independent of the number of links: The transmission time would be 10 sec-
onds if the end-to-end circuit passed through one link or a hundred links. (The actual
end-to-end delay also includes a propagation delay; see Section 1.4.)
Packet Switching Versus Circuit Switching
Having described circuit switching and packet switching, let us compare the two.
Critics of packet switching have often argued that packet switching is not suitable
for real-time services (for example, telephone calls and video conference calls)
because of its variable and unpredictable end-to-end delays (due primarily to vari-
able and unpredictable queuing delays). Proponents of packet switching argue that
(1) it offers better sharing of transmission capacity than circuit switching and (2) it
is simpler, more efficient, and less costly to implement than circuit switching. An
interesting discussion of packet switching versus circuit switching is [Molinero-
Fernandez 2002]. Generally speaking, people who do not like to hassle with restau-
rant reservations prefer packet switching to circuit switching.
Why is packet switching more efficient? Lets look at a simple example. Sup-
pose users share a 1 Mbps link. Also suppose that each user alternates between peri-
ods of activity, when a user generates data at a constant rate of 100 kbps, and periods
of inactivity, when a user generates no data. Suppose further that a user is active
only 10 percent of the time (and is idly drinking coffee during the remaining 90 per-
cent of the time). With circuit switching, 100 kbps must be reserved for each user at
all times. For example, with circuit-switched TDM, if a one-second frame is divided
into 10 time slots of 100 ms each, then each user would be allocated one time slot
per frame.
Thus, the circuit-switched link can support only 10 (= 1 Mbps/100 kbps) simul-
taneous users. With packet switching, the probability that a specific user is active is
0.1 (that is, 10 percent). If there are 35 users, the probability that there are 11 or
more simultaneously active users is approximately 0.0004. (Homework Problem P8
outlines how this probability is obtained.) When there are 10 or fewer simultane-
ously active users (which happens with probability 0.9996), the aggregate arrival
rate of data is less than or equal to 1 Mbps, the output rate of the link. Thus, when
there are 10 or fewer active users, users packets flow through the link essentially
without delay, as is the case with circuit switching. When there are more than 10
simultaneously active users, then the aggregate arrival rate of packets exceeds the
output capacity of the link, and the output queue will begin to grow. (It continues to
grow until the aggregate input rate falls back below 1 Mbps, at which point the
queue will begin to diminish in length.) Because the probability of having more than
10 simultaneously active users is minuscule in this example, packet switching pro-
vides essentially the same performance as circuit switching, but does so while
allowing for more than three times the number of users.
Lets now consider a second simple example. Suppose there are 10 users and that
one user suddenly generates one thousand 1,000-bit packets, while other users
remain quiescent and do not generate packets. Under TDM circuit switching with 10
slots per frame and each slot consisting of 1,000 bits, the active user can only use its
one time slot per frame to transmit data, while the remaining nine time slots in each
frame remain idle. It will be 10 seconds before all of the active users one million bits
of data has been transmitted. In the case of packet switching, the active user can con-
tinuously send its packets at the full link rate of 1 Mbps, since there are no other users
generating packets that need to be multiplexed with the active users packets. In this
case, all of the active users data will be transmitted within 1 second.
The above examples illustrate two ways in which the performance of packet
switching can be superior to that of circuit switching. They also highlight the crucial
difference between the two forms of sharing a links transmission rate among multi-
ple data streams. Circuit switching pre-allocates use of the transmission link regard-
less of demand, with allocated but unneeded link time going unused. Packet
switching on the other hand allocates link use on demand. Link transmission capac-
ity will be shared on a packet-by-packet basis only among those users who have
packets that need to be transmitted over the link.
Although packet switching and circuit switching are both prevalent in todays
telecommunication networks, the trend has certainly been in the direction of packet
switching. Even many of todays circuit-switched telephone networks are slowly
migrating toward packet switching. In particular, telephone networks often use
packet switching for the expensive overseas portion of a telephone call.
1.3 THE NETWORK CORE 31
1.3.3 A Network of Networks
We saw earlier that end systems (PCs, smartphones, Web servers, mail servers,
and so on) connect into the Internet via an access ISP. The access ISP can pro-
vide either wired or wireless connectivity, using an array of access technologies
including DSL, cable, FTTH, Wi-Fi, and cellular. Note that the access ISP does
not have to be a telco or a cable company; instead it can be, for example, a uni-
versity (providing Internet access to students, staff, and faculty), or a company
(providing access for its employees). But connecting end users and content
providers into an access ISP is only a small piece of solving the puzzle of con-
necting the billions of end systems that make up the Internet. To complete this
puzzle, the access ISPs themselves must be interconnected. This is done by cre-
ating a network of networksunderstanding this phrase is the key to understand-
ing the Internet.
Over the years, the network of networks that forms the Internet has evolved into
a very complex structure. Much of this evolution is driven by economics and
national policy, rather than by performance considerations. In order to understand
todays Internet network structure, lets incrementally build a series of network
structures, with each new structure being a better approximation of the complex
Internet that we have today. Recall that the overarching goal is to interconnect the
access ISPs so that all end systems can send packets to each other. One naive
approach would be to have each access ISP directly connect with every other access
ISP. Such a mesh design is, of course, much too costly for the access ISPs, as it
would require each access ISP to have a separate communication link to each of the
hundreds of thousands of other access ISPs all over the world.
Our first network structure, Network Structure 1, interconnects all of the
access ISPs with a single global transit ISP. Our (imaginary) global transit ISP is
a network of routers and communication links that not only spans the globe, but
also has at least one router near each of the hundreds of thousands of access ISPs.
Of course, it would be very costly for the global ISP to build such an extensive
network. To be profitable, it would naturally charge each of the access ISPs for
connectivity, with the pricing reflecting (but not necessarily directly proportional
to) the amount of traffic an access ISP exchanges with the global ISP. Since the
access ISP pays the global transit ISP, the access ISP is said to be a customer and
the global transit ISP is said to be a provider.
Now if some company builds and operates a global transit ISP that is profitable,
then it is natural for other companies to build their own global transit ISPs and com-
pete with the original global transit ISP. This leads to Network Structure 2, which
consists of the hundreds of thousands of access ISPs and multiple global transit
ISPs. The access ISPs certainly prefer Network Structure 2 over Network Structure
1 since they can now choose among the competing global transit providers as a
function of their pricing and services. Note, however, that the global transit ISPs
32 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
themselves must interconnect: Otherwise access ISPs connected to one of the global
transit providers would not be able to communicate with access ISPs connected to
the other global transit providers.
Network Structure 2, just described, is a two-tier hierarchy with global transit
providers residing at the top tier and access ISPs at the bottom tier. This assumes
that global transit ISPs are not only capable of getting close to each and every access
ISP, but also find it economically desirable to do so. In reality, although some ISPs
do have impressive global coverage and do directly connect with many access ISPs,
no ISP has presence in each and every city in the world. Instead, in any given region,
there may be a regional ISP to which the access ISPs in the region connect. Each
regional ISP then connects to tier-1 ISPs. Tier-1 ISPs are similar to our (imaginary)
global transit ISP; but tier-1 ISPs, which actually do exist, do not have a presence in
every city in the world. There are approximately a dozen tier-1 ISPs, including
Level 3 Communications, AT&T, Sprint, and NTT. Interestingly, no group officially
sanctions tier-1 status; as the saying goesif you have to ask if youre a member of
a group, youre probably not.
Returning to this network of networks, not only are there multiple competing tier-
1 ISPs, there may be multiple competing regional ISPs in a region. In such a hierar-
chy, each access ISP pays the regional ISP to which it connects, and each regional ISP
pays the tier-1 ISP to which it connects. (An access ISP can also connect directly to a
tier-1 ISP, in which case it pays the tier-1 ISP). Thus, there is customer-provider
relationship at each level of the hierarchy. Note that the tier-1 ISPs do not pay anyone
as they are at the top of the hierarchy. To further complicate matters, in some regions,
there may be a larger regional ISP (possibly spanning an entire country) to which the
smaller regional ISPs in that region connect; the larger regional ISP then connects to a
tier-1 ISP. For example, in China, there are access ISPs in each city, which connect to
provincial ISPs, which in turn connect to national ISPs, which finally connect to tier-1
ISPs [Tian 2012]. We refer to this multi-tier hierarchy, which is still only a crude
approximation of todays Internet, as Network Structure 3.
To build a network that more closely resembles todays Internet, we must add
points of presence (PoPs), multi-homing, peering, and Internet exchange points
(IXPs) to the hierarchical Network Structure 3. PoPs exist in all levels of the hier-
archy, except for the bottom (access ISP) level. A PoP is simply a group of one or
more routers (at the same location) in the providers network where customer ISPs
can connect into the provider ISP. For a customer network to connect to a
providers PoP, it can lease a high-speed link from a third-party telecommunica-
tions provider to directly connect one of its routers to a router at the PoP. Any ISP
(except for tier-1 ISPs) may choose to multi-home, that is, to connect to two or
more provider ISPs. So, for example, an access ISP may multi-home with two
regional ISPs, or it may multi-home with two regional ISPs and also with a tier-1
ISP. Similarly, a regional ISP may multi-home with multiple tier-1 ISPs. When an
1.3 THE NETWORK CORE 33
ISP multi-homes, it can continue to send and receive packets into the Internet even
if one of its providers has a failure.
As we just learned, customer ISPs pay their provider ISPs to obtain global
Internet interconnectivity. The amount that a customer ISP pays a provider ISP
reflects the amount of traffic it exchanges with the provider. To reduce these costs, a
pair of nearby ISPs at the same level of the hierarchy can peer, that is, they can
directly connect their networks together so that all the traffic between them passes
over the direct connection rather than through upstream intermediaries. When two
ISPs peer, it is typically settlement-free, that is, neither ISP pays the other. As noted
earlier, tier-1 ISPs also peer with one another, settlement-free. For a readable dis-
cussion of peering and customer-provider relationships, see [Van der Berg 2008].
Along these same lines, a third-party company can create an Internet Exchange
Point (IXP) (typically in a stand-alone building with its own switches), which is a
meeting point where multiple ISPs can peer together. There are roughly 300 IXPs in
the Internet today [Augustin 2009]. We refer to this ecosystemconsisting of
access ISPs, regional ISPs, tier-1 ISPs, PoPs, multi-homing, peering, and IXPsas
Network Structure 4.
We now finally arrive at Network Structure 5, which describes the Internet of
2012. Network Structure 5, illustrated in Figure 1.15, builds on top of Network
Structure 4 by adding content provider networks. Google is currently one of the
leading examples of such a content provider network. As of this writing, it is esti-
mated that Google has 30 to 50 data centers distributed across North America,
Europe, Asia, South America, and Australia. Some of these data centers house
over one hundred thousand servers, while other data centers are smaller, housing
only hundreds of servers. The Google data centers are all interconnected via
Googles private TCP/IP network, which spans the entire globe but is nevertheless
separate from the public Internet. Importantly, the Google private network only
34 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
access
ISP
access
ISP
access
ISP
access
ISP
access
ISP
access
ISP
access
ISP
access
ISP
Regional
ISP
Tier 1
ISP
Content provider
(e.g., Google)
Tier 1
ISP
IXP
Regional
ISP
IXP IXP
Figure 1.15 Interconnection of ISPs
carries traffic to/from Google servers. As shown in Figure 1.15, the Google
private network attempts to bypass the upper tiers of the Internet by peering
(settlement free) with lower-tier ISPs, either by directly connecting with them or
by connecting with them at IXPs [Labovitz 2010]. However, because many access
ISPs can still only be reached by transiting through tier-1 networks, the Google
network also connects to tier-1 ISPs, and pays those ISPs for the traffic it
exchanges with them. By creating its own network, a content provider not only
reduces its payments to upper-tier ISPs, but also has greater control of how its
services are ultimately delivered to end users. Googles network infrastructure is
described in greater detail in Section 7.2.4.
In summary, todays Interneta network of networksis complex, consisting of
a dozen or so tier-1 ISPs and hundreds of thousands of lower-tier ISPs. The ISPs are
diverse in their coverage, with some spanning multiple continents and oceans, and
others limited to narrow geographic regions. The lower-tier ISPs connect to the
higher-tier ISPs, and the higher-tier ISPs interconnect with one another. Users and
content providers are customers of lower-tier ISPs, and lower-tier ISPs are customers
of higher-tier ISPs. In recent years, major content providers have also created their
own networks and connect directly into lower-tier ISPs where possible.
1.4 Delay, Loss, and Throughput
in Packet-Switched Networks
Back in Section 1.1 we said that the Internet can be viewed as an infrastructure
that provides services to distributed applications running on end systems. Ideally,
we would like Internet services to be able to move as much data as we want
between any two end systems, instantaneously, without any loss of data. Alas, this
is a lofty goal, one that is unachievable in reality. Instead, computer networks nec-
essarily constrain throughput (the amount of data per second that can be trans-
ferred) between end systems, introduce delays between end systems, and can
actually lose packets. On one hand, it is unfortunate that the physical laws of real-
ity introduce delay and loss as well as constrain throughput. On the other hand,
because computer networks have these problems, there are many fascinating
issues surrounding how to deal with the problemsmore than enough issues to
fill a course on computer networking and to motivate thousands of PhD theses! In
this section, well begin to examine and quantify delay, loss, and throughput in
computer networks.
1.4.1 Overview of Delay in Packet-Switched Networks
Recall that a packet starts in a host (the source), passes through a series of routers,
and ends its journey in another host (the destination). As a packet travels from one
node (host or router) to the subsequent node (host or router) along this path, the
1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 35
packet suffers from several types of delays at each node along the path. The most
important of these delays are the nodal processing delay, queuing delay, transmis-
sion delay, and propagation delay; together, these delays accumulate to give a total
nodal delay. The performance of many Internet applicationssuch as search, Web
browsing, email, maps, instant messaging, and voice-over-IPare greatly affected
by network delays. In order to acquire a deep understanding of packet switching and
computer networks, we must understand the nature and importance of these delays.
Types of Delay
Lets explore these delays in the context of Figure 1.16. As part of its end-to-end
route between source and destination, a packet is sent from the upstream node
through router A to router B. Our goal is to characterize the nodal delay at router A.
Note that router A has an outbound link leading to router B. This link is preceded by
a queue (also known as a buffer). When the packet arrives at router A from the
upstream node, router A examines the packets header to determine the appropriate
outbound link for the packet and then directs the packet to this link. In this example,
the outbound link for the packet is the one that leads to router B. A packet can be
transmitted on a link only if there is no other packet currently being transmitted on
the link and if there are no other packets preceding it in the queue; if the link is
currently busy or if there are other packets already queued for the link, the newly
arriving packet will then join the queue.
Processing Delay
The time required to examine the packets header and determine where to direct the
packet is part of the processing delay. The processing delay can also include other
factors, such as the time needed to check for bit-level errors in the packet that occurred
in transmitting the packets bits from the upstream node to router A. Processing delays
36 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
A
B
Nodal
processing
Queueing
(waiting for
transmission)
Transmission
Propagation
Figure 1.16 The nodal delay at router A
1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 37
in high-speed routers are typically on the order of microseconds or less. After this
nodal processing, the router directs the packet to the queue that precedes the link to
router B. (In Chapter 4 well study the details of how a router operates.)
Queuing Delay
At the queue, the packet experiences a queuing delay as it waits to be transmitted onto
the link. The length of the queuing delay of a specific packet will depend on the num-
ber of earlier-arriving packets that are queued and waiting for transmission onto the
link. If the queue is empty and no other packet is currently being transmitted, then our
packets queuing delay will be zero. On the other hand, if the traffic is heavy and many
other packets are also waiting to be transmitted, the queuing delay will be long. We
will see shortly that the number of packets that an arriving packet might expect to find
is a function of the intensity and nature of the traffic arriving at the queue. Queuing
delays can be on the order of microseconds to milliseconds in practice.
Transmission Delay
Assuming that packets are transmitted in a first-come-first-served manner, as is com-
mon in packet-switched networks, our packet can be transmitted only after all the
packets that have arrived before it have been transmitted. Denote the length of the
packet by L bits, and denote the transmission rate of the link from router A to router
B by R bits/sec. For example, for a 10 Mbps Ethernet link, the rate is R = 10 Mbps;
for a 100 Mbps Ethernet link, the rate is R = 100 Mbps. The transmission delay is
L/R. This is the amount of time required to push (that is, transmit) all of the packets
bits into the link. Transmission delays are typically on the order of microseconds to
milliseconds in practice.
Propagation Delay
Once a bit is pushed into the link, it needs to propagate to router B. The time
required to propagate from the beginning of the link to router B is the propagation
delay. The bit propagates at the propagation speed of the link. The propagation
speed depends on the physical medium of the link (that is, fiber optics, twisted-pair
copper wire, and so on) and is in the range of
2 108 meters/sec to 3 108 meters/sec
which is equal to, or a little less than, the speed of light. The propagation delay is
the distance between two routers divided by the propagation speed. That is, the
propagation delay is d/s, where d is the distance between router A and router B and s
is the propagation speed of the link. Once the last bit of the packet propagates to
node B, it and all the preceding bits of the packet are stored in router B. The whole
process then continues with router B now performing the forwarding. In wide-area
networks, propagation delays are on the order of milliseconds.
Comparing Transmission and Propagation Delay
Newcomers to the field of computer networking sometimes have difficulty under-
standing the difference between transmission delay and propagation delay. The differ-
ence is subtle but important. The transmission delay is the amount of time required for
the router to push out the packet; it is a function of the packets length and the trans-
mission rate of the link, but has nothing to do with the distance between the two
routers. The propagation delay, on the other hand, is the time it takes a bit to propagate
from one router to the next; it is a function of the distance between the two routers, but
has nothing to do with the packets length or the transmission rate of the link.
An analogy might clarify the notions of transmission and propagation delay. Con-
sider a highway that has a tollbooth every 100 kilometers, as shown in Figure 1.17.
You can think of the highway segments between tollbooths as links and the toll-
booths as routers. Suppose that cars travel (that is, propagate) on the highway at a
rate of 100 km/hour (that is, when a car leaves a tollbooth, it instantaneously accel-
erates to 100 km/hour and maintains that speed between tollbooths). Suppose next
that 10 cars, traveling together as a caravan, follow each other in a fixed order. You
can think of each car as a bit and the caravan as a packet. Also suppose that each
tollbooth services (that is, transmits) a car at a rate of one car per 12 seconds, and
that it is late at night so that the caravans cars are the only cars on the highway.
Finally, suppose that whenever the first car of the caravan arrives at a tollbooth, it
waits at the entrance until the other nine cars have arrived and lined up behind it.
(Thus the entire caravan must be stored at the tollbooth before it can begin to be for-
warded.) The time required for the tollbooth to push the entire caravan onto the
highway is (10 cars)/(5 cars/minute) = 2 minutes. This time is analogous to the
transmission delay in a router. The time required for a car to travel from the exit of
one tollbooth to the next tollbooth is 100 km/(100 km/hour) = 1 hour. This time is
analogous to propagation delay. Therefore, the time from when the caravan is stored
in front of a tollbooth until the caravan is stored in front of the next tollbooth is the
sum of transmission delay and propagation delayin this example, 62 minutes.
38 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
Ten-car
caravan
Toll
booth
Toll
booth
100 km 100 km
Figure 1.17 Caravan analogy
1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 39
Lets explore this analogy a bit more. What would happen if the tollbooth service
time for a caravan were greater than the time for a car to travel between tollbooths?
For example, suppose now that the cars travel at the rate of 1,000 km/hour and the toll-
booth services cars at the rate of one car per minute. Then the traveling delay between
two tollbooths is 6 minutes and the time to serve a caravan is 10 minutes. In this case,
the first few cars in the caravan will arrive at the second tollbooth before the last cars
in the caravan leave the first tollbooth. This situation also arises in packet-switched
networksthe first bits in a packet can arrive at a router while many of the remaining
bits in the packet are still waiting to be transmitted by the preceding router.
If a picture speaks a thousand words, then an animation must speak a million
words. The companion Web site for this textbook provides an interactive Java applet
that nicely illustrates and contrasts transmission delay and propagation delay. The
reader is highly encouraged to visit that applet. [Smith 2009] also provides a very
readable discussion of propagation, queueing, and transmission delays.
If we let dproc, dqueue, dtrans, and dprop denote the processing, queuing, transmis-
sion, and propagation delays, then the total nodal delay is given by
dnodal = dproc + dqueue + dtrans + dprop
The contribution of these delay components can vary significantly. For example,
dprop can be negligible (for example, a couple of microseconds) for a link connect-
ing two routers on the same university campus; however, dprop is hundreds of mil-
liseconds for two routers interconnected by a geostationary satellite link, and can be
the dominant term in dnodal. Similarly, dtrans can range from negligible to significant.
Its contribution is typically negligible for transmission rates of 10 Mbps and higher
(for example, for LANs); however, it can be hundreds of milliseconds for large
Internet packets sent over low-speed dial-up modem links. The processing delay,
dproc, is often negligible; however, it strongly influences a routers maximum
throughput, which is the maximum rate at which a router can forward packets.
1.4.2 Queuing Delay and Packet Loss
The most complicated and interesting component of nodal delay is the queuing
delay, dqueue. In fact, queuing delay is so important and interesting in computer net-
working that thousands of papers and numerous books have been written about it
[Bertsekas 1991; Daigle 1991; Kleinrock 1975, 1976; Ross 1995]. We give only a
high-level, intuitive discussion of queuing delay here; the more curious reader may
want to browse through some of the books (or even eventually write a PhD thesis
on the subject!). Unlike the other three delays (namely, dproc, dtrans, and dprop), the
queuing delay can vary from packet to packet. For example, if 10 packets arrive
at an empty queue at the same time, the first packet transmitted will suffer no queu-
ing delay, while the last packet transmitted will suffer a relatively large queuing
delay (while it waits for the other nine packets to be transmitted). Therefore, when
40 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
characterizing queuing delay, one typically uses statistical measures, such as aver-
age queuing delay, variance of queuing delay, and the probability that the queuing
delay exceeds some specified value.
When is the queuing delay large and when is it insignificant? The answer to this
question depends on the rate at which traffic arrives at the queue, the transmission
rate of the link, and the nature of the arriving traffic, that is, whether the traffic arrives
periodically or arrives in bursts. To gain some insight here, let a denote the average
rate at which packets arrive at the queue (a is in units of packets/sec). Recall that R is
the transmission rate; that is, it is the rate (in bits/sec) at which bits are pushed out of
the queue. Also suppose, for simplicity, that all packets consist of L bits. Then the
average rate at which bits arrive at the queue is La bits/sec. Finally, assume that the
queue is very big, so that it can hold essentially an infinite number of bits. The ratio
La/R, called the traffic intensity, often plays an important role in estimating the
extent of the queuing delay. If La/R > 1, then the average rate at which bits arrive at
the queue exceeds the rate at which the bits can be transmitted from the queue. In this
unfortunate situation, the queue will tend to increase without bound and the queuing
delay will approach infinity! Therefore, one of the golden rules in traffic engineering
is: Design your system so that the traffic intensity is no greater than 1.
Now consider the case La/R 1. Here, the nature of the arriving traffic impacts
the queuing delay. For example, if packets arrive periodicallythat is, one packet
arrives every L/R secondsthen every packet will arrive at an empty queue and
there will be no queuing delay. On the other hand, if packets arrive in bursts but
periodically, there can be a significant average queuing delay. For example, suppose
N packets arrive simultaneously every (L/R)N seconds. Then the first packet trans-
mitted has no queuing delay; the second packet transmitted has a queuing delay of
L/R seconds; and more generally, the nth packet transmitted has a queuing delay of
(n 1)L/R seconds. We leave it as an exercise for you to calculate the average queu-
ing delay in this example.
The two examples of periodic arrivals described above are a bit academic. Typ-
ically, the arrival process to a queue is random; that is, the arrivals do not follow any
pattern and the packets are spaced apart by random amounts of time. In this more
realistic case, the quantity La/R is not usually sufficient to fully characterize the
queueing delay statistics. Nonetheless, it is useful in gaining an intuitive understand-
ing of the extent of the queuing delay. In particular, if the traffic intensity is close to
zero, then packet arrivals are few and far between and it is unlikely that an arriving
packet will find another packet in the queue. Hence, the average queuing delay will
be close to zero. On the other hand, when the traffic intensity is close to 1, there will
be intervals of time when the arrival rate exceeds the transmission capacity (due to
variations in packet arrival rate), and a queue will form during these periods of time;
when the arrival rate is less than the transmission capacity, the length of the queue
will shrink. Nonetheless, as the traffic intensity approaches 1, the average queue
length gets larger and larger. The qualitative dependence of average queuing delay
on the traffic intensity is shown in Figure 1.18.
1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 41
One important aspect of Figure 1.18 is the fact that as the traffic intensity
approaches 1, the average queuing delay increases rapidly. A small percentage
increase in the intensity will result in a much larger percentage-wise increase in
delay. Perhaps you have experienced this phenomenon on the highway. If you regu-
larly drive on a road that is typically congested, the fact that the road is typically
congested means that its traffic intensity is close to 1. If some event causes an even
slightly larger-than-usual amount of traffic, the delays you experience can be huge.
To really get a good feel for what queuing delays are about, you are encouraged
once again to visit the companion Web site, which provides an interactive Java
applet for a queue. If you set the packet arrival rate high enough so that the traffic
intensity exceeds 1, you will see the queue slowly build up over time.
Packet Loss
In our discussions above, we have assumed that the queue is capable of holding an
infinite number of packets. In reality a queue preceding a link has finite capacity,
although the queuing capacity greatly depends on the router design and cost.
Because the queue capacity is finite, packet delays do not really approach infinity as
the traffic intensity approaches 1. Instead, a packet can arrive to find a full queue.
With no place to store such a packet, a router will drop that packet; that is, the
packet will be lost. This overflow at a queue can again be seen in the Java applet for
a queue when the traffic intensity is greater than 1.
From an end-system viewpoint, a packet loss will look like a packet having
been transmitted into the network core but never emerging from the network at the
destination. The fraction of lost packets increases as the traffic intensity increases.
Therefore, performance at a node is often measured not only in terms of delay, but
also in terms of the probability of packet loss. As well discuss in the subsequent
A
ve
ra
g
e
q
u
e
u
in
g
d
e
la
y
La/R
1
Figure 1.18 Dependence of average queuing delay on traffic intensity
chapters, a lost packet may be retransmitted on an end-to-end basis in order to
ensure that all data are eventually transferred from source to destination
1.4.3 End-to-End Delay
Our discussion up to this point has focused on the nodal delay, that is, the delay at a
single router. Lets now consider the total delay from source to destination. To get a
handle on this concept, suppose there are N 1 routers between the source host and
the destination host. Lets also suppose for the moment that the network is uncon-
gested (so that queuing delays are negligible), the processing delay at each router
and at the source host is dproc, the transmission rate out of each router and out of the
source host is R bits/sec, and the propagation on each link is dprop. The nodal delays
accumulate and give an end-to-end delay,
dend-end = N (dproc + dtrans + dprop) (1.2)
where, once again, dtrans = L/R, where L is the packet size. Note that Equation 1.2 is a
generalization of Equation 1.1, which did not take into account processing and propa-
gation delays. We leave it to you to generalize Equation 1.2 to the case of heteroge-
neous delays at the nodes and to the presence of an average queuing delay at each node.
Traceroute
To get a hands-on feel for end-to-end delay in a computer network, we can make use
of the Traceroute program. Traceroute is a simple program that can run in any Inter-
net host. When the user specifies a destination hostname, the program in the source
host sends multiple, special packets toward that destination. As these packets work
their way toward the destination, they pass through a series of routers. When a
router receives one of these special packets, it sends back to the source a short mes-
sage that contains the name and address of the router.
More specifically, suppose there are N 1 routers between the source and the
destination. Then the source will send N special packets into the network, with each
packet addressed to the ultimate destination. These N special packets are marked 1
through N, with the first packet marked 1 and the last packet marked N. When the nth
router receives the nth packet marked n, the router does not forward the packet
toward its destination, but instead sends a message back to the source. When the des-
tination host receives the Nth packet, it too returns a message back to the source. The
source records the time that elapses between when it sends a packet and when it
receives the corresponding return message; it also records the name and address of
the router (or the destination host) that returns the message. In this manner, the source
can reconstruct the route taken by packets flowing from source to destination, and
the source can determine the round-trip delays to all the intervening routers. Trace-
route actually repeats the experiment just described three times, so the source actually
sends 3 N packets to the destination. RFC 1393 describes Traceroute in detail.
42 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET
VideoNote
Using Traceroute to
discover network
paths and measure
network delay
1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 43
Here is an example of the output of the Traceroute program, where the route
was being traced from the source host gaia.cs.umass.edu (at the University of Mass-
achusetts) to the host cis.poly.edu (at Polytechnic University in Brooklyn). The out-
put has six columns: the first column is the n value described above, that is, the
number of the router along the route; the second column is the name of the router;
the third column is the address of the router (of the form xxx.xxx.xxx.xxx); the last
three columns are the round-trip delays for three experiments. If the source receives
fewer than three messages from any given router (due to packet loss in the network),
Traceroute places an asterisk just after the router number and reports fewer than
three round-trip times for that router.
1 cs-gw (128.119.240.254) 1.009 ms 0.899 ms 0.993 ms
2 128.119.3.154 (128.119.3.154) 0.931 ms 0.441 ms 0.651 ms
3 border4-rt-gi-1-3.gw.umass.edu (128.119.2.194) 1.032 ms 0.484 ms 0.451 ms
4 acr1-ge-2-1-0.Boston.cw.net (208.172.51.129) 10.006 ms 8.150 ms 8.460 ms
5 agr4-loopback.NewYork.cw.net (206.24.194.104) 12.272 ms 14.344 ms 13.267 ms
6 acr2-loopback.NewYork.cw.net (206.24.194.62) 13.225 ms 12.292 ms 12.148 ms
7 pos10-2.core2.NewYork1.Level3.net (209.244.160.133) 12.218 ms 11.823 ms 11.793 ms
8 gige9-1-52.hsipaccess1.NewYork1.Level3.net (64.159.17.39) 13.081 ms 11.556 ms 13.297 ms
9 p0-0.polyu.bbnplanet.net (4.25.109.122) 12.716 ms 13.052 ms 12.786 ms
10 cis.poly.edu (128.238.32.126) 14.080 ms 13.035 ms 12.802 ms
In the trace above there are nine routers between the source and the destination.
Most of these routers have a name, and all of them have addresses. For example, the
name of Router 3 is border4-rt-gi-1-3.gw.umass.edu and its address is
128.119.2.194. Looking at the data provided for this same router, we see that
in the first of the three trials the round-trip delay between the source and the router
was 1.03 msec. The round-trip delays for the subsequent two trials were 0.48 and
0.45 msec. These round-trip delays include all of the delays just discussed, includ-
ing transmission delays, propagation delays, router processing delays, and queuing
delays. Because the queuing delay is varying with time, the round-trip delay of
packet n sent to a router n can sometimes be longer than the round-trip delay of
packet n+1 sent to router n+1. Indeed, we observe this phenomenon in the above
example: the delays to Router 6 are larger than the delays to Router 7!
Want to try out Traceroute for yourself? We highly recommended that you visit
http://www.traceroute.org, which provides a Web interface to an extensive list of
sources for route tracing. You choose a source and supply the hostname for any des-
tination. The Traceroute program then does all the work. There are a number of free
software programs that provide a graphical interface to Traceroute; one of our
favorites is PingPlotter [PingPlotter 2012].
End System, Application, and Other Delays
In addition to processing, transmission, and propagation delays, there can be addi-
tional significant delays in the end systems. For example, an end system wanting to
http://www.traceroute.org
transmit a packet into a shared medium (e.g., as in a WiFi or cable modem scenario)
may purposefully delay its transmission as part of its protocol for sharing the
medium with other end systems; well consider such protocols in detail in Chapter
5. Another important delay is media packetization delay, which is present in Voice-
over-IP (VoIP) applications. In VoIP, the sending side must first fill a packet with
encoded digitized speech before passing the packet to the Internet. This time to fill a
packetcalled the packetization delaycan be significant and can impact the user-
perceived quality of a VoIP call. This issue will be further explored in a homework
problem at the end of this chapter.
1.4.4 Throughput in Computer Networks
In addition to delay and packet loss, another critical performance measure in com-
puter networks is end-to-end throughput. To define throughput, consider transfer-
ring a large file from Host A to Host B across a computer network. This transfer
might be, for example, a large video clip from one peer to another in a P2P file
sharing system. The instantaneous throughput at any instant of time is the rate
(in bits/sec) at which Host B is receiving the file. (Many applications, including
many P2P file sharing systems, display the instantaneous throughput during
downloads in the user interfaceperhaps you have observed this before!) If the
file consists of F bits and the transfer takes T seconds for Host B to receive all F
bits, then the average throughput of the file transfer is F/T bits/sec. For some
applications, such as Internet telephony, it is desirable to have a low delay and an
instantaneous throughput consistently above some threshold (for example, over
24 kbps for some Internet telephony applications and over 256 kbps for some real-
time video applications). For other applications, including those involving file
transfers, delay is not critical, but it is desirable to have the highest possible
throughput.
To gain further insight into the important concept of throughput, lets consider a
few examples. Figure 1.19(a) shows two end systems, a server and a client, con-
nected by two communication links and a router. Consider the throughput for a file
transfer from the server to the client. Let Rs denote the rate of the link between
the server and the router; and Rc denote the rate of the link between the router and
the client. Suppose that the only bits being sent in the entire network are those
from the server to the client. We now ask, in this ideal scenario, what is the server-
to-client throughput? To answer this question, we may think of bits as fluid and
communication links as pipes. Clearly, the server cannot pump bits through its link
at a rate faster than Rs bps; and the router cannot forward bits at a rate faster than Rc
bps. If Rs < Rc, then the bits pumped by the server will flow right through therouter and arrive at the client at a rate of Rs bps, giving a throughput of Rs bps. If, onthe other hand, Rc < Rs, then the router will not be able to forward bits as quickly asit receives them. In this case, bits will only leave the router at rate Rc, giving an 44 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET1.4 DELAY, LOSS, AND THROUGHPUT IN PACKET-SWITCHED NETWORKS 45end-to-end throughput of Rc. (Note also that if bits continue to arrive at the router atrate Rs, and continue to leave the router at Rc, the backlog of bits at the router wait-ing for transmission to the client will grow and growa most undesirable situation!)Thus, for this simple two-link network, the throughput is min{Rc, Rs}, that is, it isthe transmission rate of the bottleneck link. Having determined the throughput, wecan now approximate the time it takes to transfer a large file of F bits from server toclient as F/min{Rs, Rc}. For a specific example, suppose you are downloading anMP3 file of F = 32 million bits, the server has a transmission rate of Rs = 2 Mbps,and you have an access link of Rc = 1 Mbps. The time needed to transfer the file isthen 32 seconds. Of course, these expressions for throughput and transfer time areonly approximations, as they do not account for store-and-forward and processingdelays as well as protocol issues. Figure 1.19(b) now shows a network with N links between the server and theclient, with the transmission rates of the N links being R1, R2,…, RN. Applying thesame analysis as for the two-link network, we find that the throughput for a filetransfer from server to client is min{R1, R2,…, RN}, which is once again the trans-mission rate of the bottleneck link along the path between server and client.Now consider another example motivated by todays Internet. Figure 1.20(a)shows two end systems, a server and a client, connected to a computer network.Consider the throughput for a file transfer from the server to the client. The server isconnected to the network with an access link of rate Rs and the client is connected tothe network with an access link of rate Rc. Now suppose that all the links in the coreof the communication network have very high transmission rates, much higher thanRs and Rc. Indeed, today, the core of the Internet is over-provisioned with high speedlinks that experience little congestion. Also suppose that the only bits being sent inthe entire network are those from the server to the client. Because the core of thecomputer network is like a wide pipe in this example, the rate at which bits can flowServerRsR1 R2 RNRcClientServera.b.ClientFigure 1.19 Throughput for a file transfer from server to clientfrom source to destination is again the minimum of Rs and Rc, that is, throughput =min{Rs, Rc}. Therefore, the constraining factor for throughput in todays Internet istypically the access network.For a final example, consider Figure 1.20(b) in which there are 10 servers and10 clients connected to the core of the computer network. In this example, there are10 simultaneous downloads taking place, involving 10 client-server pairs. Supposethat these 10 downloads are the only traffic in the network at the current time. Asshown in the figure, there is a link in the core that is traversed by all 10 downloads.Denote R for the transmission rate of this link R. Lets suppose that all server accesslinks have the same rate Rs, all client access links have the same rate Rc, and thetransmission rates of all the links in the coreexcept the one common link of rateRare much larger than Rs, Rc, and R. Now we ask, what are the throughputs of thedownloads? Clearly, if the rate of the common link, R, is largesay a hundred timeslarger than both Rs and Rcthen the throughput for each download will once againbe min{Rs, Rc}. But what if the rate of the common link is of the same order as Rsand Rc? What will the throughput be in this case? Lets take a look at a specific46 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETServerRsRca. b.Client 10 Clients10 ServersBottlenecklink ofcapacity RFigure 1.20 End-to-end throughput: (a) Client downloads a file fromserver; (b) 10 clients downloading with 10 serversexample. Suppose Rs = 2 Mbps, Rc = 1 Mbps, R = 5 Mbps, and the common linkdivides its transmission rate equally among the 10 downloads. Then the bottleneckfor each download is no longer in the access network, but is now instead the sharedlink in the core, which only provides each download with 500 kbps of throughput.Thus the end-to-end throughput for each download is now reduced to 500 kbps.The examples in Figure 1.19 and Figure 1.20(a) show that throughput dependson the transmission rates of the links over which the data flows. We saw that whenthere is no other intervening traffic, the throughput can simply be approximated asthe minimum transmission rate along the path between source and destination. Theexample in Figure 1.20(b) shows that more generally the throughput depends notonly on the transmission rates of the links along the path, but also on the interveningtraffic. In particular, a link with a high transmission rate may nonetheless be the bot-tleneck link for a file transfer if many other data flows are also passing through thatlink. We will examine throughput in computer networks more closely in the home-work problems and in the subsequent chapters.1.5 Protocol Layers and Their Service ModelsFrom our discussion thus far, it is apparent that the Internet is an extremely compli-cated system. We have seen that there are many pieces to the Internet: numerousapplications and protocols, various types of end systems, packet switches, and vari-ous types of link-level media. Given this enormous complexity, is there any hope oforganizing a network architecture, or at least our discussion of network architecture?Fortunately, the answer to both questions is yes.1.5.1 Layered ArchitectureBefore attempting to organize our thoughts on Internet architecture, lets look for ahuman analogy. Actually, we deal with complex systems all the time in our every-day life. Imagine if someone asked you to describe, for example, the airline system.How would you find the structure to describe this complex system that has ticketingagents, baggage checkers, gate personnel, pilots, airplanes, air traffic control, and aworldwide system for routing airplanes? One way to describe this system might beto describe the series of actions you take (or others take for you) when you fly on anairline. You purchase your ticket, check your bags, go to the gate, and eventually getloaded onto the plane. The plane takes off and is routed to its destination. After yourplane lands, you deplane at the gate and claim your bags. If the trip was bad, youcomplain about the flight to the ticket agent (getting nothing for your effort). Thisscenario is shown in Figure 1.21.Already, we can see some analogies here with computer networking: You arebeing shipped from source to destination by the airline; a packet is shipped from1.5 PROTOCOL LAYERS AND THEIR SERVICE MODELS 47source host to destination host in the Internet. But this is not quite the analogy weare after. We are looking for some structure in Figure 1.21. Looking at Figure 1.21,we note that there is a ticketing function at each end; there is also a baggage func-tion for already-ticketed passengers, and a gate function for already-ticketed andalready-baggage-checked passengers. For passengers who have made it through thegate (that is, passengers who are already ticketed, baggage-checked, and through thegate), there is a takeoff and landing function, and while in flight, there is an airplane-routing function. This suggests that we can look at the functionality in Figure 1.21in a horizontal manner, as shown in Figure 1.22.Figure 1.22 has divided the airline functionality into layers, providing a frame-work in which we can discuss airline travel. Note that each layer, combined with the48 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETTicket (purchase)Baggage (check)Gates (load)Runway takeoffAirplane routing Airplane routing Airplane routingTicket (complain)Baggage (claim)Gates (unload)Runway landingAirplane routingTicketBaggageGateTakeoff/LandingDeparture airport Intermediate air-trafficcontrol centersFigure 1.22 Horizontal layering of airline functionalityTicket (purchase)Baggage (check)Gates (load)Runway takeoffAirplane routingTicket (complain)Baggage (claim)Gates (unload)Runway landingAirplane routingAirplane routingFigure 1.21 Taking an airplane trip: actions1.5 PROTOCOL LAYERS AND THEIR SERVICE MODELS 49layers below it, implements some functionality, some service. At the ticketing layerand below, airline-counter-to-airline-counter transfer of a person is accomplished.At the baggage layer and below, baggage-check-to-baggage-claim transfer of a per-son and bags is accomplished. Note that the baggage layer provides this service onlyto an already-ticketed person. At the gate layer, departure-gate-to-arrival-gate trans-fer of a person and bags is accomplished. At the takeoff/landing layer, runway-to-runway transfer of people and their bags is accomplished. Each layer provides itsservice by (1) performing certain actions within that layer (for example, at the gatelayer, loading and unloading people from an airplane) and by (2) using the servicesof the layer directly below it (for example, in the gate layer, using the runway-to-runway passenger transfer service of the takeoff/landing layer).A layered architecture allows us to discuss a well-defined, specific part of alarge and complex system. This simplification itself is of considerable value by pro-viding modularity, making it much easier to change the implementation of the serv-ice provided by the layer. As long as the layer provides the same service to the layerabove it, and uses the same services from the layer below it, the remainder of thesystem remains unchanged when a layers implementation is changed. (Note thatchanging the implementation of a service is very different from changing the serv-ice itself!) For example, if the gate functions were changed (for instance, to havepeople board and disembark by height), the remainder of the airline system wouldremain unchanged since the gate layer still provides the same function (loading andunloading people); it simply implements that function in a different manner after thechange. For large and complex systems that are constantly being updated, the abilityto change the implementation of a service without affecting other components of thesystem is another important advantage of layering.Protocol LayeringBut enough about airlines. Lets now turn our attention to network protocols. Toprovide structure to the design of network protocols, network designers organizeprotocolsand the network hardware and software that implement the protocolsin layers. Each protocol belongs to one of the layers, just as each function in theairline architecture in Figure 1.22 belonged to a layer. We are again interested inthe services that a layer offers to the layer abovethe so-called service model ofa layer. Just as in the case of our airline example, each layer provides its serviceby (1) performing certain actions within that layer and by (2) using the services ofthe layer directly below it. For example, the services provided by layer n mayinclude reliable delivery of messages from one edge of the network to the other.This might be implemented by using an unreliable edge-to-edge message deliveryservice of layer n 1, and adding layer n functionality to detect and retransmitlost messages.A protocol layer can be implemented in software, in hardware, or in a combina-tion of the two. Application-layer protocolssuch as HTTP and SMTPare almostalways implemented in software in the end systems; so are transport-layer protocols.Because the physical layer and data link layers are responsible for handling commu-nication over a specific link, they are typically implemented in a network interfacecard (for example, Ethernet or WiFi interface cards) associated with a given link.The network layer is often a mixed implementation of hardware and software. Alsonote that just as the functions in the layered airline architecture were distributedamong the various airports and flight control centers that make up the system, so toois a layer n protocol distributed among the end systems, packet switches, and othercomponents that make up the network. That is, theres often a piece of a layer n pro-tocol in each of these network components.Protocol layering has conceptual and structural advantages [RFC 3439]. As wehave seen, layering provides a structured way to discuss system components. Mod-ularity makes it easier to update system components. We mention, however, thatsome researchers and networking engineers are vehemently opposed to layering[Wakeman 1992]. One potential drawback of layering is that one layer may dupli-cate lower-layer functionality. For example, many protocol stacks provide errorrecovery on both a per-link basis and an end-to-end basis. A second potential draw-back is that functionality at one layer may need information (for example, a time-stamp value) that is present only in another layer; this violates the goal ofseparation of layers.When taken together, the protocols of the various layers are called the protocolstack. The Internet protocol stack consists of five layers: the physical, link, network,transport, and application layers, as shown in Figure 1.23(a). If you examine theTable of Contents, you will see that we have roughly organized this book using thelayers of the Internet protocol stack. We take a top-down approach, first coveringthe application layer and then proceeding downward.50 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETTransportApplicationNetworkLinkPhysicala.Five-layer Internet protocol stackTransportSessionApplicationPresentationNetworkLinkPhysicalb.Seven-layer ISO OSI reference modelFigure 1.23 The Internet protocol stack (a) and OSI reference model (b)1.5 PROTOCOL LAYERS AND THEIR SERVICE MODELS 51Application LayerThe application layer is where network applications and their application-layer proto-cols reside. The Internets application layer includes many protocols, such as the HTTPprotocol (which provides for Web document request and transfer), SMTP (which pro-vides for the transfer of e-mail messages), and FTP (which provides for the transfer offiles between two end systems). Well see that certain network functions, such as thetranslation of human-friendly names for Internet end systems like www.ietf.org to a32-bit network address, are also done with the help of a specific application-layer pro-tocol, namely, the domain name system (DNS). Well see in Chapter 2 that it is veryeasy to create and deploy our own new application-layer protocols.An application-layer protocol is distributed over multiple end systems, with theapplication in one end system using the protocol to exchange packets of informationwith the application in another end system. Well refer to this packet of informationat the application layer as a message.Transport LayerThe Internets transport layer transports application-layer messages betweenapplication endpoints. In the Internet there are two transport protocols, TCP andUDP, either of which can transport application-layer messages. TCP provides aconnection-oriented service to its applications. This service includes guaranteeddelivery of application-layer messages to the destination and flow control (that is,sender/receiver speed matching). TCP also breaks long messages into shorter seg-ments and provides a congestion-control mechanism, so that a source throttles itstransmission rate when the network is congested. The UDP protocol provides a con-nectionless service to its applications. This is a no-frills service that provides noreliability, no flow control, and no congestion control. In this book, well refer to atransport-layer packet as a segment.Network LayerThe Internets network layer is responsible for moving network-layer packetsknown as datagrams from one host to another. The Internet transport-layer proto-col (TCP or UDP) in a source host passes a transport-layer segment and a destina-tion address to the network layer, just as you would give the postal service a letterwith a destination address. The network layer then provides the service of deliver-ing the segment to the transport layer in the destination host.The Internets network layer includes the celebrated IP Protocol, which definesthe fields in the datagram as well as how the end systems and routers act on thesefields. There is only one IP protocol, and all Internet components that have a net-work layer must run the IP protocol. The Internets network layer also contains rout-ing protocols that determine the routes that datagrams take between sources andwww.ietf.orgdestinations. The Internet has many routing protocols. As we saw in Section 1.3, theInternet is a network of networks, and within a network, the network administratorcan run any routing protocol desired. Although the network layer contains both theIP protocol and numerous routing protocols, it is often simply referred to as the IPlayer, reflecting the fact that IP is the glue that binds the Internet together.Link LayerThe Internets network layer routes a datagram through a series of routers betweenthe source and destination. To move a packet from one node (host or router) to thenext node in the route, the network layer relies on the services of the link layer. Inparticular, at each node, the network layer passes the datagram down to the linklayer, which delivers the datagram to the next node along the route. At this nextnode, the link layer passes the datagram up to the network layer.The services provided by the link layer depend on the specific link-layer proto-col that is employed over the link. For example, some link-layer protocols providereliable delivery, from transmitting node, over one link, to receiving node. Note thatthis reliable delivery service is different from the reliable delivery service of TCP,which provides reliable delivery from one end system to another. Examples of link-layer protocols include Ethernet, WiFi, and the cable access networks DOCSIS pro-tocol. As datagrams typically need to traverse several links to travel from source todestination, a datagram may be handled by different link-layer protocols at differentlinks along its route. For example, a datagram may be handled by Ethernet on onelink and by PPP on the next link. The network layer will receive a different servicefrom each of the different link-layer protocols. In this book, well refer to the link-layer packets as frames.Physical LayerWhile the job of the link layer is to move entire frames from one network elementto an adjacent network element, the job of the physical layer is to move the individ-ual bits within the frame from one node to the next. The protocols in this layer areagain link dependent and further depend on the actual transmission medium of thelink (for example, twisted-pair copper wire, single-mode fiber optics). For example,Ethernet has many physical-layer protocols: one for twisted-pair copper wire,another for coaxial cable, another for fiber, and so on. In each case, a bit is movedacross the link in a different way.The OSI ModelHaving discussed the Internet protocol stack in detail, we should mention that it isnot the only protocol stack around. In particular, back in the late 1970s, the Interna-tional Organization for Standardization (ISO) proposed that computer networks be52 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETorganized around seven layers, called the Open Systems Interconnection (OSI)model [ISO 2012]. The OSI model took shape when the protocols that were tobecome the Internet protocols were in their infancy, and were but one of many dif-ferent protocol suites under development; in fact, the inventors of the original OSImodel probably did not have the Internet in mind when creating it. Nevertheless,beginning in the late 1970s, many training and university courses picked up on theISO mandate and organized courses around the seven-layer model. Because of itsearly impact on networking education, the seven-layer model continues to linger onin some networking textbooks and training courses.The seven layers of the OSI reference model, shown in Figure 1.23(b), are:application layer, presentation layer, session layer, transport layer, network layer,data link layer, and physical layer. The functionality of five of these layers isroughly the same as their similarly named Internet counterparts. Thus, lets considerthe two additional layers present in the OSI reference modelthe presentation layerand the session layer. The role of the presentation layer is to provide services thatallow communicating applications to interpret the meaning of data exchanged.These services include data compression and data encryption (which are self-explanatory) as well as data description (which, as we will see in Chapter 9, freesthe applications from having to worry about the internal format in which data arerepresented/storedformats that may differ from one computer to another). Thesession layer provides for delimiting and synchronization of data exchange, includ-ing the means to build a checkpointing and recovery scheme.The fact that the Internet lacks two layers found in the OSI reference modelposes a couple of interesting questions: Are the services provided by these layersunimportant? What if an application needs one of these services? The Internetsanswer to both of these questions is the sameits up to the application developer.Its up to the application developer to decide if a service is important, and if theservice is important, its up to the application developer to build that functionalityinto the application.1.5.2 EncapsulationFigure 1.24 shows the physical path that data takes down a sending end systemsprotocol stack, up and down the protocol stacks of an intervening link-layer switchand router, and then up the protocol stack at the receiving end system. As we discusslater in this book, routers and link-layer switches are both packet switches. Similarto end systems, routers and link-layer switches organize their networking hardwareand software into layers. But routers and link-layer switches do not implement all ofthe layers in the protocol stack; they typically implement only the bottom layers. Asshown in Figure 1.24, link-layer switches implement layers 1 and 2; routers imple-ment layers 1 through 3. This means, for example, that Internet routers are capableof implementing the IP protocol (a layer 3 protocol), while link-layer switches arenot. Well see later that while link-layer switches do not recognize IP addresses, they1.5 PROTOCOL LAYERS AND THEIR SERVICE MODELS 5354 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETare capable of recognizing layer 2 addresses, such as Ethernet addresses. Note thathosts implement all five layers; this is consistent with the view that the Internetarchitecture puts much of its complexity at the edges of the network.Figure 1.24 also illustrates the important concept of encapsulation. At thesending host, an application-layer message (M in Figure 1.24) is passed to thetransport layer. In the simplest case, the transport layer takes the message andappends additional information (so-called transport-layer header information, Htin Figure 1.24) that will be used by the receiver-side transport layer. The applica-tion-layer message and the transport-layer header information together constitutethe transport-layer segment. The transport-layer segment thus encapsulates theapplication-layer message. The added information might include informationallowing the receiver-side transport layer to deliver the message up to the appro-priate application, and error-detection bits that allow the receiver to determinewhether bits in the message have been changed in route. The transport layer thenpasses the segment to the network layer, which adds network-layer header infor-mation (Hn in Figure 1.24) such as source and destination end system addresses,MMMMHtHtHtHnHnHlHtHnHlLink-layer switchRouterApplicationTransportNetworkLinkPhysicalMessageSegmentDatagramFrameMMMMHtHtHtHnHnHlLinkPhysicalSourceNetworkLinkPhysicalDestinationApplicationTransportNetworkLinkPhysicalM HtHnHl MHtHn M HtHn MHtHnHl M HtHnHl MFigure 1.24 Hosts, routers, and link-layer switches; each contains adifferent set of layers, reflecting their differences infunctionalitycreating a network-layer datagram. The datagram is then passed to the linklayer, which (of course!) will add its own link-layer header information and createa link-layer frame. Thus, we see that at each layer, a packet has two types offields: header fields and a payload field. The payload is typically a packet fromthe layer above.A useful analogy here is the sending of an interoffice memo from one corpo-rate branch office to another via the public postal service. Suppose Alice, who isin one branch office, wants to send a memo to Bob, who is in another branchoffice. The memo is analogous to the application-layer message. Alice puts thememo in an interoffice envelope with Bobs name and department written on thefront of the envelope. The interoffice envelope is analogous to a transport-layersegmentit contains header information (Bobs name and department number)and it encapsulates the application-layer message (the memo). When the sendingbranch-office mailroom receives the interoffice envelope, it puts the interofficeenvelope inside yet another envelope, which is suitable for sending through thepublic postal service. The sending mailroom also writes the postal address of thesending and receiving branch offices on the postal envelope. Here, the postalenvelope is analogous to the datagramit encapsulates the transport-layer seg-ment (the interoffice envelope), which encapsulates the original message (thememo). The postal service delivers the postal envelope to the receiving branch-office mailroom. There, the process of de-encapsulation is begun. The mailroomextracts the interoffice memo and forwards it to Bob. Finally, Bob opens the enve-lope and removes the memo.The process of encapsulation can be more complex than that described above.For example, a large message may be divided into multiple transport-layer segments(which might themselves each be divided into multiple network-layer datagrams).At the receiving end, such a segment must then be reconstructed from its constituentdatagrams.1.6 Networks Under AttackThe Internet has become mission critical for many institutions today, including largeand small companies, universities, and government agencies. Many individuals alsorely on the Internet for many of their professional, social, and personal activities.But behind all this utility and excitement, there is a dark side, a side where badguys attempt to wreak havoc in our daily lives by damaging our Internet-connectedcomputers, violating our privacy, and rendering inoperable the Internet services onwhich we depend.The field of network security is about how the bad guys can attack computernetworks and about how we, soon-to-be experts in computer networking, can1.6 NETWORKS UNDER ATTACK 5556 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETdefend networks against those attacks, or better yet, design new architectures thatare immune to such attacks in the first place. Given the frequency and variety ofexisting attacks as well as the threat of new and more destructive future attacks,network security has become a central topic in the field of computer networking.One of the features of this textbook is that it brings network security issues to theforefront.Since we dont yet have expertise in computer networking and Internet proto-cols, well begin here by surveying some of todays more prevalent security-related problems. This will whet our appetite for more substantial discussions inthe upcoming chapters. So we begin here by simply asking, what can go wrong?How are computer networks vulnerable? What are some of the more prevalenttypes of attacks today?The bad guys can put malware into your host via the InternetWe attach devices to the Internet because we want to receive/send data from/tothe Internet. This includes all kinds of good stuff, including Web pages, e-mailmessages, MP3s, telephone calls, live video, search engine results, and so on.But, unfortunately, along with all that good stuff comes malicious stuffcollec-tively known as malwarethat can also enter and infect our devices. Once mal-ware infects our device it can do all kinds of devious things, including deletingour files; installing spyware that collects our private information, such as socialsecurity numbers, passwords, and keystrokes, and then sends this (over the Inter-net, of course!) back to the bad guys. Our compromised host may also beenrolled in a network of thousands of similarly compromised devices, collec-tively known as a botnet, which the bad guys control and leverage for spam e-mail distribution or distributed denial-of-service attacks (soon to be discussed)against targeted hosts.Much of the malware out there today is self-replicating: once it infects onehost, from that host it seeks entry into other hosts over the Internet, and from thenewly infected hosts, it seeks entry into yet more hosts. In this manner, self-replicating malware can spread exponentially fast. Malware can spread in theform of a virus or a worm. Viruses are malware that require some form of userinteraction to infect the users device. The classic example is an e-mail attachmentcontaining malicious executable code. If a user receives and opens such an attach-ment, the user inadvertently runs the malware on the device. Typically, such e-mail viruses are self-replicating: once executed, the virus may send an identicalmessage with an identical malicious attachment to, for example, every recipient inthe users address book. Worms are malware that can enter a device without anyexplicit user interaction. For example, a user may be running a vulnerable networkapplication to which an attacker can send malware. In some cases, without anyuser intervention, the application may accept the malware from the Internet andrun it, creating a worm. The worm in the newly infected device then scans theInternet, searching for other hosts running the same vulnerable network applica-tion. When it finds other vulnerable hosts, it sends a copy of itself to those hosts.Today, malware, is pervasive and costly to defend against. As you work throughthis textbook, we encourage you to think about the following question: What cancomputer network designers do to defend Internet-attached devices from malwareattacks?The bad guys can attack servers and network infrastructureAnother broad class of security threats are known as denial-of-service (DoS)attacks. As the name suggests, a DoS attack renders a network, host, or other pieceof infrastructure unusable by legitimate users. Web servers, e-mail servers, DNSservers (discussed in Chapter 2), and institutional networks can all be subject to DoSattacks. Internet DoS attacks are extremely common, with thousands of DoS attacksoccurring every year [Moore 2001; Mirkovic 2005]. Most Internet DoS attacks fallinto one of three categories: Vulnerability attack. This involves sending a few well-crafted messages to a vul-nerable application or operating system running on a targeted host. If the rightsequence of packets is sent to a vulnerable application or operating system, theservice can stop or, worse, the host can crash. Bandwidth flooding. The attacker sends a deluge of packets to the targetedhostso many packets that the targets access link becomes clogged, preventinglegitimate packets from reaching the server. Connection flooding. The attacker establishes a large number of half-open orfully open TCP connections (TCP connections are discussed in Chapter 3) at thetarget host. The host can become so bogged down with these bogus connectionsthat it stops accepting legitimate connections.Lets now explore the bandwidth-flooding attack in more detail. Recalling ourdelay and loss analysis discussion in Section 1.4.2, its evident that if the server has anaccess rate of R bps, then the attacker will need to send traffic at a rate of approxi-mately R bps to cause damage. If R is very large, a single attack source may not beable to generate enough traffic to harm the server. Furthermore, if all the trafficemanates from a single source, an upstream router may be able to detect the attack andblock all traffic from that source before the traffic gets near the server. In a distributedDoS (DDoS) attack, illustrated in Figure 1.25, the attacker controls multiple sourcesand has each source blast traffic at the target. With this approach, the aggregate trafficrate across all the controlled sources needs to be approximately R to cripple the1.6 NETWORKS UNDER ATTACK 57service. DDoS attacks leveraging botnets with thousands of comprised hosts are acommon occurrence today [Mirkovic 2005]. DDos attacks are much harder to detectand defend against than a DoS attack from a single host.We encourage you to consider the following question as you work your waythrough this book: What can computer network designers do to defend against DoSattacks? We will see that different defenses are needed for the three types of DoSattacks.The bad guys can sniff packetsMany users today access the Internet via wireless devices, such as WiFi-connectedlaptops or handheld devices with cellular Internet connections (covered in Chapter6). While ubiquitous Internet access is extremely convenient and enables marvelousnew applications for mobile users, it also creates a major security vulnerabilitybyplacing a passive receiver in the vicinity of the wireless transmitter, that receiver canobtain a copy of every packet that is transmitted! These packets can contain all kindsof sensitive information, including passwords, social security numbers, tradesecrets, and private personal messages. A passive receiver that records a copy ofevery packet that flies by is called a packet sniffer.Sniffers can be deployed in wired environments as well. In wired broadcast envi-ronments, as in many Ethernet LANs, a packet sniffer can obtain copies of broadcastpackets sent over the LAN. As described in Section 1.2, cable access technologiesalso broadcast packets and are thus vulnerable to sniffing. Furthermore, a bad guywho gains access to an institutions access router or access link to the Internet may58 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETAttackerstartattackSlaveSlaveSlaveVictimSlaveSlaveFigure 1.25 A distributed denial-of-service attack1.6 NETWORKS UNDER ATTACK 59be able to plant a sniffer that makes a copy of every packet going to/from the organi-zation. Sniffed packets can then be analyzed offline for sensitive information.Packet-sniffing software is freely available at various Web sites and as commercialproducts. Professors teaching a networking course have been known to assign lab exer-cises that involve writing a packet-sniffing and application-layer data reconstructionprogram. Indeed, the Wireshark [Wireshark 2012] labs associated with this text (see theintroductory Wireshark lab at the end of this chapter) use exactly such a packet sniffer!Because packet sniffers are passivethat is, they do not inject packets into thechannelthey are difficult to detect. So, when we send packets into a wireless chan-nel, we must accept the possibility that some bad guy may be recording copies ofour packets. As you may have guessed, some of the best defenses against packetsniffing involve cryptography. We will examine cryptography as it applies to net-work security in Chapter 8.The bad guys can masquerade as someone you trustIt is surprisingly easy (you will have the knowledge to do so shortly as you proceedthrough this text!) to create a packet with an arbitrary source address, packet con-tent, and destination address and then transmit this hand-crafted packet into theInternet, which will dutifully forward the packet to its destination. Imagine theunsuspecting receiver (say an Internet router) who receives such a packet, takes the(false) source address as being truthful, and then performs some command embed-ded in the packets contents (say modifies its forwarding table). The ability to injectpackets into the Internet with a false source address is known as IP spoofing, and isbut one of many ways in which one user can masquerade as another user.To solve this problem, we will need end-point authentication, that is, a mecha-nism that will allow us to determine with certainty if a message originates fromwhere we think it does. Once again, we encourage you to think about how this canbe done for network applications and protocols as you progress through the chaptersof this book. We will explore mechanisms for end-point authentication in Chapter 8.In closing this section, its worth considering how the Internet got to be such aninsecure place in the first place. The answer, in essence, is that the Internet was orig-inally designed to be that way, based on the model of a group of mutually trustingusers attached to a transparent network [Blumenthal 2001]a model in which (bydefinition) there is no need for security. Many aspects of the original Internet archi-tecture deeply reflect this notion of mutual trust. For example, the ability for oneuser to send a packet to any other user is the default rather than a requested/grantedcapability, and user identity is taken at declared face value, rather than being authen-ticated by default.But todays Internet certainly does not involve mutually trusting users.Nonetheless, todays users still need to communicate when they dont necessarilytrust each other, may wish to communicate anonymously, may communicate indi-rectly through third parties (e.g., Web caches, which well study in Chapter 2, or60 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETmobility-assisting agents, which well study in Chapter 6), and may distrust thehardware, software, and even the air through which they communicate. We nowhave many security-related challenges before us as we progress through this book:We should seek defenses against sniffing, end-point masquerading, man-in-the-middle attacks, DDoS attacks, malware, and more. We should keep in mind thatcommunication among mutually trusted users is the exception rather than the rule.Welcome to the world of modern computer networking!1.7 History of Computer Networking and the InternetSections 1.1 through 1.6 presented an overview of the technology of computer net-working and the Internet. You should know enough now to impress your family andfriends! However, if you really want to be a big hit at the next cocktail party, youshould sprinkle your discourse with tidbits about the fascinating history of the Inter-net [Segaller 1998].1.7.1 The Development of Packet Switching: 19611972The field of computer networking and todays Internet trace their beginningsback to the early 1960s, when the telephone network was the worlds dominantcommunication network. Recall from Section 1.3 that the telephone network usescircuit switching to transmit information from a sender to a receiveran appro-priate choice given that voice is transmitted at a constant rate between senderand receiver. Given the increasing importance of computers in the early 1960sand the advent of timeshared computers, it was perhaps natural to consider howto hook computers together so that they could be shared among geographicallydistributed users. The traffic generated by such users was likely to be burstyintervals of activity, such as the sending of a command to a remote computer, fol-lowed by periods of inactivity while waiting for a reply or while contemplatingthe received response.Three research groups around the world, each unaware of the others work[Leiner 1998], began inventing packet switching as an efficient and robust alterna-tive to circuit switching. The first published work on packet-switching techniqueswas that of Leonard Kleinrock [Kleinrock 1961; Kleinrock 1964], then a graduatestudent at MIT. Using queuing theory, Kleinrocks work elegantly demonstrated theeffectiveness of the packet-switching approach for bursty traffic sources. In 1964,Paul Baran [Baran 1964] at the Rand Institute had begun investigating the use ofpacket switching for secure voice over military networks, and at the National Physi-cal Laboratory in England, Donald Davies and Roger Scantlebury were also devel-oping their ideas on packet switching.The work at MIT, Rand, and the NPL laid the foundations for todays Inter-net. But the Internet also has a long history of a lets-build-it-and-demonstrate-itattitude that also dates back to the 1960s. J. C. R. Licklider [DEC 1990] andLawrence Roberts, both colleagues of Kleinrocks at MIT, went on to lead thecomputer science program at the Advanced Research Projects Agency (ARPA) inthe United States. Roberts published an overall plan for the ARPAnet [Roberts1967], the first packet-switched computer network and a direct ancestor of todayspublic Internet. On Labor Day in 1969, the first packet switch was installed atUCLA under Kleinrocks supervision, and three additional packet switches wereinstalled shortly thereafter at the Stanford Research Institute (SRI), UC Santa Bar-bara, and the University of Utah (Figure 1.26). The fledgling precursor to the1.7 HISTORY OF COMPUTER NETWORKING AND THE INTERNET 61Figure 1.26 An early packet switch62 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETInternet was four nodes large by the end of 1969. Kleinrock recalls the very firstuse of the network to perform a remote login from UCLA to SRI, crashing the sys-tem [Kleinrock 2004].By 1972, ARPAnet had grown to approximately 15 nodes and was given its firstpublic demonstration by Robert Kahn. The first host-to-host protocol betweenARPAnet end systems, known as the network-control protocol (NCP), was com-pleted [RFC 001]. With an end-to-end protocol available, applications could now bewritten. Ray Tomlinson wrote the first e-mail program in 1972.1.7.2 Proprietary Networks and Internetworking: 19721980The initial ARPAnet was a single, closed network. In order to communicate with anARPAnet host, one had to be actually attached to another ARPAnet IMP. In the earlyto mid-1970s, additional stand-alone packet-switching networks besides ARPAnetcame into being: ALOHANet, a microwave network linking universities on theHawaiian islands [Abramson 1970], as well as DARPAs packet-satellite [RFC 829]and packet-radio networks [Kahn 1978]; Telenet, a BBN commercial packet-switching network based on ARPAnet technology; Cyclades, a French packet-switching network pioneered by Louis Pouzin [Think 2012]; Time-sharing networkssuch as Tymnet and the GE Information Services network, among others, in the late1960s and early 1970s [Schwartz 1977]; IBMs SNA (19691974), which paral-leled the ARPAnet work [Schwartz 1977].The number of networks was growing. With perfect hindsight we can see thatthe time was ripe for developing an encompassing architecture for connecting net-works together. Pioneering work on interconnecting networks (under the sponsor-ship of the Defense Advanced Research Projects Agency (DARPA)), in essencecreating a network of networks, was done by Vinton Cerf and Robert Kahn [Cerf1974]; the term internetting was coined to describe this work.These architectural principles were embodied in TCP. The early versions ofTCP, however, were quite different from todays TCP. The early versions of TCPcombined a reliable in-sequence delivery of data via end-system retransmission(still part of todays TCP) with forwarding functions (which today are performedby IP). Early experimentation with TCP, combined with the recognition of theimportance of an unreliable, non-flow-controlled, end-to-end transport servicefor applications such as packetized voice, led to the separation of IP out of TCPand the development of the UDP protocol. The three key Internet protocols thatwe see todayTCP, UDP, and IPwere conceptually in place by the end of the1970s.In addition to the DARPA Internet-related research, many other importantnetworking activities were underway. In Hawaii, Norman Abramson was devel-oping ALOHAnet, a packet-based radio network that allowed multiple remotesites on the Hawaiian Islands to communicate with each other. The ALOHA protocol[Abramson 1970] was the first multiple-access protocol, allowing geographicallydistributed users to share a single broadcast communication medium (a radio fre-quency). Metcalfe and Boggs built on Abramsons multiple-access protocol workwhen they developed the Ethernet protocol [Metcalfe 1976] for wire-basedshared broadcast networks. Interestingly, Metcalfe and Boggs Ethernet protocolwas motivated by the need to connect multiple PCs, printers, and shared disks[Perkins 1994]. Twenty-five years ago, well before the PC revolution and theexplosion of networks, Metcalfe and Boggs were laying the foundation fortodays PC LANs.1.7.3 A Proliferation of Networks: 19801990By the end of the 1970s, approximately two hundred hosts were connected to theARPAnet. By the end of the 1980s the number of hosts connected to the publicInternet, a confederation of networks looking much like todays Internet, wouldreach a hundred thousand. The 1980s would be a time of tremendous growth.Much of that growth resulted from several distinct efforts to create computernetworks linking universities together. BITNET provided e-mail and file transfersamong several universities in the Northeast. CSNET (computer science network)was formed to link university researchers who did not have access to ARPAnet. In1986, NSFNET was created to provide access to NSF-sponsored supercomputingcenters. Starting with an initial backbone speed of 56 kbps, NSFNETs backbonewould be running at 1.5 Mbps by the end of the decade and would serve as a pri-mary backbone linking regional networks.In the ARPAnet community, many of the final pieces of todays Internet archi-tecture were falling into place. January 1, 1983 saw the official deployment ofTCP/IP as the new standard host protocol for ARPAnet (replacing the NCP proto-col). The transition [RFC 801] from NCP to TCP/IP was a flag day eventallhosts were required to transfer over to TCP/IP as of that day. In the late 1980s,important extensions were made to TCP to implement host-based congestion con-trol [Jacobson 1988]. The DNS, used to map between a human-readable Internetname (for example, gaia.cs.umass.edu) and its 32-bit IP address, was also devel-oped [RFC 1034].Paralleling this development of the ARPAnet (which was for the most part aUS effort), in the early 1980s the French launched the Minitel project, an ambi-tious plan to bring data networking into everyones home. Sponsored by theFrench government, the Minitel system consisted of a public packet-switched net-work (based on the X.25 protocol suite), Minitel servers, and inexpensive termi-nals with built-in low-speed modems. The Minitel became a huge success in 1984when the French government gave away a free Minitel terminal to each Frenchhousehold that wanted one. Minitel sites included free sitessuch as a telephonedirectory siteas well as private sites, which collected a usage-based fee from1.7 HISTORY OF COMPUTER NETWORKING AND THE INTERNET 63each user. At its peak in the mid 1990s, it offered more than 20,000 services, rang-ing from home banking to specialized research databases. The Minitel was in alarge proportion of French homes 10 years before most Americans had ever heardof the Internet.1.7.4 The Internet Explosion: The 1990sThe 1990s were ushered in with a number of events that symbolized the continuedevolution and the soon-to-arrive commercialization of the Internet. ARPAnet, theprogenitor of the Internet, ceased to exist. In 1991, NSFNET lifted its restrictions onthe use of NSFNET for commercial purposes. NSFNET itself would be decommis-sioned in 1995, with Internet backbone traffic being carried by commercial InternetService Providers.The main event of the 1990s was to be the emergence of the World Wide Webapplication, which brought the Internet into the homes and businesses of millions ofpeople worldwide. The Web served as a platform for enabling and deploying hun-dreds of new applications that we take for granted today, including search (e.g.,Google and Bing) Internet commerce (e.g., Amazon and eBay) and social networks(e.g., Facebook).The Web was invented at CERN by Tim Berners-Lee between 1989 and 1991[Berners-Lee 1989], based on ideas originating in earlier work on hypertext fromthe 1940s by Vannevar Bush [Bush 1945] and since the 1960s by Ted Nelson[Xanadu 2012]. Berners-Lee and his associates developed initial versions of HTML,HTTP, a Web server, and a browserthe four key components of the Web. Aroundthe end of 1993 there were about two hundred Web servers in operation, this collec-tion of servers being just a harbinger of what was about to come. At about this timeseveral researchers were developing Web browsers with GUI interfaces, includingMarc Andreessen, who along with Jim Clark, formed Mosaic Communications,which later became Netscape Communications Corporation [Cusumano 1998; Quittner1998]. By 1995, university students were using Netscape browsers to surf the Webon a daily basis. At about this time companiesbig and smallbegan to operateWeb servers and transact commerce over the Web. In 1996, Microsoft started tomake browsers, which started the browser war between Netscape and Microsoft,which Microsoft won a few years later [Cusumano 1998].The second half of the 1990s was a period of tremendous growth and innova-tion for the Internet, with major corporations and thousands of startups creatingInternet products and services. By the end of the millennium the Internet was sup-porting hundreds of popular applications, including four killer applications: E-mail, including attachments and Web-accessible e-mail The Web, including Web browsing and Internet commerce64 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET Instant messaging, with contact lists Peer-to-peer file sharing of MP3s, pioneered by NapsterInterestingly, the first two killer applications came from the research community,whereas the last two were created by a few young entrepreneurs.The period from 1995 to 2001 was a roller-coaster ride for the Internet in thefinancial markets. Before they were even profitable, hundreds of Internet startupsmade initial public offerings and started to be traded in a stock market. Many com-panies were valued in the billions of dollars without having any significant revenuestreams. The Internet stocks collapsed in 20002001, and many startups shut down.Nevertheless, a number of companies emerged as big winners in the Internet space,including Microsoft, Cisco, Yahoo, e-Bay, Google, and Amazon.1.7.5 The New MillenniumInnovation in computer networking continues at a rapid pace. Advances are beingmade on all fronts, including deployments of faster routers and higher transmissionspeeds in both access networks and in network backbones. But the following devel-opments merit special attention: Since the beginning of the millennium, we have been seeing aggressive deploy-ment of broadband Internet access to homesnot only cable modems and DSLbut also fiber to the home, as discussed in Section 1.2. This high-speed Internetaccess has set the stage for a wealth of video applications, including the distribu-tion of user-generated video (for example, YouTube), on-demand streaming ofmovies and television shows (e.g., Netflix) , and multi-person video conference(e.g., Skype). The increasing ubiquity of high-speed (54 Mbps and higher) public WiFi net-works and medium-speed (up to a few Mbps) Internet access via 3G and 4Gcellular telephony networks is not only making it possible to remain con-stantly connected while on the move, but also enabling new location-specificapplications. The number of wireless devices connecting to the Internet sur-passed the number of wired devices in 2011. This high-speed wireless accesshas set the stage for the rapid emergence of hand-held computers (iPhones,Androids, iPads, and so on), which enjoy constant and untethered access tothe Internet. Online social networks, such as Facebook and Twitter, have created massive peo-ple networks on top of the Internet. Many Internet users today live primarilywithin Facebook. Through their APIs, the online social networks create plat-forms for new networked applications and distributed games.1.7 HISTORY OF COMPUTER NETWORKING AND THE INTERNET 6566 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNET As discussed in Section 1.3.3, online service providers, such as Google andMicrosoft, have deployed their own extensive private networks, which not onlyconnect together their globally distributed data centers, but are used to bypassthe Internet as much as possible by peering directly with lower-tier ISPs. As aresult, Google provides search results and email access almost instantaneously,as if their data centers were running within ones own computer. Many Internet commerce companies are now running their applications in thecloudsuch as in Amazons EC2, in Googles Application Engine, or inMicrosofts Azure. Many companies and universities have also migrated their Inter-net applications (e.g., email and Web hosting) to the cloud. Cloud companies notonly provide applications scalable computing and storage environments, but alsoprovide the applications implicit access to their high-performance private networks.1.8 SummaryIn this chapter weve covered a tremendous amount of material! Weve looked at thevarious pieces of hardware and software that make up the Internet in particular andcomputer networks in general. We started at the edge of the network, looking at endsystems and applications, and at the transport service provided to the applicationsrunning on the end systems. We also looked at the link-layer technologies and phys-ical media typically found in the access network. We then dove deeper inside thenetwork, into the network core, identifying packet switching and circuit switchingas the two basic approaches for transporting data through a telecommunication net-work, and we examined the strengths and weaknesses of each approach. We alsoexamined the structure of the global Internet, learning that the Internet is a networkof networks. We saw that the Internets hierarchical structure, consisting of higher-and lower-tier ISPs, has allowed it to scale to include thousands of networks.In the second part of this introductory chapter, we examined several topics centralto the field of computer networking. We first examined the causes of delay, through-put and packet loss in a packet-switched network. We developed simple quantitativemodels for transmission, propagation, and queuing delays as well as for throughput;well make extensive use of these delay models in the homework problems through-out this book. Next we examined protocol layering and service models, key architec-tural principles in networking that we will also refer back to throughout this book. Wealso surveyed some of the more prevalent security attacks in the Internet day. We fin-ished our introduction to networking with a brief history of computer networking. Thefirst chapter in itself constitutes a mini-course in computer networking.So, we have indeed covered a tremendous amount of ground in this first chapter!If youre a bit overwhelmed, dont worry. In the following chapters well revisit all ofthese ideas, covering them in much more detail (thats a promise, not a threat!). At thispoint, we hope you leave this chapter with a still-developing intuition for the piecesthat make up a network, a still-developing command of the vocabulary of networking(dont be shy about referring back to this chapter), and an ever-growing desire to learnmore about networking. Thats the task ahead of us for the rest of this book.Road-Mapping This BookBefore starting any trip, you should always glance at a road map in order to becomefamiliar with the major roads and junctures that lie ahead. For the trip we are aboutto embark on, the ultimate destination is a deep understanding of the how, what, andwhy of computer networks. Our road map is the sequence of chapters of this book:1. Computer Networks and the Internet2. Application Layer3. Transport Layer4. Network Layer5. Link Layer and Local Area Networks6. Wireless and Mobile Networks7. Multimedia Networking8. Security in Computer Networks9. Network ManagementChapters 2 through 5 are the four core chapters of this book. You should noticethat these chapters are organized around the top four layers of the five-layer Internetprotocol stack, one chapter for each layer. Further note that our journey will begin atthe top of the Internet protocol stack, namely, the application layer, and will workits way downward. The rationale behind this top-down journey is that once weunderstand the applications, we can understand the network services needed to sup-port these applications. We can then, in turn, examine the various ways in whichsuch services might be implemented by a network architecture. Covering applica-tions early thus provides motivation for the remainder of the text.The second half of the bookChapters 6 through 9zooms in on four enor-mously important (and somewhat independent) topics in modern computer network-ing. In Chapter 6, we examine wireless and mobile networks, including wirelessLANs (including WiFi and Bluetooth), Cellular telephony networks (includingGSM, 3G, and 4G), and mobility (in both IP and GSM networks). In Chapter 7(Multimedia Networking) we examine audio and video applications such as Internetphone, video conferencing, and streaming of stored media. We also look at how apacket-switched network can be designed to provide consistent quality of service toaudio and video applications. In Chapter 8 (Security in Computer Networks), wefirst look at the underpinnings of encryption and network security, and then weexamine how the basic theory is being applied in a broad range of Internet contexts.The last chapter (Network Management) examines the key issues in network man-agement as well as the primary Internet protocols used for network management.1.8 SUMMARY 67Homework Problems and QuestionsChapter 1 Review QuestionsSECTION 1.1R1. What is the difference between a host and an end system? List several differenttypes of end systems. Is a Web server an end system?R2. The word protocol is often used to describe diplomatic relations. How doesWikipedia describe diplomatic protocol?R3. Why are standards important for protocols?SECTION 1.2R4. List six access technologies. Classify each one as home access, enterpriseaccess, or wide-area wireless access.R5. Is HFC transmission rate dedicated or shared among users? Are collisionspossible in a downstream HFC channel? Why or why not?R6. List the available residential access technologies in your city. For each typeof access, provide the advertised downstream rate, upstream rate, andmonthly price.R7. What is the transmission rate of Ethernet LANs? R8. What are some of the physical media that Ethernet can run over?R9. Dial-up modems, HFC, DSL and FTTH are all used for residential access.For each of these access technologies, provide a range of transmission ratesand comment on whether the transmission rate is shared or dedicated.R10. Describe the most popular wireless Internet access technologies today. Com-pare and contrast them.SECTION 1.3R11. Suppose there is exactly one packet switch between a sending host and areceiving host. The transmission rates between the sending host and theswitch and between the switch and the receiving host are R1 and R2, respec-tively. Assuming that the switch uses store-and-forward packet switching,what is the total end-to-end delay to send a packet of length L? (Ignorequeuing, propagation delay, and processing delay.)R12. What advantage does a circuit-switched network have over a packet-switchednetwork? What advantages does TDM have over FDM in a circuit-switchednetwork?R13. Suppose users share a 2 Mbps link. Also suppose each user transmits continu-ously at 1 Mbps when transmitting, but each user transmits only 20 percent ofthe time. (See the discussion of statistical multiplexing in Section 1.3.)68 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETa. When circuit switching is used, how many users can be supported?b. For the remainder of this problem, suppose packet switching is used. Whywill there be essentially no queuing delay before the link if two or fewerusers transmit at the same time? Why will there be a queuing delay ifthree users transmit at the same time?c. Find the probability that a given user is transmitting.d. Suppose now there are three users. Find the probability that at any giventime, all three users are transmitting simultaneously. Find the fraction oftime during which the queue grows.R14. Why will two ISPs at the same level of the hierarchy often peer with eachother? How does an IXP earn money?R15. Some content providers have created their own networks. Describe Googlesnetwork. What motivates content providers to create these networks?SECTION 1.4R16. Consider sending a packet from a source host to a destination host over afixed route. List the delay components in the end-to-end delay. Which ofthese delays are constant and which are variable?R17. Visit the Transmission Versus Propagation Delay applet at the companionWeb site. Among the rates, propagation delay, and packet sizes available,find a combination for which the sender finishes transmitting before the firstbit of the packet reaches the receiver. Find another combination for whichthe first bit of the packet reaches the receiver before the sender finishestransmitting.R18. How long does it take a packet of length 1,000 bytes to propagate over a linkof distance 2,500 km, propagation speed 2.5 108 m/s, and transmission rate2 Mbps? More generally, how long does it take a packet of length L to propa-gate over a link of distance d, propagation speed s, and transmission rate Rbps? Does this delay depend on packet length? Does this delay depend ontransmission rate?R19. Suppose Host A wants to send a large file to Host B. The path from Host A toHost B has three links, of rates R1 = 500 kbps, R2 = 2 Mbps, and R3 = 1 Mbps.a. Assuming no other traffic in the network, what is the throughput for thefile transfer?b. Suppose the file is 4 million bytes. Dividing the file size by the throughput,roughly how long will it take to transfer the file to Host B?c. Repeat (a) and (b), but now with R2 reduced to 100 kbps.R20. Suppose end system A wants to send a large file to end system B. At a veryhigh level, describe how end system A creates packets from the file. WhenHOMEWORK PROBLEMS AND QUESTIONS 69one of these packets arrives to a packet switch, what information in thepacket does the switch use to determine the link onto which the packet isforwarded? Why is packet switching in the Internet analogous to driving fromone city to another and asking directions along the way?R21. Visit the Queuing and Loss applet at the companion Web site. What is themaximum emission rate and the minimum transmission rate? With thoserates, what is the traffic intensity? Run the applet with these rates and deter-mine how long it takes for packet loss to occur. Then repeat the experiment asecond time and determine again how long it takes for packet loss to occur.Are the values different? Why or why not?SECTION 1.5R22. List five tasks that a layer can perform. Is it possible that one (or more) ofthese tasks could be performed by two (or more) layers?R23. What are the five layers in the Internet protocol stack? What are the principalresponsibilities of each of these layers?R24. What is an application-layer message? A transport-layer segment? A network-layer datagram? A link-layer frame?R25. Which layers in the Internet protocol stack does a router process? Whichlayers does a link-layer switch process? Which layers does a host process?SECTION 1.6R26. What is the difference between a virus and a worm?R27. Describe how a botnet can be created, and how it can be used for a DDoSattack.R28. Suppose Alice and Bob are sending packets to each other over a computernetwork. Suppose Trudy positions herself in the network so that she cancapture all the packets sent by Alice and send whatever she wants to Bob;she can also capture all the packets sent by Bob and send whatever shewants to Alice. List some of the malicious things Trudy can do from thisposition.ProblemsP1. Design and describe an application-level protocol to be used between anautomatic teller machine and a banks centralized computer. Your protocolshould allow a users card and password to be verified, the account balance(which is maintained at the centralized computer) to be queried, and anaccount withdrawal to be made (that is, money disbursed to the user). Your70 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETprotocol entities should be able to handle the all-too-common case in whichthere is not enough money in the account to cover the withdrawal. Specifyyour protocol by listing the messages exchanged and the action taken by theautomatic teller machine or the banks centralized computer on transmissionand receipt of messages. Sketch the operation of your protocol for the case of asimple withdrawal with no errors, using a diagram similar to that in Figure 1.2.Explicitly state the assumptions made by your protocol about the underlyingend-to-end transport service.P2. Equation 1.1 gives a formula for the end-to-end delay of sending one packetof length L over N links of transmission rate R. Generalize this formula forsending P such packets back-to-back over the N links.P3. Consider an application that transmits data at a steady rate (for example, thesender generates an N-bit unit of data every k time units, where k is small andfixed). Also, when such an application starts, it will continue running for arelatively long period of time. Answer the following questions, briefly justi-fying your answer:a. Would a packet-switched network or a circuit-switched network be moreappropriate for this application? Why?b. Suppose that a packet-switched network is used and the only traffic inthis network comes from such applications as described above. Further-more, assume that the sum of the application data rates is less than thecapacities of each and every link. Is some form of congestion controlneeded? Why?P4. Consider the circuit-switched network in Figure 1.13. Recall that there are 4circuits on each link. Label the four switches A, B, C and D, going in theclockwise direction.a. What is the maximum number of simultaneous connections that can be inprogress at any one time in this network?b. Suppose that all connections are between switches A and C. What is themaximum number of simultaneous connections that can be in progress?c. Suppose we want to make four connections between switches A and C,and another four connections between switches B and D. Can we routethese calls through the four links to accommodate all eight connections?P5. Review the car-caravan analogy in Section 1.4. Assume a propagation speedof 100 km/hour.a. Suppose the caravan travels 150 km, beginning in front of one tollbooth,passing through a second tollbooth, and finishing just after a third toll-booth. What is the end-to-end delay?b. Repeat (a), now assuming that there are eight cars in the caravan instead of ten.PROBLEMS 71P6. This elementary problem begins to explore propagation delay and transmis-sion delay, two central concepts in data networking. Consider two hosts, Aand B, connected by a single link of rate R bps. Suppose that the two hostsare separated by m meters, and suppose the propagation speed along the linkis s meters/sec. Host A is to send a packet of size L bits to Host B.a. Express the propagation delay, dprop, in terms of m and s.b. Determine the transmission time of the packet, dtrans, in terms of Land R.c. Ignoring processing and queuing delays, obtain an expression for the end-to-end delay.d. Suppose Host A begins to transmit the packet at time t = 0. At time t = dtrans,where is the last bit of the packet?e. Suppose dprop is greater than dtrans. At time t = dtrans, where is the first bit ofthe packet?f. Suppose dprop is less than dtrans. At time t = dtrans, where is the first bit ofthe packet?g. Suppose s = 2.5 108, L = 120 bits, and R = 56 kbps. Find the distance mso that dprop equals dtrans.P7. In this problem, we consider sending real-time voice from Host A to Host Bover a packet-switched network (VoIP). Host A converts analog voice to adigital 64 kbps bit stream on the fly. Host A then groups the bits into 56-bytepackets. There is one link between Hosts A and B; its transmission rate is 2Mbps and its propagation delay is 10 msec. As soon as Host A gathers apacket, it sends it to Host B. As soon as Host B receives an entire packet, itconverts the packets bits to an analog signal. How much time elapses fromthe time a bit is created (from the original analog signal at Host A) until thebit is decoded (as part of the analog signal at Host B)?P8. Suppose users share a 3 Mbps link. Also suppose each user requires 150 kbps when transmitting, but each user transmits only 10 percent of thetime. (See the discussion of packet switching versus circuit switching inSection 1.3.)a. When circuit switching is used, how many users can be supported?b. For the remainder of this problem, suppose packet switching is used. Findthe probability that a given user is transmitting.c. Suppose there are 120 users. Find the probability that at any given time,exactly n users are transmitting simultaneously. (Hint: Use the binomialdistribution.)d. Find the probability that there are 21 or more users transmittingsimultaneously.72 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETVideoNoteExploring propagationdelay and transmissiondelayP9. Consider the discussion in Section 1.3 of packet switching versus circuitswitching in which an example is provided with a 1 Mbps link. Users aregenerating data at a rate of 100 kbps when busy, but are busy generatingdata only with probability p = 0.1. Suppose that the 1 Mbps link is replacedby a 1 Gbps link.a. What is N, the maximum number of users that can be supportedsimultaneously under circuit switching?b. Now consider packet switching and a user population of M users. Give aformula (in terms of p, M, N) for the probability that more than N users aresending data.P10. Consider a packet of length L which begins at end system A and travels overthree links to a destination end system. These three links are connected bytwo packet switches. Let di, si, and Ri denote the length, propagation speed,and the transmission rate of link i, for i = 1, 2, 3. The packet switch delayseach packet by dproc. Assuming no queuing delays, in terms of di, si, Ri,(i = 1,2,3), and L, what is the total end-to-end delay for the packet? Supposenow the packet is 1,500 bytes, the propagation speed on all three links is 2.5 108 m/s, the transmission rates of all three links are 2 Mbps, the packet switchprocessing delay is 3 msec, the length of the first link is 5,000 km, the lengthof the second link is 4,000 km, and the length of the last link is 1,000 km. Forthese values, what is the end-to-end delay?P11. In the above problem, suppose R1 = R2 = R3 = R and dproc = 0. Further sup-pose the packet switch does not store-and-forward packets but instead imme-diately transmits each bit it receives before waiting for the entire packet toarrive. What is the end-to-end delay?P12. A packet switch receives a packet and determines the outbound link to whichthe packet should be forwarded. When the packet arrives, one other packet ishalfway done being transmitted on this outbound link and four other packetsare waiting to be transmitted. Packets are transmitted in order of arrival.Suppose all packets are 1,500 bytes and the link rate is 2 Mbps. What is thequeuing delay for the packet? More generally, what is the queuing delaywhen all packets have length L, the transmission rate is R, x bits of thecurrently-being-transmitted packet have been transmitted, and n packets arealready in the queue?P13. (a) Suppose N packets arrive simultaneously to a link at which no packetsare currently being transmitted or queued. Each packet is of length Land the link has transmission rate R. What is the average queuing delayfor the N packets?(b) Now suppose that N such packets arrive to the link every LN/R seconds.What is the average queuing delay of a packet?PROBLEMS 73P14. Consider the queuing delay in a router buffer. Let I denote traffic intensity;that is, I = La/R. Suppose that the queuing delay takes the form IL/R (1 I)for I < 1.a. Provide a formula for the total delay, that is, the queuing delay plus thetransmission delay.b. Plot the total delay as a function of L/R.P15. Let a denote the rate of packets arriving at a link in packets/sec, and let denote the links transmission rate in packets/sec. Based on the formula forthe total delay (i.e., the queuing delay plus the transmission delay) derived inthe previous problem, derive a formula for the total delay in terms of a and .P16. Consider a router buffer preceding an outbound link. In this problem, you willuse Littles formula, a famous formula from queuing theory. Let N denote theaverage number of packets in the buffer plus the packet being transmitted. Leta denote the rate of packets arriving at the link. Let d denote the average totaldelay (i.e., the queuing delay plus the transmission delay) experienced by apacket. Littles formula is N = a d. Suppose that on average, the buffer con-tains 10 packets, and the average packet queuing delay is 10 msec. The linkstransmission rate is 100 packets/sec. Using Littles formula, what is the aver-age packet arrival rate, assuming there is no packet loss? P17. a. Generalize Equation 1.2 in Section 1.4.3 for heterogeneous processingrates, transmission rates, and propagation delays.b. Repeat (a), but now also suppose that there is an average queuing delay ofdqueue at each node.P18. Perform a Traceroute between source and destination on the same continentat three different hours of the day.a. Find the average and standard deviation of the round-trip delays at each ofthe three hours.b. Find the number of routers in the path at each of the three hours. Did thepaths change during any of the hours?c. Try to identify the number of ISP networks that the Traceroute packets passthrough from source to destination. Routers with similar names and/or similarIP addresses should be considered as part of the same ISP. In your experiments,do the largest delays occur at the peering interfaces between adjacent ISPs?d. Repeat the above for a source and destination on different continents.Compare the intra-continent and inter-continent results.P19. (a) Visit the site www.traceroute.org and performtraceroutes from two differentcities in France to the same destination host in the United States. How manylinks are the same in the two traceroutes? Is the transatlantic link the same? 74 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETVideoNoteUsing Traceroute todiscover networkpaths and measurenetwork delaywww.traceroute.org(b) Repeat (a) but this time choose one city in France and another city inGermany.(c) Pick a city in the United States, and perform traceroutes to two hosts, eachin a different city in China. How many links are common in the twotraceroutes? Do the two traceroutes diverge before reaching China?P20. Consider the throughput example corresponding to Figure 1.20(b). Nowsuppose that there are M client-server pairs rather than 10. Denote Rs, Rc, andR for the rates of the server links, client links, and network link. Assume allother links have abundant capacity and that there is no other traffic in thenetwork besides the traffic generated by the M client-server pairs. Derive ageneral expression for throughput in terms of Rs, Rc, R, and M.P21. Consider Figure 1.19(b). Now suppose that there are M paths between theserver and the client. No two paths share any link. Path k (k = 1, . . ., M ) con-sists of N links with transmission rates Rk1, Rk2, . . ., RkN. If the server can onlyuse one path to send data to the client, what is the maximum throughput thatthe server can achieve? If the server can use all M paths to send data, what isthe maximum throughput that the server can achieve?P22. Consider Figure 1.19(b). Suppose that each link between the server and theclient has a packet loss probability p, and the packet loss probabilities forthese links are independent. What is the probability that a packet (sent by theserver) is successfully received by the receiver? If a packet is lost in the pathfrom the server to the client, then the server will re-transmit the packet. Onaverage, how many times will the server re-transmit the packet in order forthe client to successfully receive the packet?P23. Consider Figure 1.19(a). Assume that we know the bottleneck link along thepath from the server to the client is the first link with rate Rs bits/sec. Supposewe send a pair of packets back to back from the server to the client, and thereis no other traffic on this path. Assume each packet of size L bits, and bothlinks have the same propagation delay dprop.a. What is the packet inter-arrival time at the destination? That is, how muchtime elapses from when the last bit of the first packet arrives until the lastbit of the second packet arrives?b. Now assume that the second link is the bottleneck link (i.e., Rc < Rs). Is itpossible that the second packet queues at the input queue of the secondlink? Explain. Now suppose that the server sends the second packet T sec-onds after sending the first packet. How large must T be to ensure noqueuing before the second link? Explain.P24. Suppose you would like to urgently deliver 40 terabytes data from Boston toLos Angeles. You have available a 100 Mbps dedicated link for data transfer.Would you prefer to transmit the data via this link or instead use FedEx over-night delivery? Explain. PROBLEMS 75P25. Suppose two hosts, A and B, are separated by 20,000 kilometers and areconnected by a direct link of R = 2 Mbps. Suppose the propagation speedover the link is 2.5 108 meters/sec.a. Calculate the bandwidth-delay product, R dprop.b. Consider sending a file of 800,000 bits from Host A to Host B. Supposethe file is sent continuously as one large message. What is the maximumnumber of bits that will be in the link at any given time?c. Provide an interpretation of the bandwidth-delay product.d. What is the width (in meters) of a bit in the link? Is it longer than a foot-ball field?e. Derive a general expression for the width of a bit in terms of the propaga-tion speed s, the transmission rate R, and the length of the link m.P26. Referring to problem P25, suppose we can modify R. For what value of R isthe width of a bit as long as the length of the link?P27. Consider problem P25 but now with a link of R = 1 Gbps.a. Calculate the bandwidth-delay product, R dprop.b. Consider sending a file of 800,000 bits from Host A to Host B. Supposethe file is sent continuously as one big message. What is the maximumnumber of bits that will be in the link at any given time?c. What is the width (in meters) of a bit in the link?P28. Refer again to problem P25.a. How long does it take to send the file, assuming it is sent continuously?b. Suppose now the file is broken up into 20 packets with each packet con-taining 40,000 bits. Suppose that each packet is acknowledged by thereceiver and the transmission time of an acknowledgment packet isnegligible. Finally, assume that the sender cannot send a packet until thepreceding one is acknowledged. How long does it take to send the file?c. Compare the results from (a) and (b).P29. Suppose there is a 10 Mbps microwave link between a geostationary satelliteand its base station on Earth. Every minute the satellite takes a digital photo andsends it to the base station. Assume a propagation speed of 2.4 108 meters/sec.a. What is the propagation delay of the link?b. What is the bandwidth-delay product, R dprop?c. Let x denote the size of the photo. What is the minimum value of x for themicrowave link to be continuously transmitting?P30. Consider the airline travel analogy in our discussion of layering in Section1.5, and the addition of headers to protocol data units as they flow down76 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETthe protocol stack. Is there an equivalent notion of header information thatis added to passengers and baggage as they move down the airline protocolstack?P31. In modern packet-switched networks, including the Internet, the source hostsegments long, application-layer messages (for example, an image or a musicfile) into smaller packets and sends the packets into the network. The receiverthen reassembles the packets back into the original message. We refer to thisprocess as message segmentation. Figure 1.27 illustrates the end-to-endtransport of a message with and without message segmentation. Consider amessage that is 8 106 bits long that is to be sent from source to destination inFigure 1.27. Suppose each link in the figure is 2 Mbps. Ignore propagation,queuing, and processing delays.a. Consider sending the message from source to destination without messagesegmentation. How long does it take to move the message from the sourcehost to the first packet switch? Keeping in mind that each switch usesstore-and-forward packet switching, what is the total time to move themessage from source host to destination host?b. Now suppose that the message is segmented into 800 packets, with eachpacket being 10,000 bits long. How long does it take to move the firstpacket from source host to the first switch? When the first packet is beingsent from the first switch to the second switch, the second packet is beingsent from the source host to the first switch. At what time will the secondpacket be fully received at the first switch?c. How long does it take to move the file from source host to destination hostwhen message segmentation is used? Compare this result with youranswer in part (a) and comment.PROBLEMS 77Sourcea. Packet switch Packet switch DestinationMessageSourceb. Packet switchPacketPacket switch DestinationFigure 1.27 End-to-end message transport: (a) without messagesegmentation; (b) with message segmentationd. In addition to reducing delay, what are reasons to use message segmentation?e. Discuss the drawbacks of message segmentation.P32. Experiment with the Message Segmentation applet at the books Web site. Dothe delays in the applet correspond to the delays in the previous problem?How do link propagation delays affect the overall end-to-end delay for packetswitching (with message segmentation) and for message switching?P33. Consider sending a large file of F bits from Host A to Host B. There are threelinks (and two switches) between A and B, and the links are uncongested (that is, no queuing delays). Host A segments the file into segments of S bits each andadds 80 bits of header to each segment, forming packets of L = 80 + S bits. Eachlink has a transmission rate of R bps. Find the value of S that minimizes thedelay of moving the file from Host A to Host B. Disregard propagation delay.P34. Skype offers a service that allows you to make a phone call from a PC to anordinary phone. This means that the voice call must pass through both the Internet and through a telephone network. Discuss how this might be done.Wireshark LabTell me and I forget. Show me and I remember. Involve me and I understand.Chinese proverbOnes understanding of network protocols can often be greatly deepened by seeingthem in action and by playing around with themobserving the sequence of mes-sages exchanged between two protocol entities, delving into the details of protocoloperation, causing protocols to perform certain actions, and observing these actionsand their consequences. This can be done in simulated scenarios or in a real networkenvironment such as the Internet. The Java applets at the textbook Web site take thefirst approach. In the Wireshark labs, well take the latter approach. Youll run net-work applications in various scenarios using a computer on your desk, at home, orin a lab. Youll observe the network protocols in your computer, interacting andexchanging messages with protocol entities executing elsewhere in the Internet.Thus, you and your computer will be an integral part of these live labs. Youllobserveand youll learnby doing.The basic tool for observing the messages exchanged between executingprotocol entities is called a packet sniffer. As the name suggests, a packet snifferpassively copies (sniffs) messages being sent from and received by your computer;it also displays the contents of the various protocol fields of these captured mes-sages. A screenshot of the Wireshark packet sniffer is shown in Figure 1.28. Wire-shark is a free packet sniffer that runs on Windows, Linux/Unix, and Mac78 CHAPTER 1 COMPUTER NETWORKS AND THE INTERNETWIRESHARK LAB 79CommandmenusListing ofcapturedpacketsDetails ofselectedpacketheaderPacketcontents inhexadecimaland ASCIIFigure 1.28 A Wireshark screen shot (Wireshark screenshot reprintedby permission of the Wireshark Foundation.)computers. Throughout the textbook, you will find Wireshark labs that allow you toexplore a number of the protocols studied in the chapter. In this first Wireshark lab,youll obtain and install a copy of Wireshark, access a Web site, and capture andexamine the protocol messages being exchanged between your Web browser and theWeb server.You can find full details about this first Wireshark lab (including instructionsabout how to obtain and install Wireshark) at the Web site http://www.awl.com/kurose-ross.http://www.awl.com/kurose-rosshttp://www.awl.com/kurose-ross80Leonard KleinrockLeonard Kleinrock is a professor of computer science at the Universityof California, Los Angeles. In 1969, his computer at UCLA becamethe first node of the Internet. His creation of packet-switching princi-ples in 1961 became the technology behind the Internet. Hereceived his B.E.E. from the City College of New York (CCNY) andhis masters and PhD in electrical engineering from MIT.AN INTERVIEW WITH…What made you decide to specialize in networking/Internet technology?As a PhD student at MIT in 1959, I looked around and found that most of my classmateswere doing research in the area of information theory and coding theory. At MIT, there wasthe great researcher, Claude Shannon, who had launched these fields and had solved most ofthe important problems already. The research problems that were left were hard and of less-er consequence. So I decided to launch out in a new area that no one else had yet conceivedof. Remember that at MIT I was surrounded by lots of computers, and it was clear to methat soon these machines would need to communicate with each other. At the time, therewas no effective way for them to do so, so I decided to develop the technology that wouldpermit efficient and reliable data networks to be created.What was your first job in the computer industry? What did it entail?I went to the evening session at CCNY from 1951 to 1957 for my bachelors degree inelectrical engineering. During the day, I worked first as a technician and then as an engi-neer at a small, industrial electronics firm called Photobell. While there, I introduceddigital technology to their product line. Essentially, we were using photoelectric devicesto detect the presence of certain items (boxes, people, etc.) and the use of a circuitknown then as a bistable multivibrator was just the kind of technology we needed tobring digital processing into this field of detection. These circuits happen to be the build-ing blocks for computers, and have come to be known as flip-flops or switches in todaysvernacular.What was going through your mind when you sent the first host-to-host message (fromUCLA to the Stanford Research Institute)?Frankly, we had no idea of the importance of that event. We had not prepared a special mes-sage of historic significance, as did so many inventors of the past (Samuel Morse with Whathath God wrought. or Alexander Graham Bell with Watson, come here! I want you. orNeal Amstrong with Thats one small step for a man, one giant leap for mankind.) Thoseguys were smart! They understood media and public relations. All we wanted to do was tologin to the SRI computer. So we typed the L, which was correctly received, we typed theo which was received, and then we typed the g which caused the SRI host computer tocrash! So, it turned out that our message was the shortest and perhaps the most propheticmessage ever, namely Lo! as in Lo and behold!Earlier that year, I was quoted in a UCLA press release saying that once the network wasup and running, it would be possible to gain access to computer utilities from our homes andoffices as easily as we gain access to electricity and telephone connectivity. So my vision atthat time was that the Internet would be ubiquitous, always on, always available, anyone withany device could connect from any location, and it would be invisible. However, I neveranticipated that my 99-year-old mother would use the Internetand indeed she did!What is your vision for the future of networking?The easy part of the vision is to predict the infrastructure itself. I anticipate that we see con-siderable deployment of nomadic computing, mobile devices, and smart spaces. Indeed, theavailability of lightweight, inexpensive, high-performance, portable computing, and commu-nication devices (plus the ubiquity of the Internet) has enabled us to become nomads.Nomadic computing refers to the technology that enables end users who travel from place toplace to gain access to Internet services in a transparent fashion, no matter where they traveland no matter what device they carry or gain access to. The harder part of the vision is topredict the applications and services, which have consistently surprised us in dramatic ways(email, search technologies, the world-wide-web, blogs, social networks, user generation, andsharing of music, photos, and videos, etc.). We are on the verge of a new class of surprisingand innovative mobile applications delivered to our hand-held devices. The next step will enable us to move out from the netherworld of cyberspace to thephysical world of smart spaces. Our environments (desks, walls, vehicles, watches, belts, andso on) will come alive with technology, through actuators, sensors, logic, processing, storage,cameras, microphones, speakers, displays, and communication. This embedded technologywill allow our environment to provide the IP services we want. When I walk into a room, theroom will know I entered. I will be able to communicate with my environment naturally, asin spoken English; my requests will generate replies that present Web pages to me from walldisplays, through my eyeglasses, as speech, holograms, and so forth.Looking a bit further out, I see a networking future that includes the following addi-tional key components. I see intelligent software agents deployed across the networkwhose function it is to mine data, act on that data, observe trends, and carry out tasksdynamically and adaptively. I see considerably more network traffic generated not so muchby humans, but by these embedded devices and these intelligent software agents. I seelarge collections of self-organizing systems controlling this vast, fast network. I see hugeamounts of information flashing across this network instantaneously with this informationundergoing enormous processing and filtering. The Internet will essentially be a pervasiveglobal nervous system. I see all these things and more as we move headlong through thetwenty-first century.81What people have inspired you professionally?By far, it was Claude Shannon from MIT, a brilliant researcher who had the ability to relatehis mathematical ideas to the physical world in highly intuitive ways. He was on my PhDthesis committee.Do you have any advice for students entering the networking/Internet field?The Internet and all that it enables is a vast new frontier, full of amazing challenges. Thereis room for great innovation. Dont be constrained by todays technology. Reach out andimagine what could be and then make it happen.82CHAPTER 2ApplicationLayer83Network applications are the raisons dtre of a computer networkif we couldntconceive of any useful applications, there wouldnt be any need for networking proto-cols that support these applications. Since the Internets inception, numerous useful andentertaining applications have indeed been created. These applications have been thedriving force behind the Internets success, motivating people in homes, schools, gov-ernments, and businesses to make the Internet an integral part of their daily activities.Internet applications include the classic text-based applications that becamepopular in the 1970s and 1980s: text email, remote access to computers, file trans-fers, and newsgroups. They include the killer application of the mid-1990s, theWorld Wide Web, encompassing Web surfing, search, and electronic commerce.They include instant messaging and P2P file sharing, the two killer applicationsintroduced at the end of the millennium. Since 2000, we have seen an explosion ofpopular voice and video applications, including: voice-over-IP (VoIP) and videoconferencing over IP such as Skype; user-generated video distribution such asYouTube; and movies on demand such as Netflix. During this same period we havealso seen the immergence of highly engaging multi-player online games, includingSecond Life and World of Warcraft. And most recently, we have seen the emergenceof a new generation of social networking applications, such as Facebook and Twitter,which have created engaging human networks on top of the Internets network ofrouters and communication links. Clearly, there has been no slowing down of newand exciting Internet applications. Perhaps some of the readers of this text will cre-ate the next generation of killer Internet applications!In this chapter we study the conceptual and implementation aspects of networkapplications. We begin by defining key application-layer concepts, including networkservices required by applications, clients and servers, processes, and transport-layerinterfaces. We examine several network applications in detail, including the Web,e-mail, DNS, and peer-to-peer (P2P) file distribution (Chapter 8 focuses on multime-dia applications, including streaming video and VoIP). We then cover network applica-tion development, over both TCP and UDP. In particular, we study the socket APIand walk through some simple client-server applications in Python. We also provideseveral fun and interesting socket programming assignments at the end of the chapter.The application layer is a particularly good place to start our study of protocols.Its familiar ground. Were acquainted with many of the applications that rely on theprotocols well study. It will give us a good feel for what protocols are all about andwill introduce us to many of the same issues that well see again when we study trans-port, network, and link layer protocols.2.1 Principles of Network ApplicationsSuppose you have an idea for a new network application. Perhaps this applicationwill be a great service to humanity, or will please your professor, or will bring yougreat wealth, or will simply be fun to develop. Whatever the motivation may be, letsnow examine how you transform the idea into a real-world network application.At the core of network application development is writing programs that run ondifferent end systems and communicate with each other over the network. Forexample, in the Web application there are two distinct programs that communicatewith each other: the browser program running in the users host (desktop, laptop,tablet, smartphone, and so on); and the Web server program running in the Webserver host. As another example, in a P2P file-sharing system there is a program ineach host that participates in the file-sharing community. In this case, the programsin the various hosts may be similar or identical.Thus, when developing your new application, you need to write software thatwill run on multiple end systems. This software could be written, for example, in C,Java, or Python. Importantly, you do not need to write software that runs on network-core devices, such as routers or link-layer switches. Even if you wanted to writeapplication software for these network-core devices, you wouldnt be able to do so.As we learned in Chapter 1, and as shown earlier in Figure 1.24, network-coredevices do not function at the application layer but instead function at lower layersspecifically at the network layer and below. This basic designnamely, confiningapplication software to the end systemsas shown in Figure 2.1, has facilitated therapid development and deployment of a vast array of network applications.84 CHAPTER 2 APPLICATION LAYER2.1 PRINCIPLES OF NETWORK APPLICATIONS 85Mobile NetworkTransportNetworkLinkPhysicalApplicationNational orGlobal ISPLocal orRegional ISPEnterprise NetworkHome NetworkTransportNetworkLinkApplicationPhysicalTransportNetworkLinkPhysicalApplicationFigure 2.1 Communication for a network application takes placebetween end systems at the application layer2.1.1 Network Application ArchitecturesBefore diving into software coding, you should have a broad architectural plan foryour application. Keep in mind that an applications architecture is distinctly differ-ent from the network architecture (e.g., the five-layer Internet architecture discussedin Chapter 1). From the application developers perspective, the network architec-ture is fixed and provides a specific set of services to applications. The applicationarchitecture, on the other hand, is designed by the application developer and dic-tates how the application is structured over the various end systems. In choosing theapplication architecture, an application developer will likely draw on one of the twopredominant architectural paradigms used in modern network applications: theclient-server architecture or the peer-to-peer (P2P) architectureIn a client-server architecture, there is an always-on host, called the server,which services requests from many other hosts, called clients. A classic example is theWeb application for which an always-on Web server services requests from browsersrunning on client hosts. When a Web server receives a request for an object from aclient host, it responds by sending the requested object to the client host. Note thatwith the client-server architecture, clients do not directly communicate with eachother; for example, in the Web application, two browsers do not directly communi-cate. Another characteristic of the client-server architecture is that the server has afixed, well-known address, called an IP address (which well discuss soon). Becausethe server has a fixed, well-known address, and because the server is always on, aclient can always contact the server by sending a packet to the servers IP address.Some of the better-known applications with a client-server architecture include theWeb, FTP, Telnet, and e-mail. The client-server architecture is shown in Figure 2.2(a).Often in a client-server application, a single-server host is incapable of keeping upwith all the requests from clients. For example, a popular social-networking site canquickly become overwhelmed if it has only one server handling all of its requests. Forthis reason, a data center, housing a large number of hosts, is often used to create apowerful virtual server. The most popular Internet servicessuch as search engines(e.g., Google and Bing), Internet commerce (e.g., Amazon and e-Bay), Web-basedemail (e.g., Gmail and Yahoo Mail), social networking (e.g., Facebook and Twitter)employ one or more data centers. As discussed in Section 1.3.3, Google has 30 to 50data centers distributed around the world, which collectively handle search, YouTube,Gmail, and other services. A data center can have hundreds of thousands of servers,which must be powered and maintained. Additionally, the service providers must payrecurring interconnection and bandwidth costs for sending data from their data centers.In a P2P architecture, there is minimal (or no) reliance on dedicated servers indata centers. Instead the application exploits direct communication between pairs ofintermittently connected hosts, called peers. The peers are not owned by the serviceprovider, but are instead desktops and laptops controlled by users, with most of thepeers residing in homes, universities, and offices. Because the peers communicatewithout passing through a dedicated server, the architecture is called peer-to-peer.Many of todays most popular and traffic-intensive applications are based on P2Parchitectures. These applications include file sharing (e.g., BitTorrent), peer-assisted86 CHAPTER 2 APPLICATION LAYERdownload acceleration (e.g., Xunlei), Internet Telephony (e.g., Skype), and IPTV (e.g.,Kankan and PPstream). The P2P architecture is illustrated in Figure 2.2(b). We men-tion that some applications have hybrid architectures, combining both client-serverand P2P elements. For example, for many instant messaging applications, servers areused to track the IP addresses of users, but user-to-user messages are sent directlybetween user hosts (without passing through intermediate servers).One of the most compelling features of P2P architectures is their self-scalability.For example, in a P2P file-sharing application, although each peer generatesworkload by requesting files, each peer also adds service capacity to the systemby distributing files to other peers. P2P architectures are also cost effective, sincethey normally dont require significant server infrastructure and server bandwidth(in contrast with clients-server designs with datacenters). However, future P2Papplications face three major challenges:1. ISP Friendly. Most residential ISPs (including DSL and cable ISPs) have beendimensioned for asymmetrical bandwidth usage, that is, for much more2.1 PRINCIPLES OF NETWORK APPLICATIONS 87Figure 2.2 (a) Client-server architecture; (b) P2P architecturea. Client-server architecture b. Peer-to-peer architecturedownstream than upstream traffic. But P2P video streaming and file distribu-tion applications shift upstream traffic from servers to residential ISPs, therebyputting significant stress on the ISPs. Future P2P applications need to bedesigned so that they are friendly to ISPs [Xie 2008].2. Security. Because of their highly distributed and open nature, P2P applicationscan be a challenge to secure [Doucer 2002; Yu 2006; Liang 2006; Naoumov2006; Dhungel 2008; LeBlond 2011].3. Incentives. The success of future P2P applications also depends on convincingusers to volunteer bandwidth, storage, and computation resources to the appli-cations, which is the challenge of incentive design [Feldman 2005; Piatek2008; Aperjis 2008; Liu 2010].2.1.2 Processes CommunicatingBefore building your network application, you also need a basic understanding ofhow the programs, running in multiple end systems, communicate with each other.In the jargon of operating systems, it is not actually programs but processes thatcommunicate. A process can be thought of as a program that is running within anend system. When processes are running on the same end system, they can com-municate with each other with interprocess communication, using rules that aregoverned by the end systems operating system. But in this book we are not par-ticularly interested in how processes in the same host communicate, but instead inhow processes running on different hosts (with potentially different operating sys-tems) communicate.Processes on two different end systems communicate with each other by exchang-ing messages across the computer network. A sending process creates and sends mes-sages into the network; a receiving process receives these messages and possiblyresponds by sending messages back. Figure 2.1 illustrates that processes communicat-ing with each other reside in the application layer of the five-layer protocol stack.Client and Server ProcessesA network application consists of pairs of processes that send messages to eachother over a network. For example, in the Web application a client browserprocess exchanges messages with a Web server process. In a P2P file-sharing sys-tem, a file is transferred from a process in one peer to a process in another peer.For each pair of communicating processes, we typically label one of the twoprocesses as the client and the other process as the server. With the Web, abrowser is a client process and a Web server is a server process. With P2P filesharing, the peer that is downloading the file is labeled as the client, and the peerthat is uploading the file is labeled as the server.You may have observed that in some applications, such as in P2P file sharing, aprocess can be both a client and a server. Indeed, a process in a P2P file-sharing sys-tem can both upload and download files. Nevertheless, in the context of any given88 CHAPTER 2 APPLICATION LAYERcommunication session between a pair of processes, we can still label one processas the client and the other process as the server. We define the client and serverprocesses as follows:In the context of a communication session between a pair of processes, theprocess that initiates the communication (that is, initially contacts the otherprocess at the beginning of the session) is labeled as the client. The processthat waits to be contacted to begin the session is the server.In the Web, a browser process initializes contact with a Web server process;hence the browser process is the client and the Web server process is the server. InP2P file sharing, when Peer A asks Peer B to send a specific file, Peer A is the clientand Peer B is the server in the context of this specific communication session. Whentheres no confusion, well sometimes also use the terminology client side andserver side of an application. At the end of this chapter, well step through simplecode for both the client and server sides of network applications.The Interface Between the Process and the Computer NetworkAs noted above, most applications consist of pairs of communicating processes,with the two processes in each pair sending messages to each other. Any messagesent from one process to another must go through the underlying network. A processsends messages into, and receives messages from, the network through a softwareinterface called a socket. Lets consider an analogy to help us understand processesand sockets. A process is analogous to a house and its socket is analogous to its door.When a process wants to send a message to another process on another host, itshoves the message out its door (socket). This sending process assumes that there isa transportation infrastructure on the other side of its door that will transport themessage to the door of the destination process. Once the message arrives at the des-tination host, the message passes through the receiving processs door (socket), andthe receiving process then acts on the messageFigure 2.3 illustrates socket communication between two processes that com-municate over the Internet. (Figure 2.3 assumes that the underlying transport pro-tocol used by the processes is the Internets TCP protocol.) As shown in thisfigure, a socket is the interface between the application layer and the transportlayer within a host. It is also referred to as the Application Programming Inter-face (API) between the application and the network, since the socket is the pro-gramming interface with which network applications are built. The applicationdeveloper has control of everything on the application-layer side of the socket buthas little control of the transport-layer side of the socket. The only control that theapplication developer has on the transport-layer side is (1) the choice of transportprotocol and (2) perhaps the ability to fix a few transport-layer parameters such asmaximum buffer and maximum segment sizes (to be covered in Chapter 3). Oncethe application developer chooses a transport protocol (if a choice is available),2.1 PRINCIPLES OF NETWORK APPLICATIONS 89the application is built using the transport-layer services provided by that proto-col. Well explore sockets in some detail in Section 2.7.Addressing ProcessesIn order to send postal mail to a particular destination, the destination needs to havean address. Similarly, in order for a process running on one host to send packets to aprocess running on another host, the receiving process needs to have an address. To identify the receiving process, two pieces of information need to be specified:(1) the address of the host and (2) an identifier that specifies the receiving processin the destination host.In the Internet, the host is identified by its IP address. Well discuss IPaddresses in great detail in Chapter 4. For now, all we need to know is that an IPaddress is a 32-bit quantity that we can think of as uniquely identifying the host.In addition to knowing the address of the host to which a message is destined, thesending process must also identify the receiving process (more specifically, thereceiving socket) running in the host. This information is needed because in gen-eral a host could be running many network applications. A destination port num-ber serves this purpose. Popular applications have been assigned specific portnumbers. For example, a Web server is identified by port number 80. A mailserver process (using the SMTP protocol) is identified by port number 25. A listof well-known port numbers for all Internet standard protocols can be found athttp://www.iana.org. Well examine port numbers in detail in Chapter 3.90 CHAPTER 2 APPLICATION LAYERProcessHost orserverHost orserverControlledby applicationdeveloperControlledby applicationdeveloperProcessTCP withbuffers,variables InternetControlledby operatingsystemControlledby operatingsystemTCP withbuffers,variablesSocket SocketFigure 2.3 Application processes, sockets, and underlying transport protocolhttp://www.iana.org2.1.3 Transport Services Available to ApplicationsRecall that a socket is the interface between the application process and the transport-layer protocol. The application at the sending side pushes messagesthrough the socket. At the other side of the socket, the transport-layer protocolhas the responsibility of getting the messages to the socket of the receivingprocess.Many networks, including the Internet, provide more than one transport-layerprotocol. When you develop an application, you must choose one of the availabletransport-layer protocols. How do you make this choice? Most likely, you wouldstudy the services provided by the available transport-layer protocols, and then pickthe protocol with the services that best match your applications needs. The situa-tion is similar to choosing either train or airplane transport for travel between twocities. You have to choose one or the other, and each transportation mode offers dif-ferent services. (For example, the train offers downtown pickup and drop-off,whereas the plane offers shorter travel time.)What are the services that a transport-layer protocol can offer to applicationsinvoking it? We can broadly classify the possible services along four dimensions:reliable data transfer, throughput, timing, and security.Reliable Data TransferAs discussed in Chapter 1, packets can get lost within a computer network. Forexample, a packet can overflow a buffer in a router, or can be discarded by a hostor router after having some of its bits corrupted. For many applicationssuch aselectronic mail, file transfer, remote host access, Web document transfers, andfinancial applicationsdata loss can have devastating consequences (in the lattercase, for either the bank or the customer!). Thus, to support these applications,something has to be done to guarantee that the data sent by one end of the appli-cation is delivered correctly and completely to the other end of the application. Ifa protocol provides such a guaranteed data delivery service, it is said to providereliable data transfer. One important service that a transport-layer protocol canpotentially provide to an application is process-to-process reliable data transfer.When a transport protocol provides this service, the sending process can just passits data into the socket and know with complete confidence that the data willarrive without errors at the receiving process.When a transport-layer protocol doesnt provide reliable data transfer, some ofthe data sent by the sending process may never arrive at the receiving process. Thismay be acceptable for loss-tolerant applications, most notably multimedia applica-tions such as conversational audio/video that can tolerate some amount of data loss.In these multimedia applications, lost data might result in a small glitch in theaudio/videonot a crucial impairment.2.1 PRINCIPLES OF NETWORK APPLICATIONS 91ThroughputIn Chapter 1 we introduced the concept of available throughput, which, in the con-text of a communication session between two processes along a network path, isthe rate at which the sending process can deliver bits to the receiving process.Because other sessions will be sharing the bandwidth along the network path, andbecause these other sessions will be coming and going, the available throughputcan fluctuate with time. These observations lead to another natural service that atransport-layer protocol could provide, namely, guaranteed available throughputat some specified rate. With such a service, the application could request a guar-anteed throughput of r bits/sec, and the transport protocol would then ensure thatthe available throughput is always at least r bits/sec. Such a guaranteed through-put service would appeal to many applications. For example, if an Internet teleph-ony application encodes voice at 32 kbps, it needs to send data into the networkand have data delivered to the receiving application at this rate. If the transportprotocol cannot provide this throughput, the application would need to encode ata lower rate (and receive enough throughput to sustain this lower coding rate) ormay have to give up, since receiving, say, half of the needed throughput is of littleor no use to this Internet telephony application. Applications that have throughputrequirements are said to be bandwidth-sensitive applications. Many currentmultimedia applications are bandwidth sensitive, although some multimediaapplications may use adaptive coding techniques to encode digitized voice orvideo at a rate that matches the currently available throughput.While bandwidth-sensitive applications have specific throughput require-ments, elastic applications can make use of as much, or as little, throughput as happens to be available. Electronic mail, file transfer, and Web transfers areall elastic applications. Of course, the more throughput, the better. Theresan adage that says that one cannot be too rich, too thin, or have too muchthroughput!TimingA transport-layer protocol can also provide timing guarantees. As with throughputguarantees, timing guarantees can come in many shapes and forms. An exampleguarantee might be that every bit that the sender pumps into the socket arrives at thereceivers socket no more than 100 msec later. Such a service would be appealing tointeractive real-time applications, such as Internet telephony, virtual environments,teleconferencing, and multiplayer games, all of which require tight timing con-straints on data delivery in order to be effective. (See Chapter 7, [Gauthier 1999;Ramjee 1994].) Long delays in Internet telephony, for example, tend to result inunnatural pauses in the conversation; in a multiplayer game or virtual interactiveenvironment, a long delay between taking an action and seeing the response fromthe environment (for example, from another player at the end of an end-to-end con-nection) makes the application feel less realistic. For non-real-time applications,92 CHAPTER 2 APPLICATION LAYERlower delay is always preferable to higher delay, but no tight constraint is placed onthe end-to-end delays.SecurityFinally, a transport protocol can provide an application with one or more securityservices. For example, in the sending host, a transport protocol can encrypt all datatransmitted by the sending process, and in the receiving host, the transport-layerprotocol can decrypt the data before delivering the data to the receiving process.Such a service would provide confidentiality between the two processes, even if thedata is somehow observed between sending and receiving processes. A transportprotocol can also provide other security services in addition to confidentiality,including data integrity and end-point authentication, topics that well cover indetail in Chapter 8.2.1.4 Transport Services Provided by the InternetUp until this point, we have been considering transport services that a computernetwork could provide in general. Lets now get more specific and examine thetype of transport services provided by the Internet. The Internet (and, more gen-erally, TCP/IP networks) makes two transport protocols available to applications,UDP and TCP. When you (as an application developer) create a new networkapplication for the Internet, one of the first decisions you have to make iswhether to use UDP or TCP. Each of these protocols offers a different set of serv-ices to the invoking applications. Figure 2.4 shows the service requirements forsome selected applications.2.1 PRINCIPLES OF NETWORK APPLICATIONS 93Application Data Loss Throughput Time-SensitiveFile transfer/download No loss Elastic NoE-mail No loss Elastic NoWeb documents No loss Elastic (few kbps) NoFigure 2.4 Requirements of selected network applicationsInternet telephony/Video conferencingLoss-tolerant Audio: few kbps1MbpsVideo: 10 kbps5 MbpsYes: 100s of msecStreaming stored Loss-tolerant Same as above Yes: few secondsaudio/videoInteractive games Loss-tolerant Few kbps10 kbps Yes: 100s of msecInstant messaging No loss Elastic Yes and noTCP ServicesThe TCP service model includes a connection-oriented service and a reliable datatransfer service. When an application invokes TCP as its transport protocol, theapplication receives both of these services from TCP. Connection-oriented service. TCP has the client and server exchange transport-layer control information with each other before the application-level messagesbegin to flow. This so-called handshaking procedure alerts the client and server,allowing them to prepare for an onslaught of packets. After the handshaking phase,a TCP connection is said to exist between the sockets of the two processes. Theconnection is a full-duplex connection in that the two processes can send messagesto each other over the connection at the same time. When the application finishessending messages, it must tear down the connection. In Chapter 3 well discussconnection-oriented service in detail and examine how it is implemented.94 CHAPTER 2 APPLICATION LAYERSECURING TCPNeither TCP nor UDP provide any encryptionthe data that the sending process pass-es into its socket is the same data that travels over the network to the destinationprocess. So, for example, if the sending process sends a password in cleartext (i.e.,unencrypted) into its socket, the cleartext password will travel over all the links betweensender and receiver, potentially getting sniffed and discovered at any of the interveninglinks. Because privacy and other security issues have become critical for many applica-tions, the Internet community has developed an enhancement for TCP, called SecureSockets Layer (SSL). TCP-enhanced-with-SSL not only does everything that traditionalTCP does but also provides critical process-to-process security services, includingencryption, data integrity, and end-point authentication. We emphasize that SSL is nota third Internet transport protocol, on the same level as TCP and UDP, but instead is anenhancement of TCP, with the enhancements being implemented in the applicationlayer. In particular, if an application wants to use the services of SSL, it needs toinclude SSL code (existing, highly optimized libraries and classes) in both the client andserver sides of the application. SSL has its own socket API that is similar to the tradition-al TCP socket API. When an application uses SSL, the sending process passes cleartextdata to the SSL socket; SSL in the sending host then encrypts the data and passes theencrypted data to the TCP socket. The encrypted data travels over the Internet to theTCP socket in the receiving process. The receiving socket passes the encrypted data toSSL, which decrypts the data. Finally, SSL passes the cleartext data through its SSLsocket to the receiving process. Well cover SSL in some detail in Chapter 8.FOCUS ON SECURITY Reliable data transfer service. The communicating processes can rely on TCPto deliver all data sent without error and in the proper order. When one side ofthe application passes a stream of bytes into a socket, it can count on TCP todeliver the same stream of bytes to the receiving socket, with no missing orduplicate bytes.TCP also includes a congestion-control mechanism, a service for the generalwelfare of the Internet rather than for the direct benefit of the communicatingprocesses. The TCP congestion-control mechanism throttles a sending process (clientor server) when the network is congested between sender and receiver. As we willsee in Chapter 3, TCP congestion control also attempts to limit each TCP connectionto its fair share of network bandwidth.UDP ServicesUDP is a no-frills, lightweight transport protocol, providing minimal services. UDPis connectionless, so there is no handshaking before the two processes start tocommunicate. UDP provides an unreliable data transfer servicethat is, when a processsends a message into a UDP socket, UDP provides no guarantee that the messagewill ever reach the receiving process. Furthermore, messages that do arrive at thereceiving process may arrive out of order.UDP does not include a congestion-control mechanism, so the sending side ofUDP can pump data into the layer below (the network layer) at any rate it pleases.(Note, however, that the actual end-to-end throughput may be less than this rate dueto the limited transmission capacity of intervening links or due to congestion). Services Not Provided by Internet Transport ProtocolsWe have organized transport protocol services along four dimensions: reliable datatransfer, throughput, timing, and security. Which of these services are provided byTCP and UDP? We have already noted that TCP provides reliable end-to-end datatransfer. And we also know that TCP can be easily enhanced at the application layerwith SSL to provide security services. But in our brief description of TCP and UDP,conspicuously missing was any mention of throughput or timing guaranteesserv-ices not provided by todays Internet transport protocols. Does this mean that time-sensitive applications such as Internet telephony cannot run in todays Internet? Theanswer is clearly nothe Internet has been hosting time-sensitive applications formany years. These applications often work fairly well because they have beendesigned to cope, to the greatest extent possible, with this lack of guarantee. Wellinvestigate several of these design tricks in Chapter 7. Nevertheless, clever designhas its limitations when delay is excessive, or the end-to-end throughput is limited.In summary, todays Internet can often provide satisfactory service to time-sensitiveapplications, but it cannot provide any timing or throughput guarantees.2.1 PRINCIPLES OF NETWORK APPLICATIONS 95Figure 2.5 indicates the transport protocols used by some popular Internetapplications. We see that e-mail, remote terminal access, the Web, and file trans-fer all use TCP. These applications have chosen TCP primarily because TCP pro-videsreliable data transfer, guaranteeing that all data will eventually get to itsdestination. Because Internet telephony applications (such as Skype) can oftentolerate some loss but require a minimal rate to be effective, developers of Inter-net telephony applications usually prefer to run their applications over UDP,thereby circumventing TCPs congestion control mechanism and packet over-heads. But because many firewalls are configured to block (most types of) UDPtraffic, Internet telephony applications often are designed to use TCP as a backupif UDP communication fails.2.1.5 Application-Layer ProtocolsWe have just learned that network processes communicate with each other by send-ing messages into sockets. But how are these messages structured? What are themeanings of the various fields in the messages? When do the processes send the mes-sages? These questions bring us into the realm of application-layer protocols. Anapplication-layer protocol defines how an applications processes, running on dif-ferent end systems, pass messages to each other. In particular, an application-layerprotocol defines: The types of messages exchanged, for example, request messages and responsemessages The syntax of the various message types, such as the fields in the message andhow the fields are delineated96 CHAPTER 2 APPLICATION LAYERApplication Application-Layer Protocol Underlying Transport ProtocolElectronic mail SMTP [RFC 5321] TCPRemote terminal access Telnet [RFC 854] TCPWeb HTTP [RFC 2616] TCPFile transfer FTP [RFC 959] TCPStreaming multimedia HTTP (e.g., YouTube) TCPInternet telephony SIP [RFC 3261], RTP [RFC 3550], or proprietary UDP or TCP(e.g., Skype)Figure 2.5 Popular Internet applications, their application-layer protocols, and their underlying transport protocols The semantics of the fields, that is, the meaning of the information in the fields Rules for determining when and how a process sends messages and responds tomessagesSome application-layer protocols are specified in RFCs and are therefore in the publicdomain. For example, the Webs application-layer protocol, HTTP (the HyperTextTransfer Protocol [RFC 2616]), is available as an RFC. If a browser developer followsthe rules of the HTTP RFC, the browser will be able to retrieve Web pages from any Web server that has also followed the rules of the HTTP RFC. Many otherapplication-layer protocols are proprietary and intentionally not available in the publicdomain. For example, Skype uses proprietary application-layer protocols.It is important to distinguish between network applications and application-layerprotocols. An application-layer protocol is only one piece of a network application(albeit, a very important piece of the application from our point of view!). Lets look ata couple of examples. The Web is a client-server application that allows users toobtain documents from Web servers on demand. The Web application consists ofmany components, including a standard for document formats (that is, HTML), Webbrowsers (for example, Firefox and Microsoft Internet Explorer), Web servers (forexample, Apache and Microsoft servers), and an application-layer protocol. TheWebs application-layer protocol, HTTP, defines the format and sequence of messagesexchanged between browser and Web server. Thus, HTTP is only one piece (albeit, animportant piece) of the Web application. As another example, an Internet e-mail appli-cation also has many components, including mail servers that house user mailboxes;mail clients (such as Microsoft Outlook) that allow users to read and create messages; astandard for defining the structure of an e-mail message; and application-layer proto-cols that define how messages are passed between servers, how messages are passedbetween servers and mail clients, and how the contents of message headers are to beinterpreted. The principal application-layer protocol for electronic mail is SMTP(Simple Mail Transfer Protocol) [RFC 5321]. Thus, e-mails principal application-layerprotocol, SMTP, is only one piece (albeit, an important piece) of the e-mail application.2.1.6 Network Applications Covered in This BookNew public domain and proprietary Internet applications are being developed everyday. Rather than covering a large number of Internet applications in an encyclope-dic manner, we have chosen to focus on a small number of applications that are bothpervasive and important. In this chapter we discuss five important applications: theWeb, file transfer, electronic mail, directory service, and P2P applications. We firstdiscuss the Web, not only because it is an enormously popular application, but alsobecause its application-layer protocol, HTTP, is straightforward and easy to under-stand. After covering the Web, we briefly examine FTP, because it provides a nicecontrast to HTTP. We then discuss electronic mail, the Internets first killer applica-tion. E-mail is more complex than the Web in the sense that it makes use of not one2.1 PRINCIPLES OF NETWORK APPLICATIONS 97but several application-layer protocols. After e-mail, we cover DNS, which providesa directory service for the Internet. Most users do not interact with DNS directly;instead, users invoke DNS indirectly through other applications (including the Web,file transfer, and electronic mail). DNS illustrates nicely how a piece of core net-work functionality (network-name to network-address translation) can be imple-mented at the application layer in the Internet. Finally, we discuss in this chapterseveral P2P applications, focusing on file sharing applications, and distributedlookup services. In Chapter 7, well cover multimedia applications, includingstreaming video and voice-over-IP.2.2 The Web and HTTPUntil the early 1990s the Internet was used primarily by researchers, academics, anduniversity students to log in to remote hosts, to transfer files from local hosts to remotehosts and vice versa, to receive and send news, and to receive and send electronicmail. Although these applications were (and continue to be) extremely useful, theInternet was essentially unknown outside of the academic and research communities.Then, in the early 1990s, a major new application arrived on the scenethe WorldWide Web [Berners-Lee 1994]. The Web was the first Internet application that caughtthe general publics eye. It dramatically changed, and continues to change, how peo-ple interact inside and outside their work environments. It elevated the Internet fromjust one of many data networks to essentially the one and only data network.Perhaps what appeals the most to users is that the Web operates on demand.Users receive what they want, when they want it. This is unlike traditional broad-cast radio and television, which force users to tune in when the content providermakes the content available. In addition to being available on demand, the Web hasmany other wonderful features that people love and cherish. It is enormously easyfor any individual to make information available over the Webeveryone canbecome a publisher at extremely low cost. Hyperlinks and search engines help usnavigate through an ocean of Web sites. Graphics stimulate our senses. Forms,JavaScript, Java applets, and many other devices enable us to interact with pagesand sites. And the Web serves as a platform for many killer applications emergingafter 2003, including YouTube, Gmail, and Facebook.2.2.1 Overview of HTTPThe HyperText Transfer Protocol (HTTP), the Webs application-layer protocol,is at the heart of the Web. It is defined in [RFC 1945] and [RFC 2616]. HTTP isimplemented in two programs: a client program and a server program. The clientprogram and server program, executing on different end systems, talk to each otherby exchanging HTTP messages. HTTP defines the structure of these messages andhow the client and server exchange the messages. Before explaining HTTP in detail,we should review some Web terminology.98 CHAPTER 2 APPLICATION LAYERA Web page (also called a document) consists of objects. An object is simply afilesuch as an HTML file, a JPEG image, a Java applet, or a video clipthat isaddressable by a single URL. Most Web pages consist of a base HTML file andseveral referenced objects. For example, if a Web page contains HTML text and fiveJPEG images, then the Web page has six objects: the base HTML file plus the fiveimages. The base HTML file references the other objects in the page with theobjects URLs. Each URL has two components: the hostname of the server thathouses the object and the objects path name. For example, the URLhttp://www.someSchool.edu/someDepartment/picture.gifhas www.someSchool.edu for a hostname and /someDepartment/picture.gif for a path name. Because Web browsers (such as Internet Explorerand Firefox) implement the client side of HTTP, in the context of the Web, we will usethe words browser and client interchangeably. Web servers, which implement theserver side of HTTP, house Web objects, each addressable by a URL. Popular Webservers include Apache and Microsoft Internet Information Server.HTTP defines how Web clients request Web pages from Web servers and howservers transfer Web pages to clients. We discuss the interaction between client andserver in detail later, but the general idea is illustrated in Figure 2.6. When a userrequests a Web page (for example, clicks on a hyperlink), the browser sends HTTPrequest messages for the objects in the page to the server. The server receives therequests and responds with HTTP response messages that contain the objects.HTTP uses TCP as its underlying transport protocol (rather than running on top ofUDP). The HTTP client first initiates a TCP connection with the server. Once the con-nection is established, the browser and the server processes access TCP through theirsocket interfaces. As described in Section 2.1, on the client side the socket interface isthe door between the client process and the TCP connection; on the server side it is the2.2 THE WEB AND HTTP 99HTTP requestHTTP responseHTTP responseHTTP requestPC runningInternet ExplorerLinux runningFirefoxServer runningApache Web serverFigure 2.6 HTTP request-response behaviorhttp://www.someSchool.edu/someDepartment/picture.gifwww.someSchool.edudoor between the server process and the TCP connection. The client sends HTTPrequest messages into its socket interface and receives HTTP response messages fromits socket interface. Similarly, the HTTP server receives request messages from itssocket interface and sends response messages into its socket interface. Once the clientsends a message into its socket interface, the message is out of the clients hands and isin the hands of TCP. Recall from Section 2.1 that TCP provides a reliable data trans-fer service to HTTP. This implies that each HTTP request message sent by a clientprocess eventually arrives intact at the server; similarly, each HTTP response messagesent by the server process eventually arrives intact at the client. Here we see one of thegreat advantages of a layered architectureHTTP need not worry about lost data or thedetails of how TCP recovers from loss or reordering of data within the network. That isthe job of TCP and the protocols in the lower layers of the protocol stack.It is important to note that the server sends requested files to clients without stor-ing any state information about the client. If a particular client asks for the same objecttwice in a period of a few seconds, the server does not respond by saying that it justserved the object to the client; instead, the server resends the object, as it has com-pletely forgotten what it did earlier. Because an HTTP server maintains no informa-tion about the clients, HTTP is said to be a stateless protocol. We also remark that theWeb uses the client-server application architecture, as described in Section 2.1. A Webserver is always on, with a fixed IP address, and it services requests from potentiallymillions of different browsers.2.2.2 Non-Persistent and Persistent ConnectionsIn many Internet applications, the client and server communicate for an extendedperiod of time, with the client making a series of requests and the server responding toeach of the requests. Depending on the application and on how the application is beingused, the series of requests may be made back-to-back, periodically at regular intervals,or intermittently. When this client-server interaction is taking place over TCP, the appli-cation developer needs to make an important decisionshould each request/responsepair be sent over a separate TCP connection, or should all of the requests and their cor-responding responses be sent over the same TCP connection? In the former approach,the application is said to use non-persistent connections; and in the latter approach,persistent connections. To gain a deep understanding of this design issue, lets exam-ine the advantages and disadvantages of persistent connections in the context of a spe-cific application, namely, HTTP, which can use both non-persistent connections andpersistent connections. Although HTTP uses persistent connections in its default mode,HTTP clients and servers can be configured to use non-persistent connections instead.HTTP with Non-Persistent ConnectionsLets walk through the steps of transferring a Web page from server to client for thecase of non-persistent connections. Lets suppose the page consists of a base HTML100 CHAPTER 2 APPLICATION LAYERfile and 10 JPEG images, and that all 11 of these objects reside on the same server.Further suppose the URL for the base HTML file ishttp://www.someSchool.edu/someDepartment/home.indexHere is what happens:1. The HTTP client process initiates a TCP connection to the serverwww.someSchool.edu on port number 80, which is the default port num-ber for HTTP. Associated with the TCP connection, there will be a socket at theclient and a socket at the server.2. The HTTP client sends an HTTP request message to the server via its socket. Therequest message includes the path name /someDepartment/home.index.(We will discuss HTTP messages in some detail below.)3. The HTTP server process receives the request message via its socket, retrievesthe object /someDepartment/home.index from its storage (RAM ordisk), encapsulates the object in an HTTP response message, and sends theresponse message to the client via its socket.4. The HTTP server process tells TCP to close the TCP connection. (But TCPdoesnt actually terminate the connection until it knows for sure that the clienthas received the response message intact.)5. The HTTP client receives the response message. The TCP connection termi-nates. The message indicates that the encapsulated object is an HTML file. Theclient extracts the file from the response message, examines the HTML file,and finds references to the 10 JPEG objects.6. The first four steps are then repeated for each of the referenced JPEG objects.As the browser receives the Web page, it displays the page to the user. Two differ-ent browsers may interpret (that is, display to the user) a Web page in somewhat differ-ent ways. HTTP has nothing to do with how a Web page is interpreted by a client. TheHTTP specifications ([RFC 1945] and [RFC 2616]) define only the communicationprotocol between the client HTTP program and the server HTTP program.The steps above illustrate the use of non-persistent connections, where each TCPconnection is closed after the server sends the objectthe connection does not persistfor other objects. Note that each TCP connection transports exactly one request mes-sage and one response message. Thus, in this example, when a user requests the Webpage, 11 TCP connections are generated.In the steps described above, we were intentionally vague about whether theclient obtains the 10 JPEGs over 10 serial TCP connections, or whether some of theJPEGs are obtained over parallel TCP connections. Indeed, users can configuremodern browsers to control the degree of parallelism. In their default modes, mostbrowsers open 5 to 10 parallel TCP connections, and each of these connections han-dles one request-response transaction. If the user prefers, the maximum number of2.2 THE WEB AND HTTP 101http://www.someSchool.edu/someDepartment/home.indexwww.someSchool.eduparallel connections can be set to one, in which case the 10 connections are estab-lished serially. As well see in the next chapter, the use of parallel connections short-ens the response time.Before continuing, lets do a back-of-the-envelope calculation to estimate theamount of time that elapses from when a client requests the base HTML file untilthe entire file is received by the client. To this end, we define the round-trip time(RTT), which is the time it takes for a small packet to travel from client to serverand then back to the client. The RTT includes packet-propagation delays, packet-queuing delays in intermediate routers and switches, and packet-processingdelays. (These delays were discussed in Section 1.4.) Now consider what happenswhen a user clicks on a hyperlink. As shown in Figure 2.7, this causes the browserto initiate a TCP connection between the browser and the Web server; thisinvolves a three-way handshakethe client sends a small TCP segment to theserver, the server acknowledges and responds with a small TCP segment, and,finally, the client acknowledges back to the server. The first two parts of the three-way handshake take one RTT. After completing the first two parts of the hand-shake, the client sends the HTTP request message combined with the third part of102 CHAPTER 2 APPLICATION LAYERTimeat clientTimeat serverInitiate TCPconnectionRTTRequest fileRTTEntire file receivedTime to transmit fileFigure 2.7 Back-of-the-envelope calculation for the time needed torequest and receive an HTML filethe three-way handshake (the acknowledgment) into the TCP connection. Oncethe request message arrives at the server, the server sends the HTML file into theTCP connection. This HTTP request/response eats up another RTT. Thus, roughly,the total response time is two RTTs plus the transmission time at the server of theHTML file.HTTP with Persistent ConnectionsNon-persistent connections have some shortcomings. First, a brand-new connec-tion must be established and maintained for each requested object. For each ofthese connections, TCP buffers must be allocated and TCP variables must be keptin both the client and server. This can place a significant burden on the Web server,which may be serving requests from hundreds of different clients simultaneously.Second, as we just described, each object suffers a delivery delay of two RTTsone RTT to establish the TCP connection and one RTT to request and receive anobject.With persistent connections, the server leaves the TCP connection open aftersending a response. Subsequent requests and responses between the same client andserver can be sent over the same connection. In particular, an entire Web page (inthe example above, the base HTML file and the 10 images) can be sent over a singlepersistent TCP connection. Moreover, multiple Web pages residing on the sameserver can be sent from the server to the same client over a single persistent TCPconnection. These requests for objects can be made back-to-back, without waitingfor replies to pending requests (pipelining). Typically, the HTTP server closes a con-nection when it isnt used for a certain time (a configurable timeout interval). Whenthe server receives the back-to-back requests, it sends the objects back-to-back. Thedefault mode of HTTP uses persistent connections with pipelining. Well quantita-tively compare the performance of non-persistent and persistent connections in thehomework problems of Chapters 2 and 3. You are also encouraged to see [Heide-mann 1997; Nielsen 1997].2.2.3 HTTP Message FormatThe HTTP specifications [RFC 1945; RFC 2616] include the definitions of theHTTP message formats. There are two types of HTTP messages, request messagesand response messages, both of which are discussed below.HTTP Request MessageBelow we provide a typical HTTP request message:GET /somedir/page.html HTTP/1.1Host: www.someschool.edu2.2 THE WEB AND HTTP 103www.someschool.eduConnection: closeUser-agent: Mozilla/5.0Accept-language: frWe can learn a lot by taking a close look at this simple request message. First ofall, we see that the message is written in ordinary ASCII text, so that your ordinarycomputer-literate human being can read it. Second, we see that the message consistsof five lines, each followed by a carriage return and a line feed. The last line is fol-lowed by an additional carriage return and line feed. Although this particular requestmessage has five lines, a request message can have many more lines or as few asone line. The first line of an HTTP request message is called the request line; thesubsequent lines are called the header lines. The request line has three fields: themethod field, the URL field, and the HTTP version field. The method field can takeon several different values, including GET, POST, HEAD, PUT, and DELETE.The great majority of HTTP request messages use the GET method. The GETmethod is used when the browser requests an object, with the requested object iden-tified in the URL field. In this example, the browser is requesting the object/somedir/page.html. The version is self-explanatory; in this example, thebrowser implements version HTTP/1.1.Now lets look at the header lines in the example. The header line Host:www.someschool.edu specifies the host on which the object resides. You mightthink that this header line is unnecessary, as there is already a TCP connection inplace to the host. But, as well see in Section 2.2.5, the information provided by thehost header line is required by Web proxy caches. By including the Connection:close header line, the browser is telling the server that it doesnt want to botherwith persistent connections; it wants the server to close the connection after sendingthe requested object. The User-agent: header line specifies the user agent, thatis, the browser type that is making the request to the server. Here the user agent isMozilla/5.0, a Firefox browser. This header line is useful because the server canactually send different versions of the same object to different types of user agents.(Each of the versions is addressed by the same URL.) Finally, the Accept-language: header indicates that the user prefers to receive a French version ofthe object, if such an object exists on the server; otherwise, the server should sendits default version. The Accept-language: header is just one of many contentnegotiation headers available in HTTP.Having looked at an example, lets now look at the general format of a requestmessage, as shown in Figure 2.8. We see that the general format closely follows ourearlier example. You may have noticed, however, that after the header lines (and theadditional carriage return and line feed) there is an entity body. The entity body isempty with the GET method, but is used with the POST method. An HTTP clientoften uses the POST method when the user fills out a formfor example, when auser provides search words to a search engine. With a POST message, the user is stillrequesting a Web page from the server, but the specific contents of the Web page104 CHAPTER 2 APPLICATION LAYERwww.someSchool.edudepend on what the user entered into the form fields. If the value of the method fieldis POST, then the entity body contains what the user entered into the form fields.We would be remiss if we didnt mention that a request generated with a formdoes not necessarily use the POST method. Instead, HTML forms often use the GETmethod and include the inputted data (in the form fields) in the requested URL. Forexample, if a form uses the GET method, has two fields, and the inputs to the twofields are monkeys and bananas, then the URL will have the structurewww.somesite.com/animalsearch?monkeys&bananas. In your day-to-day Web surfing, you have probably noticed extended URLs of this sort.The HEAD method is similar to the GET method. When a server receives arequest with the HEAD method, it responds with an HTTP message but it leaves outthe requested object. Application developers often use the HEAD method for debug-ging. The PUT method is often used in conjunction with Web publishing tools. Itallows a user to upload an object to a specific path (directory) on a specific Webserver. The PUT method is also used by applications that need to upload objects toWeb servers. The DELETE method allows a user, or an application, to delete anobject on a Web server.HTTP Response MessageBelow we provide a typical HTTP response message. This response message couldbe the response to the example request message just discussed.HTTP/1.1 200 OKConnection: close2.2 THE WEB AND HTTP 105method sp sp cr lfcr lfheader field name:Header linesBlank lineEntity bodyRequest linevaluespcr lfcr lfheader field name: valuespURL VersionFigure 2.8 General format of an HTTP request messagewww.somesite.com/animalsearch?monkeys&bananasDate: Tue, 09 Aug 2011 15:44:04 GMTServer: Apache/2.2.3 (CentOS)Last-Modified: Tue, 09 Aug 2011 15:11:03 GMTContent-Length: 6821Content-Type: text/html(data data data data data …)Lets take a careful look at this response message. It has three sections: an ini-tial status line, six header lines, and then the entity body. The entity body is themeat of the messageit contains the requested object itself (represented by datadata data data data …). The status line has three fields: the protocol ver-sion field, a status code, and a corresponding status message. In this example, thestatus line indicates that the server is using HTTP/1.1 and that everything is OK(that is, the server has found, and is sending, the requested object).Now lets look at the header lines. The server uses the Connection: closeheader line to tell the client that it is going to close the TCP connection after sendingthe message. The Date: header line indicates the time and date when the HTTPresponse was created and sent by the server. Note that this is not the time when theobject was created or last modified; it is the time when the server retrieves theobject from its file system, inserts the object into the response message, and sendsthe response message. The Server: header line indicates that the message was gen-erated by an Apache Web server; it is analogous to the User-agent: header linein the HTTP request message. The Last-Modified: header line indicates thetime and date when the object was created or last modified. The Last-Modified:header, which we will soon cover in more detail, is critical for object caching, both inthe local client and in network cache servers (also known as proxy servers). TheContent-Length: header line indicates the number of bytes in the object beingsent. The Content-Type: header line indicates that the object in the entity body isHTML text. (The object type is officially indicated by the Content-Type: headerand not by the file extension.)Having looked at an example, lets now examine the general format of aresponse message, which is shown in Figure 2.9. This general format of the responsemessage matches the previous example of a response message. Lets say a few addi-tional words about status codes and their phrases. The status code and associatedphrase indicate the result of the request. Some common status codes and associatedphrases include: 200 OK: Request succeeded and the information is returned in the response. 301 Moved Permanently: Requested object has been permanently moved;the new URL is specified in Location: header of the response message. Theclient software will automatically retrieve the new URL.106 CHAPTER 2 APPLICATION LAYER 400 Bad Request: This is a generic error code indicating that the requestcould not be understood by the server. 404 Not Found: The requested document does not exist on this server. 505 HTTP Version Not Supported: The requested HTTP protocolversion is not supported by the server.How would you like to see a real HTTP response message? This is highly rec-ommended and very easy to do! First Telnet into your favorite Web server. Thentype in a one-line request message for some object that is housed on the server. Forexample, if you have access to a command prompt, type:telnet cis.poly.edu 80GET /~ross/ HTTP/1.1Host: cis.poly.edu(Press the carriage return twice after typing the last line.) This opens a TCP connec-tion to port 80 of the host cis.poly.edu and then sends the HTTP request mes-sage. You should see a response message that includes the base HTML file ofProfessor Rosss homepage. If youd rather just see the HTTP message lines and notreceive the object itself, replace GET with HEAD. Finally, replace /~ross/ with/~banana/ and see what kind of response message you get.In this section we discussed a number of header lines that can be used withinHTTP request and response messages. The HTTP specification defines many, many2.2 THE WEB AND HTTP 107version sp sp cr lfcr lfheader field name:Header linesBlank lineEntity bodyStatus linevaluecrspsp lfcr lfheader field name: valuestatus code phraseFigure 2.9 General format of an HTTP response messageVideoNoteUsing Wireshark toinvestigate the HTTP protocolmore header lines that can be inserted by browsers, Web servers, and network cacheservers. We have covered only a small number of the totality of header lines. Wellcover a few more below and another small number when we discuss network Webcaching in Section 2.2.5. A highly readable and comprehensive discussion of the HTTPprotocol, including its headers and status codes, is given in [Krishnamurthy 2001].How does a browser decide which header lines to include in a request mes-sage? How does a Web server decide which header lines to include in a responsemessage? A browser will generate header lines as a function of the browser typeand version (for example, an HTTP/1.0 browser will not generate any 1.1 headerlines), the user configuration of the browser (for example, preferred language), andwhether the browser currently has a cached, but possibly out-of-date, version of theobject. Web servers behave similarly: There are different products, versions, andconfigurations, all of which influence which header lines are included in responsemessages.2.2.4 User-Server Interaction: CookiesWe mentioned above that an HTTP server is stateless. This simplifies server designand has permitted engineers to develop high-performance Web servers that can han-dle thousands of simultaneous TCP connections. However, it is often desirable for aWeb site to identify users, either because the server wishes to restrict user access orbecause it wants to serve content as a function of the user identity. For these pur-poses, HTTP uses cookies. Cookies, defined in [RFC 6265], allow sites to keeptrack of users. Most major commercial Web sites use cookies today.As shown in Figure 2.10, cookie technology has four components: (1) a cookieheader line in the HTTP response message; (2) a cookie header line in the HTTPrequest message; (3) a cookie file kept on the users end system and managed by theusers browser; and (4) a back-end database at the Web site. Using Figure 2.10, letswalk through an example of how cookies work. Suppose Susan, who alwaysaccesses the Web using Internet Explorer from her home PC, contacts Amazon.comfor the first time. Let us suppose that in the past she has already visited the eBay site.When the request comes into the Amazon Web server, the server creates a uniqueidentification number and creates an entry in its back-end database that is indexedby the identification number. The Amazon Web server then responds to Susansbrowser, including in the HTTP response a Set-cookie: header, which containsthe identification number. For example, the header line might be:Set-cookie: 1678When Susans browser receives the HTTP response message, it sees the Set-cookie: header. The browser then appends a line to the special cookie file that itmanages. This line includes the hostname of the server and the identification num-ber in the Set-cookie: header. Note that the cookie file already has an entry for108 CHAPTER 2 APPLICATION LAYEReBay, since Susan has visited that site in the past. As Susan continues to browse theAmazon site, each time she requests a Web page, her browser consults her cookiefile, extracts her identification number for this site, and puts a cookie header linethat includes the identification number in the HTTP request. Specifically, each ofher HTTP requests to the Amazon server includes the header line:Cookie: 16782.2 THE WEB AND HTTP 109Client host Server hostusual http request msgusual http responseSet-cookie: 1678usual http request msgcookie: 1678usual http response msgusual http request msgcookie: 1678usual http response msgTimeOne week laterebay: 8734Server createsID 1678 for userTimeCookie fileKey:amazon: 1678ebay: 8734amazon: 1678ebay: 8734Cookie-specificactionaccessaccessentry in backenddatabaseCookie-specificactionFigure 2.10 Keeping user state with cookiesIn this manner, the Amazon server is able to track Susans activity at the Amazonsite. Although the Amazon Web site does not necessarily know Susans name, itknows exactly which pages user 1678 visited, in which order, and at what times!Amazon uses cookies to provide its shopping cart serviceAmazon can maintain alist of all of Susans intended purchases, so that she can pay for them collectively atthe end of the session.If Susan returns to Amazons site, say, one week later, her browser will continueto put the header line Cookie: 1678 in the request messages. Amazon also rec-ommends products to Susan based on Web pages she has visited at Amazon in thepast. If Susan also registers herself with Amazonproviding full name, e-mailaddress, postal address, and credit card informationAmazon can then include thisinformation in its database, thereby associating Susans name with her identificationnumber (and all of the pages she has visited at the site in the past!). This is howAmazon and other e-commerce sites provide one-click shoppingwhen Susanchooses to purchase an item during a subsequent visit, she doesnt need to re-enterher name, credit card number, or address.From this discussion we see that cookies can be used to identify a user. The firsttime a user visits a site, the user can provide a user identification (possibly his or hername). During the subsequent sessions, the browser passes a cookie header to theserver, thereby identifying the user to the server. Cookies can thus be used to createa user session layer on top of stateless HTTP. For example, when a user logs in to aWeb-based e-mail application (such as Hotmail), the browser sends cookie informa-tion to the server, permitting the server to identify the user throughout the users ses-sion with the application.Although cookies often simplify the Internet shopping experience for the user,they are controversial because they can also be considered as an invasion of privacy.As we just saw, using a combination of cookies and user-supplied account informa-tion, a Web site can learn a lot about a user and potentially sell this information to athird party. Cookie Central [Cookie Central 2012] includes extensive informationon the cookie controversy.2.2.5 Web CachingA Web cachealso called a proxy serveris a network entity that satisfies HTTPrequests on the behalf of an origin Web server. The Web cache has its own disk storageand keeps copies of recently requested objects in this storage. As shown in Figure 2.11, ausers browser can be configured so that all of the users HTTP requests are first directedto the Web cache. Once a browser is configured, each browser request for an object isfirst directed to the Web cache. As an example, suppose a browser is requesting theobject http://www.someschool.edu/campus.gif. Here is what happens:1. The browser establishes a TCP connection to the Web cache and sends anHTTP request for the object to the Web cache.110 CHAPTER 2 APPLICATION LAYERhttp://www.someschool.edu/campus.gif2. The Web cache checks to see if it has a copy of the object stored locally. If itdoes, the Web cache returns the object within an HTTP response message tothe client browser.3. If the Web cache does not have the object, the Web cache opens a TCP connec-tion to the origin server, that is, to www.someschool.edu. The Web cachethen sends an HTTP request for the object into the cache-to-server TCP con-nection. After receiving this request, the origin server sends the object withinan HTTP response to the Web cache.4. When the Web cache receives the object, it stores a copy in its local storage andsends a copy, within an HTTP response message, to the client browser (over theexisting TCP connection between the client browser and the Web cache).Note that a cache is both a server and a client at the same time. When it receivesrequests from and sends responses to a browser, it is a server. When it sends requeststo and receives responses from an origin server, it is a client.Typically a Web cache is purchased and installed by an ISP. For example, a uni-versity might install a cache on its campus network and configure all of the campusbrowsers to point to the cache. Or a major residential ISP (such as AOL) mightinstall one or more caches in its network and preconfigure its shipped browsers topoint to the installed caches.Web caching has seen deployment in the Internet for two reasons. First, a Webcache can substantially reduce the response time for a client request, particularly if thebottleneck bandwidth between the client and the origin server is much less than the bot-tleneck bandwidth between the client and the cache. If there is a high-speed connectionbetween the client and the cache, as there often is, and if the cache has the requestedobject, then the cache will be able to deliver the object rapidly to the client. Second, aswe will soon illustrate with an example, Web caches can substantially reduce traffic on2.2 THE WEB AND HTTP 111HTTP requestHTTP responseHTTP requestHTTP responseHTTP requestHTTP responseHTTP requestHTTP responseClient OriginserverOriginserverClientProxyserverFigure 2.11 Clients requesting objects through a Web cachewww.someschool.eduan institutions access link to the Internet. By reducing traffic, the institution (for exam-ple, a company or a university) does not have to upgrade bandwidth as quickly, therebyreducing costs. Furthermore, Web caches can substantially reduce Web traffic in theInternet as a whole, thereby improving performance for all applications.To gain a deeper understanding of the benefits of caches, lets consider an exam-ple in the context of Figure 2.12. This figure shows two networksthe institutionalnetwork and the rest of the public Internet. The institutional network is a high-speedLAN. A router in the institutional network and a router in the Internet are connectedby a 15 Mbps link. The origin servers are attached to the Internet but are located allover the globe. Suppose that the average object size is 1 Mbits and that the averagerequest rate from the institutions browsers to the origin servers is 15 requests persecond. Suppose that the HTTP request messages are negligibly small and thus cre-ate no traffic in the networks or in the access link (from institutional router to Inter-net router). Also suppose that the amount of time it takes from when the router on theInternet side of the access link in Figure 2.12 forwards an HTTP request (within anIP datagram) until it receives the response (typically within many IP datagrams) istwo seconds on average. Informally, we refer to this last delay as the Internet delay.112 CHAPTER 2 APPLICATION LAYERPublic InternetInstitutional network15 Mbps access link100 Mbps LANOrigin serversFigure 2.12 Bottleneck between an institutional network and the InternetThe total response timethat is, the time from the browsers request of an objectuntil its receipt of the objectis the sum of the LAN delay, the access delay (that is,the delay between the two routers), and the Internet delay. Lets now do a very crudecalculation to estimate this delay. The traffic intensity on the LAN (see Section 1.4.2) is(15 requests/sec) (1 Mbits/request)/(100 Mbps) = 0.15whereas the traffic intensity on the access link (from the Internet router to institutionrouter) is(15 requests/sec) (1 Mbits/request)/(15 Mbps) = 1A traffic intensity of 0.15 on a LAN typically results in, at most, tens of millisec-onds of delay; hence, we can neglect the LAN delay. However, as discussed inSection 1.4.2, as the traffic intensity approaches 1 (as is the case of the access linkin Figure 2.12), the delay on a link becomes very large and grows without bound.Thus, the average response time to satisfy requests is going to be on the order ofminutes, if not more, which is unacceptable for the institutions users. Clearly some-thing must be done.One possible solution is to increase the access rate from 15 Mbps to, say, 100Mbps. This will lower the traffic intensity on the access link to 0.15, which trans-lates to negligible delays between the two routers. In this case, the total responsetime will roughly be two seconds, that is, the Internet delay. But this solution alsomeans that the institution must upgrade its access link from 15 Mbps to 100 Mbps, acostly proposition.Now consider the alternative solution of not upgrading the access link butinstead installing a Web cache in the institutional network. This solution is illus-trated in Figure 2.13. Hit ratesthe fraction of requests that are satisfied by acachetypically range from 0.2 to 0.7 in practice. For illustrative purposes, letssuppose that the cache provides a hit rate of 0.4 for this institution. Because theclients and the cache are connected to the same high-speed LAN, 40 percent ofthe requests will be satisfied almost immediately, say, within 10 milliseconds, by thecache. Nevertheless, the remaining 60 percent of the requests still need to be satis-fied by the origin servers. But with only 60 percent of the requested objects passingthrough the access link, the traffic intensity on the access link is reduced from 1.0 to0.6. Typically, a traffic intensity less than 0.8 corresponds to a small delay, say, tensof milliseconds, on a 15 Mbps link. This delay is negligible compared with the two-second Internet delay. Given these considerations, average delay therefore is0.4 (0.01 seconds) + 0.6 (2.01 seconds)which is just slightly greater than 1.2 seconds. Thus, this second solution provides aneven lower response time than the first solution, and it doesnt require the institutionto upgrade its link to the Internet. The institution does, of course, have to purchase2.2 THE WEB AND HTTP 113and install a Web cache. But this cost is lowmany caches use public-domain soft-ware that runs on inexpensive PCs.Through the use of Content Distribution Networks (CDNs), Web caches areincreasingly playing an important role in the Internet. A CDN company installs manygeographically distributed caches throughout the Internet, thereby localizing much ofthe traffic. There are shared CDNs (such as Akamai and Limelight) and dedicated CDNs(such as Google and Microsoft). We will discuss CDNs in more detail in Chapter 7.2.2.6 The Conditional GETAlthough caching can reduce user-perceived response times, it introduces a new prob-lemthe copy of an object residing in the cache may be stale. In other words, theobject housed in the Web server may have been modified since the copy was cachedat the client. Fortunately, HTTP has a mechanism that allows a cache to verify that itsobjects are up to date. This mechanism is called the conditional GET. An HTTP114 CHAPTER 2 APPLICATION LAYERPublic InternetInstitutional network15 Mbps access linkInstitutionalcache100 Mbps LANOrigin serversFigure 2.13 Adding a cache to the institutional networkrequest message is a so-called conditional GET message if (1) the request messageuses the GET method and (2) the request message includes an If-Modified-Since: header line.To illustrate how the conditional GET operates, lets walk through an example.First, on the behalf of a requesting browser, a proxy cache sends a request messageto a Web server:GET /fruit/kiwi.gif HTTP/1.1Host: www.exotiquecuisine.comSecond, the Web server sends a response message with the requested object to thecache:HTTP/1.1 200 OKDate: Sat, 8 Oct 2011 15:39:29Server: Apache/1.3.0 (Unix)Last-Modified: Wed, 7 Sep 2011 09:23:24Content-Type: image/gif(data data data data data …)The cache forwards the object to the requesting browser but also caches the objectlocally. Importantly, the cache also stores the last-modified date along with theobject. Third, one week later, another browser requests the same object via thecache, and the object is still in the cache. Since this object may have been modifiedat the Web server in the past week, the cache performs an up-to-date check by issu-ing a conditional GET. Specifically, the cache sends:GET /fruit/kiwi.gif HTTP/1.1Host: www.exotiquecuisine.comIf-modified-since: Wed, 7 Sep 2011 09:23:24Note that the value of the If-modified-since: header line is exactly equal tothe value of the Last-Modified: header line that was sent by the server oneweek ago. This conditional GET is telling the server to send the object only if theobject has been modified since the specified date. Suppose the object has not beenmodified since 7 Sep 2011 09:23:24. Then, fourth, the Web server sends a responsemessage to the cache:HTTP/1.1 304 Not ModifiedDate: Sat, 15 Oct 2011 15:39:29Server: Apache/1.3.0 (Unix)(empty entity body)2.2 THE WEB AND HTTP 115www.exotiquecuisine.comwww.exotiquecuisine.comWe see that in response to the conditional GET, the Web server still sends a responsemessage but does not include the requested object in the response message. Includingthe requested object would only waste bandwidth and increase user-perceived responsetime, particularly if the object is large. Note that this last response message has 304Not Modified in the status line, which tells the cache that it can go ahead and for-ward its (the proxy caches) cached copy of the object to the requesting browser.This ends our discussion of HTTP, the first Internet protocol (an application-layerprotocol) that weve studied in detail. Weve seen the format of HTTP messages andthe actions taken by the Web client and server as these messages are sent and received.Weve also studied a bit of the Webs application infrastructure, including caches, cook-ies, and back-end databases, all of which are tied in some way to the HTTP protocol.2.3 File Transfer: FTPIn a typical FTP session, the user is sitting in front of one host (the local host)and wants to transfer files to or from a remote host. In order for the user to access the remote account, the user must provide a user identification and a pass-word. After providing this authorization information, the user can transfer filesfrom the local file system to the remote file system and vice versa. As shown inFigure 2.14, the user interacts with FTP through an FTP user agent. The user firstprovides the hostname of the remote host, causing the FTP client process in thelocal host to establish a TCP connection with the FTP server process in theremote host. The user then provides the user identification and password, whichare sent over the TCP connection as part of FTP commands. Once the server hasauthorized the user, the user copies one or more files stored in the local file sys-tem into the remote file system (or vice versa).116 CHAPTER 2 APPLICATION LAYERFTP userinterfaceLocal filesystemUseror hostRemote filesystemFTPclientFTPserverFile transferFigure 2.14 FTP moves files between local and remote file systemsHTTP and FTP are both file transfer protocols and have many common charac-teristics; for example, they both run on top of TCP. However, the two application-layerprotocols have some important differences. The most striking difference is that FTPuses two parallel TCP connections to transfer a file, a control connection and a dataconnection. The control connection is used for sending control information betweenthe two hostsinformation such as user identification, password, commands tochange remote directory, and commands to put and get files. The data connectionis used to actually send a file. Because FTP uses a separate control connection, FTP issaid to send its control information out-of-band. HTTP, as you recall, sends requestand response header lines into the same TCP connection that carries the transferredfile itself. For this reason, HTTP is said to send its control information in-band. In thenext section, well see that SMTP, the main protocol for electronic mail, also sendscontrol information in-band. The FTP control and data connections are illustrated inFigure 2.15.When a user starts an FTP session with a remote host, the client side of FTP(user) first initiates a control TCP connection with the server side (remote host) onserver port number 21. The client side of FTP sends the user identification andpassword over this control connection. The client side of FTP also sends, over thecontrol connection, commands to change the remote directory. When the serverside receives a command for a file transfer over the control connection (either to,or from, the remote host), the server side initiates a TCP data connection to theclient side. FTP sends exactly one file over the data connection and then closes thedata connection. If, during the same session, the user wants to transfer another file,FTP opens another data connection. Thus, with FTP, the control connectionremains open throughout the duration of the user session, but a new data connec-tion is created for each file transferred within a session (that is, the data connec-tions are non-persistent).Throughout a session, the FTP server must maintain state about the user. In par-ticular, the server must associate the control connection with a specific user account,and the server must keep track of the users current directory as the user wandersabout the remote directory tree. Keeping track of this state information for eachongoing user session significantly constrains the total number of sessions that FTPcan maintain simultaneously. Recall that HTTP, on the other hand, is statelessitdoes not have to keep track of any user state.2.3 FILE TRANSFER: FTP 117TCP control connection port 21TCP data connection port 20FTPclientFTPserverFigure 2.15 Control and data connections2.3.1 FTP Commands and RepliesWe end this section with a brief discussion of some of the more common FTP com-mands and replies. The commands, from client to server, and replies, from server toclient, are sent across the control connection in 7-bit ASCII format. Thus, like HTTPcommands, FTP commands are readable by people. In order to delineate successivecommands, a carriage return and line feed end each command. Each command con-sists of four uppercase ASCII characters, some with optional arguments. Some ofthe more common commands are given below: USER username: Used to send the user identification to the server. PASS password: Used to send the user password to the server. LIST: Used to ask the server to send back a list of all the files in the currentremote directory. The list of files is sent over a (new and non-persistent) dataconnection rather than the control TCP connection. RETR filename: Used to retrieve (that is, get) a file from the current direc-tory of the remote host. This command causes the remote host to initiate a dataconnection and to send the requested file over the data connection. STOR filename: Used to store (that is, put) a file into the current directoryof the remote host.There is typically a one-to-one correspondence between the command that theuser issues and the FTP command sent across the control connection. Each com-mand is followed by a reply, sent from server to client. The replies are three-digitnumbers, with an optional message following the number. This is similar in struc-ture to the status code and phrase in the status line of the HTTP response message.Some typical replies, along with their possible messages, are as follows: 331 Username OK, password required 125 Data connection already open; transfer starting 425 Cant open data connection 452 Error writing fileReaders who are interested in learning about the other FTP commands and repliesare encouraged to read RFC 959.2.4 Electronic Mail in the InternetElectronic mail has been around since the beginning of the Internet. It was the mostpopular application when the Internet was in its infancy [Segaller 1998], and has118 CHAPTER 2 APPLICATION LAYEROutgoingmessage queue Key:User mailboxSMTPUser agentUser agentUser agentUser agentUser agentUser agentMail serverMail serverMail serverSMTPSMTPbecome more and more elaborate and powerful over the years. It remains one of theInternets most important and utilized applications.As with ordinary postal mail, e-mail is an asynchronous communicationmediumpeople send and read messages when it is convenient for them, withouthaving to coordinate with other peoples schedules. In contrast with postal mail, elec-tronic mail is fast, easy to distribute, and inexpensive. Modern e-mail has many pow-erful features, including messages with attachments, hyperlinks, HTML-formattedtext, and embedded photos.In this section, we examine the application-layer protocols that are at the heartof Internet e-mail. But before we jump into an in-depth discussion of these proto-cols, lets take a high-level view of the Internet mail system and its key components.Figure 2.16 presents a high-level view of the Internet mail system. We see fromthis diagram that it has three major components: user agents, mail servers, and the2.4 ELECTRONIC MAIL IN THE INTERNET 119Figure 2.16 A high-level view of the Internet e-mail systemSimple Mail Transfer Protocol (SMTP). We now describe each of these compo-nents in the context of a sender, Alice, sending an e-mail message to a recipient,Bob. User agents allow users to read, reply to, forward, save, and compose mes-sages. Microsoft Outlook and Apple Mail are examples of user agents for e-mail.When Alice is finished composing her message, her user agent sends the message toher mail server, where the message is placed in the mail servers outgoing messagequeue. When Bob wants to read a message, his user agent retrieves the messagefrom his mailbox in his mail server.Mail servers form the core of the e-mail infrastructure. Each recipient, such asBob, has a mailbox located in one of the mail servers. Bobs mailbox manages andmaintains the messages that have been sent to him. A typical message starts its jour-ney in the senders user agent, travels to the senders mail server, and travels to the recipients mail server, where it is deposited in the recipients mailbox.120 CHAPTER 2 APPLICATION LAYERWEB E-MAILIn December 1995, just a few years after the Web was invented, Sabeer Bhatiaand Jack Smith visited the Internet venture capitalist Draper Fisher Jurvetson andproposed developing a free Web-based e-mail system. The idea was to give a freee-mail account to anyone who wanted one, and to make the accounts accessiblefrom the Web. In exchange for 15 percent of the company, Draper FisherJurvetson financed Bhatia and Smith, who formed a company called Hotmail.With three full-time people and 14 part-time people who worked for stock options,they were able to develop and launch the service in July 1996. Within a monthafter launch, they had 100,000 subscribers. In December 1997, less than 18months after launching the service, Hotmail had over 12 million subscribers andwas acquired by Microsoft, reportedly for $400 million. The success of Hotmail isoften attributed to its first-mover advantage and to the intrinsic viral marketingof e-mail. (Perhaps some of the students reading this book will be among the newentrepreneurs who conceive and develop first-mover Internet services with inherentviral marketing.)Web e-mail continues to thrive, becoming more sophisticated and powerful everyyear. One of the most popular services today is Googles gmail, which offers giga-bytes of free storage, advanced spam filtering and virus detection, e-mail encryption(using SSL), mail fetching from third-party e-mail services, and a search-oriented inter-face. Asynchronous messaging within social networks, such as Facebook, has alsobecome popular in recent years.CASE HISTORYWhen Bob wants to access the messages in his mailbox, the mail server containing his mailbox authenticates Bob (with usernames and passwords). Alicesmail server must also deal with failures in Bobs mail server. If Alices server can-not deliver mail to Bobs server, Alices server holds the message in a messagequeue and attempts to transfer the message later. Reattempts are often done every30 minutes or so; if there is no success after several days, the server removes themessage and notifies the sender (Alice) with an e-mail message.SMTP is the principal application-layer protocol for Internet electronic mail. Ituses the reliable data transfer service of TCP to transfer mail from the senders mailserver to the recipients mail server. As with most application-layer protocols,SMTP has two sides: a client side, which executes on the senders mail server, and aserver side, which executes on the recipients mail server. Both the client and serversides of SMTP run on every mail server. When a mail server sends mail to othermail servers, it acts as an SMTP client. When a mail server receives mail from othermail servers, it acts as an SMTP server.2.4.1 SMTPSMTP, defined in RFC 5321, is at the heart of Internet electronic mail. As men-tioned above, SMTP transfers messages from senders mail servers to the recipi-ents mail servers. SMTP is much older than HTTP. (The original SMTP RFCdates back to 1982, and SMTP was around long before that.) Although SMTP hasnumerous wonderful qualities, as evidenced by its ubiquity in the Internet, it isnevertheless a legacy technology that possesses certain archaic characteristics.For example, it restricts the body (not just the headers) of all mail messages tosimple 7-bit ASCII. This restriction made sense in the early 1980s when trans-mission capacity was scarce and no one was e-mailing large attachments or largeimage, audio, or video files. But today, in the multimedia era, the 7-bit ASCIIrestriction is a bit of a painit requires binary multimedia data to be encoded toASCII before being sent over SMTP; and it requires the corresponding ASCIImessage to be decoded back to binary after SMTP transport. Recall from Section2.2 that HTTP does not require multimedia data to be ASCII encoded beforetransfer.To illustrate the basic operation of SMTP, lets walk through a common sce-nario. Suppose Alice wants to send Bob a simple ASCII message.1. Alice invokes her user agent for e-mail, provides Bobs e-mail address (forexample, [email protected]), composes a message, and instructs theuser agent to send the message.2. Alices user agent sends the message to her mail server, where it is placed in amessage queue.2.4 ELECTRONIC MAIL IN THE INTERNET 1213. The client side of SMTP, running on Alices mail server, sees the message inthe message queue. It opens a TCP connection to an SMTP server, running onBobs mail server.4. After some initial SMTP handshaking, the SMTP client sends Alices messageinto the TCP connection.5. At Bobs mail server, the server side of SMTP receives the message. Bobsmail server then places the message in Bobs mailbox.6. Bob invokes his user agent to read the message at his convenience.The scenario is summarized in Figure 2.17.It is important to observe that SMTP does not normally use intermediate mailservers for sending mail, even when the two mail servers are located at oppositeends of the world. If Alices server is in Hong Kong and Bobs server is in St. Louis,the TCP connection is a direct connection between the Hong Kong and St. Louisservers. In particular, if Bobs mail server is down, the message remains in Alicesmail server and waits for a new attemptthe message does not get placed in someintermediate mail server.Lets now take a closer look at how SMTP transfers a message from a send-ing mail server to a receiving mail server. We will see that the SMTP protocol hasmany similarities with protocols that are used for face-to-face human interaction.First, the client SMTP (running on the sending mail server host) has TCP estab-lish a connection to port 25 at the server SMTP (running on the receiving mailserver host). If the server is down, the client tries again later. Once this connec-tion is established, the server and client perform some application-layerhandshakingjust as humans often introduce themselves before transferringinformation from one to another, SMTP clients and servers introduce themselvesbefore transferring information. During this SMTP handshaking phase, the122 CHAPTER 2 APPLICATION LAYERSMTPAlicesmail serverBobsmail serverAlicesagentBobsagent12 4 65Message queue Key:User mailbox3Figure 2.17 Alice sends a message to BobSMTP client indicates the e-mail address of the sender (the person who generatedthe message) and the e-mail address of the recipient. Once the SMTP client andserver have introduced themselves to each other, the client sends the message.SMTP can count on the reliable data transfer service of TCP to get the messageto the server without errors. The client then repeats this process over the sameTCP connection if it has other messages to send to the server; otherwise, itinstructs TCP to close the connection.Lets next take a look at an example transcript of messages exchangedbetween an SMTP client (C) and an SMTP server (S). The hostname of the clientis crepes.fr and the hostname of the server is hamburger.edu. TheASCII text lines prefaced with C: are exactly the lines the client sends into itsTCP socket, and the ASCII text lines prefaced with S: are exactly the lines theserver sends into its TCP socket. The following transcript begins as soon as theTCP connection is established.S: 220 hamburger.eduC: HELO crepes.frS: 250 Hello crepes.fr, pleased to meet youC: MAIL FROM:
S: 250 [email protected] Sender ok
C: RCPT TO:
S: 250 [email protected] Recipient ok
C: DATA
S: 354 Enter mail, end with . on a line by itself
C: Do you like ketchup?
C: How about pickles?
C: .
S: 250 Message accepted for delivery
C: QUIT
S: 221 hamburger.edu closing connection
In the example above, the client sends a message (Do you like ketchup?
How about pickles?) from mail server crepes.fr to mail server ham-
burger.edu. As part of the dialogue, the client issued five commands: HELO (an
abbreviation for HELLO), MAIL FROM, RCPT TO, DATA, and QUIT. These com-
mands are self-explanatory. The client also sends a line consisting of a single period,
which indicates the end of the message to the server. (In ASCII jargon, each mes-
sage ends with CRLF.CRLF, where CR and LF stand for carriage return and line
feed, respectively.) The server issues replies to each command, with each reply hav-
ing a reply code and some (optional) English-language explanation. We mention
here that SMTP uses persistent connections: If the sending mail server has several
messages to send to the same receiving mail server, it can send all of the messages
over the same TCP connection. For each message, the client begins the process with
2.4 ELECTRONIC MAIL IN THE INTERNET 123
a new MAIL FROM: crepes.fr, designates the end of message with an isolated
period, and issues QUIT only after all messages have been sent.
It is highly recommended that you use Telnet to carry out a direct dialogue with
an SMTP server. To do this, issue
telnet serverName 25
where serverName is the name of a local mail server. When you do this, you are
simply establishing a TCP connection between your local host and the mail server.
After typing this line, you should immediately receive the 220 reply from the
server. Then issue the SMTP commands HELO, MAIL FROM, RCPT TO, DATA,
CRLF.CRLF, and QUIT at the appropriate times. It is also highly recommended
that you do Programming Assignment 3 at the end of this chapter. In that assign-
ment, youll build a simple user agent that implements the client side of SMTP.
It will allow you to send an e-mail message to an arbitrary recipient via a local
mail server.
2.4.2 Comparison with HTTP
Lets now briefly compare SMTP with HTTP. Both protocols are used to transfer
files from one host to another: HTTP transfers files (also called objects) from a Web
server to a Web client (typically a browser); SMTP transfers files (that is, e-mail
messages) from one mail server to another mail server. When transferring the files,
both persistent HTTP and SMTP use persistent connections. Thus, the two protocols
have common characteristics. However, there are important differences. First,
HTTP is mainly a pull protocolsomeone loads information on a Web server and
users use HTTP to pull the information from the server at their convenience. In par-
ticular, the TCP connection is initiated by the machine that wants to receive the file.
On the other hand, SMTP is primarily a push protocolthe sending mail server
pushes the file to the receiving mail server. In particular, the TCP connection is ini-
tiated by the machine that wants to send the file.
A second difference, which we alluded to earlier, is that SMTP requires
each message, including the body of each message, to be in 7-bit ASCII format.
If the message contains characters that are not 7-bit ASCII (for example, French
characters with accents) or contains binary data (such as an image file), then the
message has to be encoded into 7-bit ASCII. HTTP data does not impose this
restriction.
A third important difference concerns how a document consisting of text and
images (along with possibly other media types) is handled. As we learned in Section
2.2, HTTP encapsulates each object in its own HTTP response message. Internet
mail places all of the messages objects into one message.
124 CHAPTER 2 APPLICATION LAYER
2.4.3 Mail Message Formats
When Alice writes an ordinary snail-mail letter to Bob, she may include all kinds of
peripheral header information at the top of the letter, such as Bobs address, her own
return address, and the date. Similarly, when an e-mail message is sent from one per-
son to another, a header containing peripheral information precedes the body of the
message itself. This peripheral information is contained in a series of header lines,
which are defined in RFC 5322. The header lines and the body of the message are
separated by a blank line (that is, by CRLF). RFC 5322 specifies the exact format for
mail header lines as well as their semantic interpretations. As with HTTP, each header
line contains readable text, consisting of a keyword followed by a colon followed by
a value. Some of the keywords are required and others are optional. Every header
must have a From: header line and a To: header line; a header may include a Sub-
ject: header line as well as other optional header lines. It is important to note that
these header lines are different from the SMTP commands we studied in Section 2.4.1
(even though they contain some common words such as from and to). The com-
mands in that section were part of the SMTP handshaking protocol; the header lines
examined in this section are part of the mail message itself.
A typical message header looks like this:
From: [email protected]
To: [email protected]
Subject: Searching for the meaning of life.
After the message header, a blank line follows; then the message body (in ASCII)
follows. You should use Telnet to send a message to a mail server that contains some
header lines, including the Subject: header line. To do this, issue telnet
serverName 25, as discussed in Section 2.4.1.
2.4.4 Mail Access Protocols
Once SMTP delivers the message from Alices mail server to Bobs mail server, the
message is placed in Bobs mailbox. Throughout this discussion we have tacitly
assumed that Bob reads his mail by logging onto the server host and then executing
a mail reader that runs on that host. Up until the early 1990s this was the standard
way of doing things. But today, mail access uses a client-server architecturethe
typical user reads e-mail with a client that executes on the users end system, for
example, on an office PC, a laptop, or a smartphone. By executing a mail client on a
local PC, users enjoy a rich set of features, including the ability to view multimedia
messages and attachments.
Given that Bob (the recipient) executes his user agent on his local PC, it is nat-
ural to consider placing a mail server on his local PC as well. With this approach,
2.4 ELECTRONIC MAIL IN THE INTERNET 125
Alices mail server would dialogue directly with Bobs PC. There is a problem with
this approach, however. Recall that a mail server manages mailboxes and runs the
client and server sides of SMTP. If Bobs mail server were to reside on his local PC,
then Bobs PC would have to remain always on, and connected to the Internet, in
order to receive new mail, which can arrive at any time. This is impractical for many
Internet users. Instead, a typical user runs a user agent on the local PC but accesses
its mailbox stored on an always-on shared mail server. This mail server is shared
with other users and is typically maintained by the users ISP (for example, univer-
sity or company).
Now lets consider the path an e-mail message takes when it is sent from Alice
to Bob. We just learned that at some point along the path the e-mail message needs
to be deposited in Bobs mail server. This could be done simply by having Alices
user agent send the message directly to Bobs mail server. And this could be done
with SMTPindeed, SMTP has been designed for pushing e-mail from one host to
another. However, typically the senders user agent does not dialogue directly with
the recipients mail server. Instead, as shown in Figure 2.18, Alices user agent uses
SMTP to push the e-mail message into her mail server, then Alices mail server uses
SMTP (as an SMTP client) to relay the e-mail message to Bobs mail server. Why
the two-step procedure? Primarily because without relaying through Alices mail
server, Alices user agent doesnt have any recourse to an unreachable destination
mail server. By having Alice first deposit the e-mail in her own mail server, Alices
mail server can repeatedly try to send the message to Bobs mail server, say every
30 minutes, until Bobs mail server becomes operational. (And if Alices mail server
is down, then she has the recourse of complaining to her system administrator!) The
SMTP RFC defines how the SMTP commands can be used to relay a message
across multiple SMTP servers.
But there is still one missing piece to the puzzle! How does a recipient like Bob,
running a user agent on his local PC, obtain his messages, which are sitting in a mail
server within Bobs ISP? Note that Bobs user agent cant use SMTP to obtain the
messages because obtaining the messages is a pull operation, whereas SMTP is a
126 CHAPTER 2 APPLICATION LAYER
SMTP
Alices
mail server
Bobs
mail server
Alices
agent
Bobs
agentSMTP POP3,
IMAP, or
HTTP
Figure 2.18 E-mail protocols and their communicating entities
push protocol. The puzzle is completed by introducing a special mail access proto-
col that transfers messages from Bobs mail server to his local PC. There are cur-
rently a number of popular mail access protocols, including Post Office
ProtocolVersion 3 (POP3), Internet Mail Access Protocol (IMAP), and HTTP.
Figure 2.18 provides a summary of the protocols that are used for Internet mail:
SMTP is used to transfer mail from the senders mail server to the recipients mail
server; SMTP is also used to transfer mail from the senders user agent to the
senders mail server. A mail access protocol, such as POP3, is used to transfer mail
from the recipients mail server to the recipients user agent.
POP3
POP3 is an extremely simple mail access protocol. It is defined in [RFC 1939], which
is short and quite readable. Because the protocol is so simple, its functionality is
rather limited. POP3 begins when the user agent (the client) opens a TCP connec-
tion to the mail server (the server) on port 110. With the TCP connection estab-
lished, POP3 progresses through three phases: authorization, transaction, and update.
During the first phase, authorization, the user agent sends a username and a password
(in the clear) to authenticate the user. During the second phase, transaction, the user
agent retrieves messages; also during this phase, the user agent can mark messages
for deletion, remove deletion marks, and obtain mail statistics. The third phase,
update, occurs after the client has issued the quit command, ending the POP3
session; at this time, the mail server deletes the messages that were marked for
deletion.
In a POP3 transaction, the user agent issues commands, and the server responds
to each command with a reply. There are two possible responses: +OK (sometimes
followed by server-to-client data), used by the server to indicate that the previous
command was fine; and -ERR, used by the server to indicate that something was
wrong with the previous command.
The authorization phase has two principal commands: user
pass
directly into a POP3 server, using port 110, and issue these commands. Suppose that
mailServer is the name of your mail server. You will see something like:
telnet mailServer 110
+OK POP3 server ready
user bob
+OK
pass hungry
+OK user successfully logged on
If you misspell a command, the POP3 server will reply with an -ERR message.
2.4 ELECTRONIC MAIL IN THE INTERNET 127
Now lets take a look at the transaction phase. A user agent using POP3 can
often be configured (by the user) to download and delete or to download and
keep. The sequence of commands issued by a POP3 user agent depends on which
of these two modes the user agent is operating in. In the download-and-delete mode,
the user agent will issue the list, retr, and dele commands. As an example,
suppose the user has two messages in his or her mailbox. In the dialogue below, C:
(standing for client) is the user agent and S: (standing for server) is the mail server.
The transaction will look something like:
C: list
S: 1 498
S: 2 912
S: .
C: retr 1
S: (blah blah
S: ..
S: .blah)
S: .
C: dele 1
C: retr 2
S: (blah blah
S: ..
S: .blah)
S: .
C: dele 2
C: quit
S: +OK POP3 server signing off
The user agent first asks the mail server to list the size of each of the stored mes-
sages. The user agent then retrieves and deletes each message from the server. Note
that after the authorization phase, the user agent employed only four commands:
list, retr, dele, and quit. The syntax for these commands is defined in RFC
1939. After processing the quit command, the POP3 server enters the update
phase and removes messages 1 and 2 from the mailbox.
A problem with this download-and-delete mode is that the recipient, Bob, may
be nomadic and may want to access his mail messages from multiple machines, for
example, his office PC, his home PC, and his portable computer. The download-
and-delete mode partitions Bobs mail messages over these three machines; in par-
ticular, if Bob first reads a message on his office PC, he will not be able to reread
the message from his portable at home later in the evening. In the download-and-
keep mode, the user agent leaves the messages on the mail server after downloading
them. In this case, Bob can reread messages from different machines; he can access
a message from work and access it again later in the week from home.
128 CHAPTER 2 APPLICATION LAYER
During a POP3 session between a user agent and the mail server, the POP3
server maintains some state information; in particular, it keeps track of which user
messages have been marked deleted. However, the POP3 server does not carry state
information across POP3 sessions. This lack of state information across sessions
greatly simplifies the implementation of a POP3 server.
IMAP
With POP3 access, once Bob has downloaded his messages to the local machine,
he can create mail folders and move the downloaded messages into the folders.
Bob can then delete messages, move messages across folders, and search for
messages (by sender name or subject). But this paradigmnamely, folders and
messages in the local machineposes a problem for the nomadic user, who
would prefer to maintain a folder hierarchy on a remote server that can be
accessed from any computer. This is not possible with POP3the POP3 protocol
does not provide any means for a user to create remote folders and assign mes-
sages to folders.
To solve this and other problems, the IMAP protocol, defined in [RFC 3501],
was invented. Like POP3, IMAP is a mail access protocol. It has many more fea-
tures than POP3, but it is also significantly more complex. (And thus the client and
server side implementations are significantly more complex.)
An IMAP server will associate each message with a folder; when a message first
arrives at the server, it is associated with the recipients INBOX folder. The recipient
can then move the message into a new, user-created folder, read the message, delete
the message, and so on. The IMAP protocol provides commands to allow users to
create folders and move messages from one folder to another. IMAP also provides
commands that allow users to search remote folders for messages matching specific
criteria. Note that, unlike POP3, an IMAP server maintains user state information
across IMAP sessionsfor example, the names of the folders and which messages
are associated with which folders.
Another important feature of IMAP is that it has commands that permit a user
agent to obtain components of messages. For example, a user agent can obtain just
the message header of a message or just one part of a multipart MIME message.
This feature is useful when there is a low-bandwidth connection (for example, a
slow-speed modem link) between the user agent and its mail server. With a low-
bandwidth connection, the user may not want to download all of the messages in
its mailbox, particularly avoiding long messages that might contain, for example,
an audio or video clip.
Web-Based E-Mail
More and more users today are sending and accessing their e-mail through their Web
browsers. Hotmail introduced Web-based access in the mid 1990s. Now Web-based
2.4 ELECTRONIC MAIL IN THE INTERNET 129
e-mail is also provided by Google, Yahoo!, as well as just about every major univer-
sity and corporation. With this service, the user agent is an ordinary Web browser,
and the user communicates with its remote mailbox via HTTP. When a recipient,
such as Bob, wants to access a message in his mailbox, the e-mail message is sent
from Bobs mail server to Bobs browser using the HTTP protocol rather than the
POP3 or IMAP protocol. When a sender, such as Alice, wants to send an e-mail
message, the e-mail message is sent from her browser to her mail server over HTTP
rather than over SMTP. Alices mail server, however, still sends messages to, and
receives messages from, other mail servers using SMTP.
2.5 DNSThe Internets Directory Service
We human beings can be identified in many ways. For example, we can be identi-
fied by the names that appear on our birth certificates. We can be identified by our
social security numbers. We can be identified by our drivers license numbers.
Although each of these identifiers can be used to identify people, within a given
context one identifier may be more appropriate than another. For example, the com-
puters at the IRS (the infamous tax-collecting agency in the United States) prefer to
use fixed-length social security numbers rather than birth certificate names. On the
other hand, ordinary people prefer the more mnemonic birth certificate names rather
than social security numbers. (Indeed, can you imagine saying, Hi. My name is
132-67-9875. Please meet my husband, 178-87-1146.)
Just as humans can be identified in many ways, so too can Internet hosts. One identi-
fier for a host is its hostname. Hostnamessuch as cnn.com, www.yahoo.
com, gaia.cs.umass.edu, and cis.poly.eduare mnemonic and are there-
fore appreciated by humans. However, hostnames provide little, if any, information about
the location within the Internet of the host. (A hostname such as www.eurecom.fr,
which ends with the country code .fr, tells us that the host is probably in France, but
doesnt say much more.) Furthermore, because hostnames can consist of variable-
length alphanumeric characters, they would be difficult to process by routers. For these
reasons, hosts are also identified by so-called IP addresses.
We discuss IP addresses in some detail in Chapter 4, but it is useful to say a few
brief words about them now. An IP address consists of four bytes and has a rigid
hierarchical structure. An IP address looks like 121.7.106.83, where each
period separates one of the bytes expressed in decimal notation from 0 to 255. An IP
address is hierarchical because as we scan the address from left to right, we obtain
more and more specific information about where the host is located in the Internet
(that is, within which network, in the network of networks). Similarly, when we scan
a postal address from bottom to top, we obtain more and more specific information
about where the addressee is located.
130 CHAPTER 2 APPLICATION LAYER
www.yahoo.com,gaia.cs.umass.edu
www.yahoo.com,gaia.cs.umass.edu
www.eurecom.fr
2.5.1 Services Provided by DNS
We have just seen that there are two ways to identify a hostby a hostname and by
an IP address. People prefer the more mnemonic hostname identifier, while routers
prefer fixed-length, hierarchically structured IP addresses. In order to reconcile
these preferences, we need a directory service that translates hostnames to IP
addresses. This is the main task of the Internets domain name system (DNS). The
DNS is (1) a distributed database implemented in a hierarchy of DNS servers, and
(2) an application-layer protocol that allows hosts to query the distributed database.
The DNS servers are often UNIX machines running the Berkeley Internet Name
Domain (BIND) software [BIND 2012]. The DNS protocol runs over UDP and uses
port 53.
DNS is commonly employed by other application-layer protocolsincluding
HTTP, SMTP, and FTPto translate user-supplied hostnames to IP addresses. As
an example, consider what happens when a browser (that is, an HTTP client),
running on some users host, requests the URL www.someschool.edu/
index.html. In order for the users host to be able to send an HTTP request mes-
sage to the Web server www.someschool.edu, the users host must first obtain
the IP address of www.someschool.edu. This is done as follows.
1. The same user machine runs the client side of the DNS application.
2. The browser extracts the hostname, www.someschool.edu, from the URL
and passes the hostname to the client side of the DNS application.
3. The DNS client sends a query containing the hostname to a DNS server.
4. The DNS client eventually receives a reply, which includes the IP address for
the hostname.
5. Once the browser receives the IP address from DNS, it can initiate a TCP con-
nection to the HTTP server process located at port 80 at that IP address.
We see from this example that DNS adds an additional delaysometimes substan-
tialto the Internet applications that use it. Fortunately, as we discuss below, the
desired IP address is often cached in a nearby DNS server, which helps to reduce
DNS network traffic as well as the average DNS delay.
DNS provides a few other important services in addition to translating host-
names to IP addresses:
Host aliasing. A host with a complicated hostname can have one or more alias
names. For example, a hostname such as relay1.west-coast.enter-
prise.com could have, say, two aliases such as enterprise.com and
www.enterprise.com. In this case, the hostname relay1.west-
coast.enterprise.com is said to be a canonical hostname. Alias host-
names, when present, are typically more mnemonic than canonical hostnames.
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 131
www.someschool.edu/index.html
www.someschool.edu/index.html
www.someschool.edu
www.someschool.edu
www.someschool.edu
www.enterprise.com
DNS can be invoked by an application to obtain the canonical hostname for a
supplied alias hostname as well as the IP address of the host.
Mail server aliasing. For obvious reasons, it is highly desirable that e-mail
addresses be mnemonic. For example, if Bob has an account with Hotmail, Bobs
e-mail address might be as simple as [email protected]. However, the host-
name of the Hotmail mail server is more complicated and much less mnemonic
than simply hotmail.com (for example, the canonical hostname might be
something like relay1.west-coast.hotmail.com). DNS can be
invoked by a mail application to obtain the canonical hostname for a supplied
alias hostname as well as the IP address of the host. In fact, the MX record (see
below) permits a companys mail server and Web server to have identical
(aliased) hostnames; for example, a companys Web server and mail server can
both be called enterprise.com.
Load distribution. DNS is also used to perform load distribution among repli-
cated servers, such as replicated Web servers. Busy sites, such as cnn.com, are
replicated over multiple servers, with each server running on a different end sys-
tem and each having a different IP address. For replicated Web servers, a set of
IP addresses is thus associated with one canonical hostname. The DNS database
contains this set of IP addresses. When clients make a DNS query for a name
mapped to a set of addresses, the server responds with the entire set of IP
addresses, but rotates the ordering of the addresses within each reply. Because a
client typically sends its HTTP request message to the IP address that is listed
first in the set, DNS rotation distributes the traffic among the replicated servers.
132 CHAPTER 2 APPLICATION LAYER
DNS: CRITICAL NETWORK FUNCTIONS VIA THE CLIENT-SERVER
PARADIGM
Like HTTP, FTP, and SMTP, the DNS protocol is an application-layer protocol since it (1) runs
between communicating end systems using the client-server paradigm and (2) relies on an
underlying end-to-end transport protocol to transfer DNS messages between communicating
end systems. In another sense, however, the role of the DNS is quite different from Web,
file transfer, and e-mail applications. Unlike these applications, the DNS is not an applica-
tion with which a user directly interacts. Instead, the DNS provides a core Internet func-
tionnamely, translating hostnames to their underlying IP addresses, for user applications
and other software in the Internet. We noted in Section 1.2 that much of the complexity in
the Internet architecture is located at the edges of the network. The DNS, which imple-
ments the critical name-to-address translation process using clients and servers located at
the edge of the network, is yet another example of that design philosophy.
PRINCIPLES IN PRACTICE
DNS rotation is also used for e-mail so that multiple mail servers can have the
same alias name. Also, content distribution companies such as Akamai have used
DNS in more sophisticated ways [Dilley 2002] to provide Web content distribu-
tion (see Chapter 7).
The DNS is specified in RFC 1034 and RFC 1035, and updated in several
additional RFCs. It is a complex system, and we only touch upon key aspects of
its operation here. The interested reader is referred to these RFCs and the book
by Albitz and Liu [Albitz 1993]; see also the retrospective paper [Mockapetris
1988], which provides a nice description of the what and why of DNS, and
[Mockapetris 2005].
2.5.2 Overview of How DNS Works
We now present a high-level overview of how DNS works. Our discussion will
focus on the hostname-to-IP-address translation service.
Suppose that some application (such as a Web browser or a mail reader) run-
ning in a users host needs to translate a hostname to an IP address. The applica-
tion will invoke the client side of DNS, specifying the hostname that needs to be
translated. (On many UNIX-based machines, gethostbyname() is the func-
tion call that an application calls in order to perform the translation.) DNS in the
users host then takes over, sending a query message into the network. All DNS
query and reply messages are sent within UDP datagrams to port 53. After a delay,
ranging from milliseconds to seconds, DNS in the users host receives a DNS
reply message that provides the desired mapping. This mapping is then passed to
the invoking application. Thus, from the perspective of the invoking application
in the users host, DNS is a black box providing a simple, straightforward transla-
tion service. But in fact, the black box that implements the service is complex,
consisting of a large number of DNS servers distributed around the globe, as well
as an application-layer protocol that specifies how the DNS servers and querying
hosts communicate.
A simple design for DNS would have one DNS server that contains all the map-
pings. In this centralized design, clients simply direct all queries to the single DNS
server, and the DNS server responds directly to the querying clients. Although the
simplicity of this design is attractive, it is inappropriate for todays Internet, with its
vast (and growing) number of hosts. The problems with a centralized design
include:
A single point of failure. If the DNS server crashes, so does the entire Internet!
Traffic volume. A single DNS server would have to handle all DNS queries (for
all the HTTP requests and e-mail messages generated from hundreds of millions
of hosts).
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 133
Distant centralized database. A single DNS server cannot be close to all the
querying clients. If we put the single DNS server in New York City, then all
queries from Australia must travel to the other side of the globe, perhaps over
slow and congested links. This can lead to significant delays.
Maintenance. The single DNS server would have to keep records for all Internet
hosts. Not only would this centralized database be huge, but it would have to be
updated frequently to account for every new host.
In summary, a centralized database in a single DNS server simply doesnt scale.
Consequently, the DNS is distributed by design. In fact, the DNS is a wonderful
example of how a distributed database can be implemented in the Internet.
A Distributed, Hierarchical Database
In order to deal with the issue of scale, the DNS uses a large number of servers,
organized in a hierarchical fashion and distributed around the world. No single DNS
server has all of the mappings for all of the hosts in the Internet. Instead, the map-
pings are distributed across the DNS servers. To a first approximation, there are
three classes of DNS serversroot DNS servers, top-level domain (TLD) DNS
servers, and authoritative DNS serversorganized in a hierarchy as shown in Fig-
ure 2.19. To understand how these three classes of servers interact, suppose a DNS
client wants to determine the IP address for the hostname www.amazon.com. To
a first approximation, the following events will take place. The client first contacts
one of the root servers, which returns IP addresses for TLD servers for the top-level
domain com. The client then contacts one of these TLD servers, which returns the
IP address of an authoritative server for amazon.com. Finally, the client contacts
one of the authoritative servers for amazon.com, which returns the IP address
134 CHAPTER 2 APPLICATION LAYER
edu DNS serversorg DNS serverscom DNS servers
poly.edu
DNS servers
yahoo.com
DNS servers
amazon.com
DNS servers
pbs.org
DNS servers
umass.edu
DNS servers
Root DNS servers
Figure 2.19 Portion of the hierarchy of DNS servers
www.amazon.com
for the hostname www.amazon.com. Well soon examine this DNS lookup
process in more detail. But lets first take a closer look at these three classes of
DNS servers:
Root DNS servers. In the Internet there are 13 root DNS servers (labeled A
through M), most of which are located in North America. An October 2006 map
of the root DNS servers is shown in Figure 2.20; a list of the current root DNS
servers is available via [Root-servers 2012]. Although we have referred to each
of the 13 root DNS servers as if it were a single server, each server is actually
a network of replicated servers, for both security and reliability purposes. All
together, there are 247 root servers as of fall 2011.
Top-level domain (TLD) servers. These servers are responsible for top-level
domains such as com, org, net, edu, and gov, and all of the country top-level domains
such as uk, fr, ca, and jp. The company Verisign Global Registry Services
maintains the TLD servers for the com top-level domain, and the company
Educause maintains the TLD servers for the edu top-level domain. See [IANA
TLD 2012] for a list of all top-level domains.
Authoritative DNS servers. Every organization with publicly accessible hosts
(such as Web servers and mail servers) on the Internet must provide publicly acces-
sible DNS records that map the names of those hosts to IP addresses. An organiza-
tions authoritative DNS server houses these DNS records. An organization can
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 135
c.
d.
h.
j.
Cogent, Herndon, VA (5 other sites)
U Maryland College Park, MD
ARL Aberdeen, MD
Verisign, Dulles VA (69 other sites )
i. Netnod, Stockholm
(37 other sites)
k. RIPE London
(17 other sites)
m. WIDE Tokyo
(5 other sites)
g. US DoD Columbus, OH
(5 other sites)
e.
f.
NASA Mt View, CA
Internet Software C.
Palo Alto, CA
(and 48 other sites)
a.
b.
l.
Verisign, Los Angeles CA
(5 other sites)
USC-ISI Marina del Rey, CA
ICANN Los Angeles, CA
(41 other sites)
Figure 2.20 DNS root servers in 2012 (name, organization, location)
www.amazon.com
choose to implement its own authoritative DNS server to hold these records; alter-
natively, the organization can pay to have these records stored in an authoritative
DNS server of some service provider. Most universities and large companies
implement and maintain their own primary and secondary (backup) authoritative
DNS server.
The root, TLD, and authoritative DNS servers all belong to the hierarchy of
DNS servers, as shown in Figure 2.19. There is another important type of DNS
server called the local DNS server. A local DNS server does not strictly belong to
the hierarchy of servers but is nevertheless central to the DNS architecture. Each
ISPsuch as a university, an academic department, an employees company, or a
residential ISPhas a local DNS server (also called a default name server). When a
host connects to an ISP, the ISP provides the host with the IP addresses of one or
more of its local DNS servers (typically through DHCP, which is discussed in Chap-
ter 4). You can easily determine the IP address of your local DNS server by access-
ing network status windows in Windows or UNIX. A hosts local DNS server is
typically close to the host. For an institutional ISP, the local DNS server may be
on the same LAN as the host; for a residential ISP, it is typically separated from the
host by no more than a few routers. When a host makes a DNS query, the query is
sent to the local DNS server, which acts a proxy, forwarding the query into the DNS
server hierarchy, as well discuss in more detail below.
Lets take a look at a simple example. Suppose the host cis.poly.edu
desires the IP address of gaia.cs.umass.edu. Also suppose that Polytechnics
local DNS server is called dns.poly.edu and that an authoritative DNS server
for gaia.cs.umass.edu is called dns.umass.edu. As shown in Figure
2.21, the host cis.poly.edu first sends a DNS query message to its local DNS
server, dns.poly.edu. The query message contains the hostname to be trans-
lated, namely, gaia.cs.umass.edu. The local DNS server forwards the query
message to a root DNS server. The root DNS server takes note of the edu suffix and
returns to the local DNS server a list of IP addresses for TLD servers responsible for
edu. The local DNS server then resends the query message to one of these TLD
servers. The TLD server takes note of the umass.edu suffix and responds with the
IP address of the authoritative DNS server for the University of Massachusetts,
namely, dns.umass.edu. Finally, the local DNS server resends the query
message directly to dns.umass.edu, which responds with the IP address of
gaia.cs.umass.edu. Note that in this example, in order to obtain the mapping
for one hostname, eight DNS messages were sent: four query messages and four
reply messages! Well soon see how DNS caching reduces this query traffic.
Our previous example assumed that the TLD server knows the authoritative
DNS server for the hostname. In general this not always true. Instead, the TLD server
may know only of an intermediate DNS server, which in turn knows the authoritative
DNS server for the hostname. For example, suppose again that the University of
136 CHAPTER 2 APPLICATION LAYER
Massachusetts has a DNS server for the university, called dns.umass.edu. Also
suppose that each of the departments at the University of Massachusetts has its own
DNS server, and that each departmental DNS server is authoritative for all hosts in
the department. In this case, when the intermediate DNS server, dns.umass.edu,
receives a query for a host with a hostname ending with cs.umass.edu, it returns
to dns.poly.edu the IP address of dns.cs.umass.edu, which is authorita-
tive for all hostnames ending with cs.umass.edu. The local DNS server
dns.poly.edu then sends the query to the authoritative DNS server, which
returns the desired mapping to the local DNS server, which in turn returns the map-
ping to the requesting host. In this case, a total of 10 DNS messages are sent!
The example shown in Figure 2.21 makes use of both recursive queries and
iterative queries. The query sent from cis.poly.edu to dns.poly.edu is a
recursive query, since the query asks dns.poly.edu to obtain the mapping on its
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 137
Requesting host
cis.poly.edu
Local DNS server TLD DNS server
dns.poly.edu
Root DNS server
1
8
2
7
4
5
3
6
Authoritative DNS server
dns.umass.edu
gaia.cs.umass.edu
Figure 2.21 Interaction of the various DNS servers
behalf. But the subsequent three queries are iterative since all of the replies are
directly returned to dns.poly.edu. In theory, any DNS query can be iterative or
recursive. For example, Figure 2.22 shows a DNS query chain for which all of
the queries are recursive. In practice, the queries typically follow the pattern in
Figure 2.21: The query from the requesting host to the local DNS server is recur-
sive, and the remaining queries are iterative.
DNS Caching
Our discussion thus far has ignored DNS caching, a critically important feature of the
DNS system. In truth, DNS extensively exploits DNS caching in order to improve
the delay performance and to reduce the number of DNS messages ricocheting around
138 CHAPTER 2 APPLICATION LAYER
Requesting host
cis.poly.edu
Local DNS server TLD DNS server
dns.poly.edu
Root DNS server
1
8
5
4
2
7
Authoritative DNS server
dns.umass.edu
gaia.cs.umass.edu
6 3
Figure 2.22 Recursive queries in DNS
the Internet. The idea behind DNS caching is very simple. In a query chain, when a
DNS server receives a DNS reply (containing, for example, a mapping from a host-
name to an IP address), it can cache the mapping in its local memory. For example,
in Figure 2.21, each time the local DNS server dns.poly.edu receives a reply
from some DNS server, it can cache any of the information contained in the reply. If a
hostname/IP address pair is cached in a DNS server and another query arrives to the
DNS server for the same hostname, the DNS server can provide the desired IP address,
even if it is not authoritative for the hostname. Because hosts and mappings between
hostnames and IP addresses are by no means permanent, DNS servers discard cached
information after a period of time (often set to two days).
As an example, suppose that a host apricot.poly.edu queries
dns.poly.edu for the IP address for the hostname cnn.com. Furthermore, sup-
pose that a few hours later, another Polytechnic University host, say, kiwi.poly.fr,
also queries dns.poly.edu with the same hostname. Because of caching, the local
DNS server will be able to immediately return the IP address of cnn.com to this sec-
ond requesting host without having to query any other DNS servers. A local DNS
server can also cache the IP addresses of TLD servers, thereby allowing the local DNS
server to bypass the root DNS servers in a query chain (this often happens).
2.5.3 DNS Records and Messages
The DNS servers that together implement the DNS distributed database store
resource records (RRs), including RRs that provide hostname-to-IP address map-
pings. Each DNS reply message carries one or more resource records. In this and
the following subsection, we provide a brief overview of DNS resource records and
messages; more details can be found in [Abitz 1993] or in the DNS RFCs [RFC
1034; RFC 1035].
A resource record is a four-tuple that contains the following fields:
(Name, Value, Type, TTL)
TTL is the time to live of the resource record; it determines when a resource should
be removed from a cache. In the example records given below, we ignore the TTL
field. The meaning of Name and Value depend on Type:
If Type=A, then Name is a hostname and Value is the IP address for the host-
name. Thus, a Type A record provides the standard hostname-to-IP address map-
ping. As an example, (relay1.bar.foo.com, 145.37.93.126, A)
is a Type A record.
If Type=NS, then Name is a domain (such as foo.com) and Value is the host-
name of an authoritative DNS server that knows how to obtain the IP addresses
for hosts in the domain. This record is used to route DNS queries further along in
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 139
the query chain. As an example, (foo.com, dns.foo.com, NS) is a Type
NS record.
If Type=CNAME, then Value is a canonical hostname for the alias hostname
Name. This record can provide querying hosts the canonical name for a host-
name. As an example, (foo.com, relay1.bar.foo.com, CNAME) is a
CNAME record.
If Type=MX, then Value is the canonical name of a mail server that has an alias
hostname Name. As an example, (foo.com, mail.bar.foo.com, MX)
is an MX record. MX records allow the hostnames of mail servers to have sim-
ple aliases. Note that by using the MX record, a company can have the same
aliased name for its mail server and for one of its other servers (such as its Web
server). To obtain the canonical name for the mail server, a DNS client would
query for an MX record; to obtain the canonical name for the other server, the
DNS client would query for the CNAME record.
If a DNS server is authoritative for a particular hostname, then the DNS server will
contain a Type A record for the hostname. (Even if the DNS server is not authoritative,
it may contain a Type A record in its cache.) If a server is not authoritative for a host-
name, then the server will contain a Type NS record for the domain that includes the
hostname; it will also contain a Type A record that provides the IP address of the DNS
server in the Value field of the NS record. As an example, suppose an edu TLD server
is not authoritative for the host gaia.cs.umass.edu. Then this server will contain
a record for a domain that includes the host gaia.cs.umass.edu, for example,
(umass.edu, dns.umass.edu, NS). The edu TLD server would also contain
a Type A record, which maps the DNS server dns.umass.edu to an IP address, for
example, (dns.umass.edu, 128.119.40.111, A).
DNS Messages
Earlier in this section, we referred to DNS query and reply messages. These are the
only two kinds of DNS messages. Furthermore, both query and reply messages have
the same format, as shown in Figure 2.23.The semantics of the various fields in a
DNS message are as follows:
The first 12 bytes is the header section, which has a number of fields. The first field
is a 16-bit number that identifies the query. This identifier is copied into the reply
message to a query, allowing the client to match received replies with sent queries.
There are a number of flags in the flag field. A 1-bit query/reply flag indicates
whether the message is a query (0) or a reply (1). A 1-bit authoritative flag is set in a
reply message when a DNS server is an authoritative server for a queried name. A
1-bit recursion-desired flag is set when a client (host or DNS server) desires that the
DNS server perform recursion when it doesnt have the record. A 1-bit recursion-
available field is set in a reply if the DNS server supports recursion. In the header,
140 CHAPTER 2 APPLICATION LAYER
there are also four number-of fields. These fields indicate the number of occurrences
of the four types of data sections that follow the header.
The question section contains information about the query that is being made.
This section includes (1) a name field that contains the name that is being
queried, and (2) a type field that indicates the type of question being asked about
the namefor example, a host address associated with a name (Type A) or the
mail server for a name (Type MX).
In a reply from a DNS server, the answer section contains the resource records
for the name that was originally queried. Recall that in each resource record there
is the Type (for example, A, NS, CNAME, and MX), the Value, and the TTL.
A reply can return multiple RRs in the answer, since a hostname can have multi-
ple IP addresses (for example, for replicated Web servers, as discussed earlier in
this section).
The authority section contains records of other authoritative servers.
The additional section contains other helpful records. For example, the answer
field in a reply to an MX query contains a resource record providing the canoni-
cal hostname of a mail server. The additional section contains a Type A record
providing the IP address for the canonical hostname of the mail server.
How would you like to send a DNS query message directly from the host
youre working on to some DNS server? This can easily be done with the nslookup
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 141
Identification
Number of questions
Number of authority RRs
Name, type fields for
a query
12 bytes
RRs in response to query
Records for
authoritative servers
Additional helpful
info that may be used
Flags
Number of answer RRs
Number of additional RRs
Authority
(variable number of resource records)
Additional information
(variable number of resource records)
Answers
(variable number of resource records)
Questions
(variable number of questions)
Figure 2.23 DNS message format
program, which is available from most Windows and UNIX platforms. For exam-
ple, from a Windows host, open the Command Prompt and invoke the nslookup pro-
gram by simply typing nslookup. After invoking nslookup, you can send a DNS
query to any DNS server (root, TLD, or authoritative). After receiving the reply
message from the DNS server, nslookup will display the records included in the
reply (in a human-readable format). As an alternative to running nslookup from your
own host, you can visit one of many Web sites that allow you to remotely employ
nslookup. (Just type nslookup into a search engine and youll be brought to one of
these sites.) The DNS Wireshark lab at the end of this chapter will allow you to
explore the DNS in much more detail.
Inserting Records into the DNS Database
The discussion above focused on how records are retrieved from the DNS database.
You might be wondering how records get into the database in the first place. Lets look
at how this is done in the context of a specific example. Suppose you have just created
an exciting new startup company called Network Utopia. The first thing youll surely
want to do is register the domain name networkutopia.com at a registrar. A
registrar is a commercial entity that verifies the uniqueness of the domain name,
enters the domain name into the DNS database (as discussed below), and collects a
small fee from you for its services. Prior to 1999, a single registrar, Network Solutions,
had a monopoly on domain name registration for com, net, and org domains. But
now there are many registrars competing for customers, and the Internet Corporation
for Assigned Names and Numbers (ICANN) accredits the various registrars. A com-
plete list of accredited registrars is available at http://www.internic.net.
When you register the domain name networkutopia.com with some reg-
istrar, you also need to provide the registrar with the names and IP addresses of your
primary and secondary authoritative DNS servers. Suppose the names and IP
addresses are dns1.networkutopia.com, dns2.networkutopia.com,
212.212.212.1, and 212.212.212.2. For each of these two authoritative
DNS servers, the registrar would then make sure that a Type NS and a Type A record
are entered into the TLD com servers. Specifically, for the primary authoritative
server for networkutopia.com, the registrar would insert the following two
resource records into the DNS system:
(networkutopia.com, dns1.networkutopia.com, NS)
(dns1.networkutopia.com, 212.212.212.1, A)
Youll also have to make sure that the Type A resource record for your Web server
www.networkutopia.com and the Type MX resource record for your mail
server mail.networkutopia.com are entered into your authoritative DNS
servers. (Until recently, the contents of each DNS server were configured statically,
142 CHAPTER 2 APPLICATION LAYER
http://www.internic.net
www.networkutopia.com
2.5 DNSTHE INTERNETS DIRECTORY SERVICE 143
DNS VULNERABIL IT IES
We have seen that DNS is a critical component of the Internet infrastructure, with
many important services including the Web and e-mail simply incapable of func-
tioning without it. We therefore naturally ask, how can DNS be attacked? Is DNS a
sitting duck, waiting to be knocked out of service, while taking most Internet applica-
tions down with it?
The first type of attack that comes to mind is a DDoS bandwidth-flooding attack (see
Section 1.6) against DNS servers. For example, an attacker could attempt to send to
each DNS root server a deluge of packets, so many that the majority of legitimate DNS
queries never get answered. Such a large-scale DDoS attack against DNS root servers
actually took place on October 21, 2002. In this attack, the attackers leveraged a bot-
net to send truck loads of ICMP ping messages to each of the 13 DNS root servers.
(ICMP messages are discussed in Chapter 4. For now, it suffices to know that ICMP pack-
ets are special types of IP datagrams.) Fortunately, this large-scale attack caused minimal
damage, having little or no impact on users Internet experience. The attackers did
succeed at directing a deluge of packets at the root servers. But many of the DNS root
servers were protected by packet filters, configured to always block all ICMP ping
messages directed at the root servers. These protected servers were thus spared and
functioned as normal. Furthermore, most local DNS servers cache the IP addresses of top-
level-domain servers, allowing the query process to often bypass the DNS root servers.
A potentially more effective DDoS attack against DNS would be send a deluge of
DNS queries to top-level-domain servers, for example, to all the top-level-domain
servers that handle the .com domain. It would be harder to filter DNS queries direct-
ed to DNS servers; and top-level-domain servers are not as easily bypassed as are
root servers. But the severity of such an attack would be partially mitigated by
caching in local DNS servers.
DNS could potentially be attacked in other ways. In a man-in-the-middle attack,
the attacker intercepts queries from hosts and returns bogus replies. In the DNS poi-
soning attack, the attacker sends bogus replies to a DNS server, tricking the server
into accepting bogus records into its cache. Either of these attacks could be used, for
example, to redirect an unsuspecting Web user to the attackers Web site. These
attacks, however, are difficult to implement, as they require intercepting packets or
throttling servers [Skoudis 2006].
Another important DNS attack is not an attack on the DNS service per se, but
instead exploits the DNS infrastructure to launch a DDoS attack against a targeted host
(for example, your universitys mail server). In this attack, the attacker sends DNS
queries to many authoritative DNS servers, with each query having the spoofed source
address of the targeted host. The DNS servers then send their replies directly to the tar-
geted host. If the queries can be crafted in such a way that a response is much larger
FOCUS ON SECURITY
for example, from a configuration file created by a system manager. More recently,
an UPDATE option has been added to the DNS protocol to allow data to be dynam-
ically added or deleted from the database via DNS messages. [RFC 2136] and [RFC
3007] specify DNS dynamic updates.)
Once all of these steps are completed, people will be able to visit your Web site
and send e-mail to the employees at your company. Lets conclude our discussion of
DNS by verifying that this statement is true. This verification also helps to solidify
what we have learned about DNS. Suppose Alice in Australia wants to view the Web
page www.networkutopia.com. As discussed earlier, her host will first send a
DNS query to her local DNS server. The local DNS server will then contact a TLD
com server. (The local DNS server will also have to contact a root DNS server if the
address of a TLD com server is not cached.) This TLD server contains the Type NS
and Type A resource records listed above, because the registrar had these resource
records inserted into all of the TLD com servers. The TLD com server sends a reply
to Alices local DNS server, with the reply containing the two resource records. The
local DNS server then sends a DNS query to 212.212.212.1, asking for the
Type A record corresponding to www.networkutopia.com. This record pro-
vides the IP address of the desired Web server, say, 212.212.71.4, which the
local DNS server passes back to Alices host. Alices browser can now initiate a TCP
connection to the host 212.212.71.4 and send an HTTP request over the con-
nection. Whew! Theres a lot more going on than what meets the eye when one surfs
the Web!
2.6 Peer-to-Peer Applications
The applications described in this chapter thus farincluding the Web, e-mail, and
DNSall employ client-server architectures with significant reliance on always-on
infrastructure servers. Recall from Section 2.1.1 that with a P2P architecture, there
is minimal (or no) reliance on always-on infrastructure servers. Instead, pairs of
intermittently connected hosts, called peers, communicate directly with each other.
144 CHAPTER 2 APPLICATION LAYER
(in bytes) than a query (so-called amplification), then the attacker can potentially over-
whelm the target without having to generate much of its own traffic. Such reflection
attacks exploiting DNS have had limited success to date [Mirkovic 2005].
In summary, DNS has demonstrated itself to be surprisingly robust against attacks.
To date, there hasnt been an attack that has successfully impeded the DNS service.
There have been successful reflector attacks; however, these attacks can be (and are
being) addressed by appropriate configuration of DNS servers.
FOCUS ON SECURITY
www.networkutopia.com
www.networkutopia.com
The peers are not owned by a service provider, but are instead desktops and laptops
controlled by users.
In this section well examine two different applications that are particularly
well-suited for P2P designs. The first is file distribution, where the application dis-
tributes a file from a single source to a large number of peers. File distribution is a
nice place to start our investigation of P2P, as it clearly exposes the self-scalability
of P2P architectures. As a specific example for file distribution, well describe
the popular BitTorrent system. The second P2P application well examine is a
database distributed over a large community of peers. For this application, well
explore the concept of a Distributed Hash Table (DHT).
2.6.1 P2P File Distribution
We begin our foray into P2P by considering a very natural application, namely,
distributing a large file from a single server to a large number of hosts (called
peers). The file might be a new version of the Linux operating system, a software
patch for an existing operating system or application, an MP3 music file, or an
MPEG video file. In client-server file distribution, the server must send a copy of
the file to each of the peersplacing an enormous burden on the server and con-
suming a large amount of server bandwidth. In P2P file distribution, each peer can
redistribute any portion of the file it has received to any other peers, thereby
assisting the server in the distribution process. As of 2012, the most popular P2P
file distribution protocol is BitTorrent. Originally developed by Bram Cohen,
there are now many different independent BitTorrent clients conforming to the
BitTorrent protocol, just as there are a number of Web browser clients that
conform to the HTTP protocol. In this subsection, we first examine the self-
scalability of P2P architectures in the context of file distribution. We then describe
BitTorrent in some detail, highlighting its most important characteristics and
features.
Scalability of P2P Architectures
To compare client-server architectures with peer-to-peer architectures, and illustrate
the inherent self-scalability of P2P, we now consider a simple quantitative model for
distributing a file to a fixed set of peers for both architecture types. As shown in Fig-
ure 2.24, the server and the peers are connected to the Internet with access links.
Denote the upload rate of the servers access link by us, the upload rate of the ith
peers access link by ui, and the download rate of the ith peers access link by di.
Also denote the size of the file to be distributed (in bits) by F and the number of
peers that want to obtain a copy of the file by N. The distribution time is the time it
takes to get a copy of the file to all N peers. In our analysis of the distribution time
below, for both client-server and P2P architectures, we make the simplifying (and
generally accurate [Akella 2003]) assumption that the Internet core has abundant
2.6 PEER-TO-PEER APPLICATIONS 145
bandwidth, implying that all of the bottlenecks are in access networks. We also sup-
pose that the server and clients are not participating in any other network applica-
tions, so that all of their upload and download access bandwidth can be fully
devoted to distributing this file.
Lets first determine the distribution time for the client-server architecture,
which we denote by Dcs. In the client-server architecture, none of the peers aids in
distributing the file. We make the following observations:
The server must transmit one copy of the file to each of the N peers. Thus the
server must transmit NF bits. Since the servers upload rate is us, the time to dis-
tribute the file must be at least NF/us.
Let dmin denote the download rate of the peer with the lowest download rate, that
is, dmin = min{d1,dp,,dN}. The peer with the lowest download rate cannot
obtain all F bits of the file in less than F/dmin seconds. Thus the minimum distri-
bution time is at least F/dmin.
Putting these two observations together, we obtain
.Dcs maxbNFus , Fdmin r
146 CHAPTER 2 APPLICATION LAYER
Internet
File: F
Server
us
u1 u2
u3
d1
d2
d3
u4
u5u6
d4
d5
d6
uN
dN
Figure 2.24 An illustrative file distribution problem
This provides a lower bound on the minimum distribution time for the client-server
architecture. In the homework problems you will be asked to show that the server
can schedule its transmissions so that the lower bound is actually achieved. So lets
take this lower bound provided above as the actual distribution time, that is,
(2.1)
We see from Equation 2.1 that for N large enough, the client-server distribution time
is given by NF/us. Thus, the distribution time increases linearly with the number of
peers N. So, for example, if the number of peers from one week to the next increases
a thousand-fold from a thousand to a million, the time required to distribute the file
to all peers increases by 1,000.
Lets now go through a similar analysis for the P2P architecture, where each
peer can assist the server in distributing the file. In particular, when a peer receives
some file data, it can use its own upload capacity to redistribute the data to other
peers. Calculating the distribution time for the P2P architecture is somewhat more
complicated than for the client-server architecture, since the distribution time
depends on how each peer distributes portions of the file to the other peers. Never-
theless, a simple expression for the minimal distribution time can be obtained
[Kumar 2006]. To this end, we first make the following observations:
At the beginning of the distribution, only the server has the file. To get this file
into the community of peers, the server must send each bit of the file at least once
into its access link. Thus, the minimum distribution time is at least F/us. (Unlike
the client-server scheme, a bit sent once by the server may not have to be sent by
the server again, as the peers may redistribute the bit among themselves.)
As with the client-server architecture, the peer with the lowest download rate
cannot obtain all F bits of the file in less than F/dmin seconds. Thus the minimum
distribution time is at least F/dmin.
Finally, observe that the total upload capacity of the system as a whole is equal
to the upload rate of the server plus the upload rates of each of the individual
peers, that is, utotal = us + u1 + + uN. The system must deliver (upload) F bits
to each of the N peers, thus delivering a total of NF bits. This cannot be done at a
rate faster than utotal. Thus, the minimum distribution time is also at least
NF/(us + u1 + + uN).
Putting these three observations together, we obtain the minimum distribution time
for P2P, denoted by DP2P.
(2.2)DP2P max c Fus, Fdmin, NF
us + a
N
i=1
ui
s
Dcs = maxbNFus , Fdmin r
2.6 PEER-TO-PEER APPLICATIONS 147
Equation 2.2 provides a lower bound for the minimum distribution time for the P2P
architecture. It turns out that if we imagine that each peer can redistribute a bit as
soon as it receives the bit, then there is a redistribution scheme that actually achieves
this lower bound [Kumar 2006]. (We will prove a special case of this result in the
homework.) In reality, where chunks of the file are redistributed rather than individ-
ual bits, Equation 2.2 serves as a good approximation of the actual minimum distri-
bution time. Thus, lets take the lower bound provided by Equation 2.2 as the actual
minimum distribution time, that is,
(2.3)
Figure 2.25 compares the minimum distribution time for the client-server and
P2P architectures assuming that all peers have the same upload rate u. In Figure
2.25, we have set F/u = 1 hour, us = 10u, and dmin us. Thus, a peer can transmit the
entire file in one hour, the server transmission rate is 10 times the peer upload rate,
and (for simplicity) the peer download rates are set large enough so as not to have
an effect. We see from Figure 2.25 that for the client-server architecture, the dis-
tribution time increases linearly and without bound as the number of peers
increases. However, for the P2P architecture, the minimal distribution time is not
only always less than the distribution time of the client-server architecture; it is also
less than one hour for any number of peers N. Thus, applications with the P2P
architecture can be self-scaling. This scalability is a direct consequence of peers
being redistributors as well as consumers of bits.
DP2P = max c Fus, Fdmin, NF
us + a
N
i=1
ui
s
148 CHAPTER 2 APPLICATION LAYER
0
5 10 15 20 25 300
N
M
in
im
u
m
d
is
tr
ib
u
ti
o
in
t
iim
e
35
0.5
1.5
2.5
1.0
3.0
2.0
3.5
Client-Server
P2P
Figure 2.25 Distribution time for P2P and client-server architectures
BitTorrent
BitTorrent is a popular P2P protocol for file distribution [Chao 2011]. In BitTor-
rent lingo, the collection of all peers participating in the distribution of a particular
file is called a torrent. Peers in a torrent download equal-size chunks of the file
from one another, with a typical chunk size of 256 KBytes. When a peer first joins
a torrent, it has no chunks. Over time it accumulates more and more chunks. While
it downloads chunks it also uploads chunks to other peers. Once a peer has
acquired the entire file, it may (selfishly) leave the torrent, or (altruistically) remain
in the torrent and continue to upload chunks to other peers. Also, any peer may leave
the torrent at any time with only a subset of chunks, and later rejoin the torrent.
Lets now take a closer look at how BitTorrent operates. Since BitTorrent is a
rather complicated protocol and system, well only describe its most important
mechanisms, sweeping some of the details under the rug; this will allow us to see
the forest through the trees. Each torrent has an infrastructure node called a tracker.
When a peer joins a torrent, it registers itself with the tracker and periodically
informs the tracker that it is still in the torrent. In this manner, the tracker keeps
track of the peers that are participating in the torrent. A given torrent may have
fewer than ten or more than a thousand peers participating at any instant of time.
As shown in Figure 2.26, when a new peer, Alice, joins the torrent, the tracker
randomly selects a subset of peers (for concreteness, say 50) from the set of participat-
ing peers, and sends the IP addresses of these 50 peers to Alice. Possessing this list of
peers, Alice attempts to establish concurrent TCP connections with all the peers on this
list. Lets call all the peers with which Alice succeeds in establishing a TCP connec-
tion neighboring peers. (In Figure 2.26, Alice is shown to have only three neighbor-
ing peers. Normally, she would have many more.) As time evolves, some of these
peers may leave and other peers (outside the initial 50) may attempt to establish TCP
connections with Alice. So a peers neighboring peers will fluctuate over time.
At any given time, each peer will have a subset of chunks from the file, with dif-
ferent peers having different subsets. Periodically, Alice will ask each of her neighbor-
ing peers (over the TCP connections) for the list of the chunks they have. If Alice has L
different neighbors, she will obtain L lists of chunks. With this knowledge, Alice will
issue requests (again over the TCP connections) for chunks she currently does not have.
So at any given instant of time, Alice will have a subset of chunks and will
know which chunks her neighbors have. With this information, Alice will have two
important decisions to make. First, which chunks should she request first from her
neighbors? And second, to which of her neighbors should she send requested
chunks? In deciding which chunks to request, Alice uses a technique called rarest
first. The idea is to determine, from among the chunks she does not have, the
chunks that are the rarest among her neighbors (that is, the chunks that have the
fewest repeated copies among her neighbors) and then request those rarest chunks
first. In this manner, the rarest chunks get more quickly redistributed, aiming to
(roughly) equalize the numbers of copies of each chunk in the torrent.
2.6 PEER-TO-PEER APPLICATIONS 149
To determine which requests she responds to, BitTorrent uses a clever trading
algorithm. The basic idea is that Alice gives priority to the neighbors that are cur-
rently supplying her data at the highest rate. Specifically, for each of her neighbors,
Alice continually measures the rate at which she receives bits and determines the four
peers that are feeding her bits at the highest rate. She then reciprocates by sending
chunks to these same four peers. Every 10 seconds, she recalculates the rates and pos-
sibly modifies the set of four peers. In BitTorrent lingo, these four peers are said to
be unchoked. Importantly, every 30 seconds, she also picks one additional neighbor
at random and sends it chunks. Lets call the randomly chosen peer Bob. In BitTor-
rent lingo, Bob is said to be optimistically unchoked. Because Alice is sending data
to Bob, she may become one of Bobs top four uploaders, in which case Bob would
start to send data to Alice. If the rate at which Bob sends data to Alice is high enough,
Bob could then, in turn, become one of Alices top four uploaders. In other words,
every 30 seconds, Alice will randomly choose a new trading partner and initiate trad-
ing with that partner. If the two peers are satisfied with the trading, they will put each
other in their top four lists and continue trading with each other until one of the peers
finds a better partner. The effect is that peers capable of uploading at compatible rates
tend to find each other. The random neighbor selection also allows new peers to get
150 CHAPTER 2 APPLICATION LAYER
Tracker
Trading chunks
Peer
Obtain
list of
peers
Alice
Figure 2.26 File distribution with BitTorrent
chunks, so that they can have something to trade. All other neighboring peers besides
these five peers (four top peers and one probing peer) are choked, that is, they do
not receive any chunks from Alice. BitTorrent has a number of interesting mecha-
nisms that are not discussed here, including pieces (mini-chunks), pipelining, random
first selection, endgame mode, and anti-snubbing [Cohen 2003].
The incentive mechanism for trading just described is often referred to as tit-for-tat
[Cohen 2003]. It has been shown that this incentive scheme can be circumvented
[Liogkas 2006; Locher 2006; Piatek 2007]. Nevertheless, the BitTorrent ecosystem is
wildly successful, with millions of simultaneous peers actively sharing files in hun-
dreds of thousands of torrents. If BitTorrent had been designed without tit-for-tat (or a
variant), but otherwise exactly the same, BitTorrent would likely not even exist now, as
the majority of the users would have been freeriders [Saroiu 2002].
Interesting variants of the BitTorrent protocol are proposed [Guo 2005; Piatek
2007]. Also, many of the P2P live streaming applications, such as PPLive and
ppstream, have been inspired by BitTorrent [Hei 2007].
2.6.2 Distributed Hash Tables (DHTs)
In this section, we will consider how to implement a simple database in a P2P net-
work. Lets begin by describing a centralized version of this simple database, which
will simply contain (key, value) pairs. For example, the keys could be social secu-
rity numbers and the values could be the corresponding human names; in this case,
an example key-value pair is (156-45-7081, Johnny Wu). Or the keys could be con-
tent names (e.g., names of movies, albums, and software), and the value could be
the IP address at which the content is stored; in this case, an example key-value pair
is (Led Zeppelin IV, 128.17.123.38). We query the database with a key. If there are
one or more key-value pairs in the database that match the query key, the database
returns the corresponding values. So, for example, if the database stores social secu-
rity numbers and their corresponding human names, we can query with a specific
social security number, and the database returns the name of the human who has that
social security number. Or, if the database stores content names and their correspon-
ding IP addresses, we can query with a specific content name, and the database
returns the IP addresses that store the specific content.
Building such a database is straightforward with a client-server architecture that
stores all the (key, value) pairs in one central server. So in this section, well instead
consider how to build a distributed, P2P version of this database that will store the
(key, value) pairs over millions of peers. In the P2P system, each peer will only hold a
small subset of the totality of the (key, value) pairs. Well allow any peer to query the
distributed database with a particular key. The distributed database will then locate the
peers that have the corresponding (key, value) pairs and return the key-value pairs to
the querying peer. Any peer will also be allowed to insert new key-value pairs into the
database. Such a distributed database is referred to as a distributed hash table
(DHT).
2.6 PEER-TO-PEER APPLICATIONS 151
Before describing how we can create a DHT, lets first describe a specific
example DHT service in the context of P2P file sharing. In this case, a key is the
content name and the value is the IP address of a peer that has a copy of the content.
So, if Bob and Charlie each have a copy of the latest Linux distribution, then the
DHT database will include the following two key-value pairs: (Linux, IPBob) and
(Linux, IPCharlie). More specifically, since the DHT database is distributed over the
peers, some peer, say Dave, will be responsible for the key Linux and will have
the corresponding key-value pairs. Now suppose Alice wants to obtain a copy of
Linux. Clearly, she first needs to know which peers have a copy of Linux before she
can begin to download it. To this end, she queries the DHT with Linux as the key.
The DHT then determines that the peer Dave is responsible for the key Linux. The
DHT then contacts peer Dave, obtains from Dave the key-value pairs (Linux, IPBob)
and (Linux, IPCharlie), and passes them on to Alice. Alice can then download the lat-
est Linux distribution from either IPBob or IPCharlie.
Now lets return to the general problem of designing a DHT for general key-
value pairs. One nave approach to building a DHT is to randomly scatter the (key,
value) pairs across all the peers and have each peer maintain a list of the IP
addresses of all participating peers. In this design, the querying peer sends its query
to all other peers, and the peers containing the (key, value) pairs that match the key
can respond with their matching pairs. Such an approach is completely unscalable,
of course, as it would require each peer to not only know about all other peers (pos-
sibly millions of such peers!) but even worse, have each query sent to all peers.
We now describe an elegant approach to designing a DHT. To this end, lets first
assign an identifier to each peer, where each identifier is an integer in the range [0,
2n 1] for some fixed n. Note that each such identifier can be expressed by an n-bit
representation. Lets also require each key to be an integer in the same range. The
astute reader may have observed that the example keys described a little earlier (social
security numbers and content names) are not integers. To create integers out of such
keys, we will use a hash function that maps each key (e.g., social security number) to
an integer in the range [0, 2n 1]. A hash function is a many-to-one function for which
two different inputs can have the same output (same integer), but the likelihood of the
having the same output is extremely small. (Readers who are unfamiliar with hash
functions may want to visit Chapter 7, in which hash functions are discussed in some
detail.) The hash function is assumed to be available to all peers in the system. Hence-
forth, when we refer to the key, we are referring to the hash of the original key. So,
for example, if the original key is Led Zeppelin IV, the key used in the DHT will be
the integer that equals the hash of Led Zeppelin IV. As you may have guessed, this
is why Hash is used in the term Distributed Hash Function.
Lets now consider the problem of storing the (key, value) pairs in the DHT. The
central issue here is defining a rule for assigning keys to peers. Given that each peer
has an integer identifier and that each key is also an integer in the same range, a natu-
ral approach is to assign each (key, value) pair to the peer whose identifier is the
closest to the key. To implement such a scheme, well need to define what is meant by
closest, for which many conventions are possible. For convenience, lets define the
152 CHAPTER 2 APPLICATION LAYER
VideoNote
Walking through
distributed hash tables
closest peer as the closest successor of the key. To gain some insight here, lets take a
look at a specific example. Suppose n 4 so that all the peer and key identifiers are in
the range [0, 15]. Further suppose that there are eight peers in the system with identi-
fiers 1, 3, 4, 5, 8, 10, 12, and 15. Finally, suppose we want to store the (key, value) pair
(11, Johnny Wu) in one of the eight peers. But in which peer? Using our closest con-
vention, since peer 12 is the closest successor for key 11, we therefore store the pair
(11, Johnny Wu) in the peer 12. [To complete our definition of closest, if the key is
exactly equal to one of the peer identifiers, we store the (key, value) pair in that match-
ing peer; and if the key is larger than all the peer identifiers, we use a modulo-2n con-
vention, storing the (key, value) pair in the peer with the smallest identifier.]
Now suppose a peer, Alice, wants to insert a (key, value) pair into the DHT.
Conceptually, this is straightforward: She first determines the peer whose identifier
is closest to the key; she then sends a message to that peer, instructing it to store the
(key, value) pair. But how does Alice determine the peer that is closest to the key? If
Alice were to keep track of all the peers in the system (peer IDs and corresponding
IP addresses), she could locally determine the closest peer. But such an approach
requires each peer to keep track of all other peers in the DHTwhich is completely
impractical for a large-scale system with millions of peers.
Circular DHT
To address this problem of scale, lets now consider organizing the peers into a
circle. In this circular arrangement, each peer only keeps track of its immediate suc-
cessor and immediate predecessor (modulo 2n). An example of such a circle is
shown in Figure 2.27(a). In this example, n is again 4 and there are the same eight
2.6 PEER-TO-PEER APPLICATIONS 153
1
3
Who is
responsible
for key 11?
4
5
8
a. b.
10
12
15
1
3
4
5
8
10
12
15
Figure 2.27 (a) A circular DHT. Peer 3 wants to determine who is
responsible for key 11. (b) A circular DHT with shortcuts
peers from the previous example. Each peer is only aware of its immediate succes-
sor and predecessor; for example, peer 5 knows the IP address and identifier for
peers 8 and 4 but does not necessarily know anything about any other peers that may
be in the DHT. This circular arrangement of the peers is a special case of an overlay
network. In an overlay network, the peers form an abstract logical network which
resides above the underlay computer network consisting of physical links, routers,
and hosts. The links in an overlay network are not physical links, but are simply vir-
tual liaisons between pairs of peers. In the overlay in Figure 2.27(a), there are eight
peers and eight overlay links; in the overlay in Figure 2.27(b) there are eight peers
and 16 overlay links. A single overlay link typically uses many physical links and
physical routers in the underlay network.
Using the circular overlay in Figure 2.27(a), now suppose that peer 3 wants to
determine which peer in the DHT is responsible for key 11. Using the circular overlay,
the origin peer (peer 3) creates a message saying Who is responsible for key 11? and
sends this message clockwise around the circle. Whenever a peer receives such a mes-
sage, because it knows the identifier of its successor and predecessor, it can determine
whether it is responsible for (that is, closest to) the key in question. If a peer is not
responsible for the key, it simply sends the message to its successor. So, for example,
when peer 4 receives the message asking about key 11, it determines that it is not
responsible for the key (because its successor is closer to the key), so it just passes the
message along to peer 5. This process continues until the message arrives at peer 12,
who determines that it is the closest peer to key 11. At this point, peer 12 can send a
message back to the querying peer, peer 3, indicating that it is responsible for key 11.
The circular DHT provides a very elegant solution for reducing the amount of
overlay information each peer must manage. In particular, each peer needs only to
be aware of two peers, its immediate successor and its immediate predecessor. But
this solution introduces yet a new problem. Although each peer is only aware of two
neighboring peers, to find the node responsible for a key (in the worst case), all N
nodes in the DHT will have to forward a message around the circle; N/2 messages
are sent on average.
Thus, in designing a DHT, there is tradeoff between the number of neighbors each
peer has to track and the number of messages that the DHT needs to send to resolve a
single query. On one hand, if each peer tracks all other peers (mesh overlay), then only
one message is sent per query, but each peer has to keep track of N peers. On the other
hand, with a circular DHT, each peer is only aware of two peers, but N/2 messages are
sent on average for each query. Fortunately, we can refine our designs of DHTs so that
the number of neighbors per peer as well as the number of messages per query is kept
to an acceptable size. One such refinement is to use the circular overlay as a founda-
tion, but add shortcuts so that each peer not only keeps track of its immediate suc-
cessor and predecessor, but also of a relatively small number of shortcut peers
scattered about the circle. An example of such a circular DHT with some shortcuts is
shown in Figure 2.27(b). Shortcuts are used to expedite the routing of query messages.
Specifically, when a peer receives a message that is querying for a key, it forwards the
154 CHAPTER 2 APPLICATION LAYER
message to the neighbor (successor neighbor or one of the shortcut neighbors) which
is the closet to the key. Thus, in Figure 2.27(b), when peer 4 receives the message ask-
ing about key 11, it determines that the closet peer to the key (among its neighbors) is
its shortcut neighbor 10 and then forwards the message directly to peer 10. Clearly,
shortcuts can significantly reduce the number of messages used to process a query.
The next natural question is How many shortcut neighbors should a peer have,
and which peers should be these shortcut neighbors? This question has received sig-
nificant attention in the research community [Balakrishnan 2003; Androutsellis-
Theotokis2004]. Importantly, it has been shown that the DHT can be designed so that
both the number of neighbors per peer as well as the number of messages per query is
O(log N), where N is the number of peers. Such designs strike a satisfactory compro-
mise between the extreme solutions of using mesh and circular overlay topologies.
Peer Churn
In P2P systems, a peer can come or go without warning. Thus, when designing a
DHT, we also must be concerned about maintaining the DHT overlay in the pres-
ence of such peer churn. To get a big-picture understanding of how this could be
accomplished, lets once again consider the circular DHT in Figure 2.27(a). To han-
dle peer churn, we will now require each peer to track (that is, know the IP address
of) its first and second successors; for example, peer 4 now tracks both peer 5 and
peer 8. We also require each peer to periodically verify that its two successors are
alive (for example, by periodically sending ping messages to them and asking for
responses). Lets now consider how the DHT is maintained when a peer abruptly
leaves. For example, suppose peer 5 in Figure 2.27(a) abruptly leaves. In this case,
the two peers preceding the departed peer (4 and 3) learn that 5 has departed, since
it no longer responds to ping messages. Peers 4 and 3 thus need to update their suc-
cessor state information. Lets consider how peer 4 updates its state:
1. Peer 4 replaces its first successor (peer 5) with its second successor (peer 8).
2. Peer 4 then asks its new first successor (peer 8) for the identifier and IP address of
its immediate successor (peer 10). Peer 4 then makes peer 10 its second successor.
In the homework problems, you will be asked to determine how peer 3 updates its
overlay routing information.
Having briefly addressed what has to be done when a peer leaves, lets now
consider what happens when a peer wants to join the DHT. Lets say a peer with
identifier 13 wants to join the DHT, and at the time of joining, it only knows about
peer 1s existence in the DHT. Peer 13 would first send peer 1 a message, saying
what will be 13s predecessor and successor? This message gets forwarded
through the DHT until it reaches peer 12, who realizes that it will be 13s predeces-
sor and that its current successor, peer 15, will become 13s successor. Next, peer 12
sends this predecessor and successor information to peer 13. Peer 13 can now join
2.6 PEER-TO-PEER APPLICATIONS 155
the DHT by making peer 15 its successor and by notifying peer 12 that it should
change its immediate successor to 13.
DHTs have been finding widespread use in practice. For example, BitTorrent
uses the Kademlia DHT to create a distributed tracker. In the BitTorrent, the key is
the torrent identifier and the value is the IP addresses of all the peers currently par-
ticipating in the torrent [Falkner 2007, Neglia 2007]. In this manner, by querying
the DHT with a torrent identifier, a newly arriving BitTorrent peer can determine the
peer that is responsible for the identifier (that is, for tracking the peers in the tor-
rent). After having found that peer, the arriving peer can query it for a list of other
peers in the torrent.
2.7 Socket Programming: Creating Network
Applications
Now that weve looked at a number of important network applications, lets explore
how network application programs are actually created. Recall from Section 2.1 that
a typical network application consists of a pair of programsa client program and a
server programresiding in two different end systems. When these two programs
are executed, a client process and a server process are created, and these processes
communicate with each other by reading from, and writing to, sockets. When creat-
ing a network application, the developers main task is therefore to write the code
for both the client and server programs.
There are two types of network applications. One type is an implementation
whose operation is specified in a protocol standard, such as an RFC or some
other standards document; such an application is sometimes referred to as
open, since the rules specifying its operation are known to all. For such an
implementation, the client and server programs must conform to the rules dic-
tated by the RFC. For example, the client program could be an implementation
of the client side of the FTP protocol, described in Section 2.3 and explicitly
defined in RFC 959; similarly, the server program could be an implementation of
the FTP server protocol, also explicitly defined in RFC 959. If one developer
writes code for the client program and another developer writes code for the
server program, and both developers carefully follow the rules of the RFC, then
the two programs will be able to interoperate. Indeed, many of todays network
applications involve communication between client and server programs that
have been created by independent developersfor example, a Firefox browser
communicating with an Apache Web server, or a BitTorrent client communicat-
ing with BitTorrent tracker.
The other type of network application is a proprietary network application.
In this case the client and server programs employ an application-layer protocol that
has not been openly published in an RFC or elsewhere. A single developer (or
156 CHAPTER 2 APPLICATION LAYER
development team) creates both the client and server programs, and the developer
has complete control over what goes in the code. But because the code does not
implement an open protocol, other independent developers will not be able to
develop code that interoperates with the application.
In this section, well examine the key issues in developing a client-server appli-
cation, and well get our hands dirty by looking at code that implements a very
simple client-server application. During the development phase, one of the first
decisions the developer must make is whether the application is to run over TCP
or over UDP. Recall that TCP is connection oriented and provides a reliable byte-
stream channel through which data flows between two end systems. UDP is
connectionless and sends independent packets of data from one end system to the
other, without any guarantees about delivery. Recall also that when a client or server
program implements a protocol defined by an RFC, it should use the well-known
port number associated with the protocol; conversely, when developing a propri-
etary application, the developer must be careful to avoid using such well-known
port numbers. (Port numbers were briefly discussed in Section 2.1. They are cov-
ered in more detail in Chapter 3.)
We introduce UDP and TCP socket programming by way of a simple UDP appli-
cation and a simple TCP application. We present the simple UDP and TCP applica-
tions in Python. We could have written the code in Java, C, or C++, but we chose
Python mostly because Python clearly exposes the key socket concepts. With Python
there are fewer lines of code, and each line can be explained to the novice program-
mer without difficulty. But theres no need to be frightened if you are not familiar with
Python. You should be able to easily follow the code if you have experience program-
ming in Java, C, or C++.
If you are interested in client-server programming with Java, you are encour-
aged to see the companion Web site for this textbook; in fact, you can find there all
the examples in this section (and associated labs) in Java. For readers who are inter-
ested in client-server programming in C, there are several good references available
[Donahoo 2001; Stevens 1997; Frost 1994; Kurose 1996]; our Python examples
below have a similar look and feel to C.
2.7.1 Socket Programming with UDP
In this subsection, well write simple client-server programs that use UDP; in the
following section, well write similar programs that use TCP.
Recall from Section 2.1 that processes running on different machines communi-
cate with each other by sending messages into sockets. We said that each process is
analogous to a house and the processs socket is analogous to a door. The application
resides on one side of the door in the house; the transport-layer protocol resides on
the other side of the door in the outside world. The application developer has control
of everything on the application-layer side of the socket; however, it has little control
of the transport-layer side.
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 157
Now lets take a closer look at the interaction between two communicating
processes that use UDP sockets. Before the sending process can push a packet of
data out the socket door, when using UDP, it must first attach a destination address
to the packet. After the packet passes through the senders socket, the Internet will
use this destination address to route the packet through the Internet to the socket in
the receiving process. When the packet arrives at the receiving socket, the receiving
process will retrieve the packet through the socket, and then inspect the packets
contents and take appropriate action.
So you may be now wondering, what goes into the destination address that is
attached to the packet? As you might expect, the destination hosts IP address is part of
the destination address. By including the destination IP address in the packet, the
routers in the Internet will be able to route the packet through the Internet to the desti-
nation host. But because a host may be running many network application processes,
each with one or more sockets, it is also necessary to identify the particular socket in
the destination host. When a socket is created, an identifier, called a port number, is
assigned to it. So, as you might expect, the packets destination address also includes
the sockets port number. In summary, the sending process attaches to the packet a des-
tination address which consists of the destination hosts IP address and the destination
sockets port number. Moreover, as we shall soon see, the senders source address
consisting of the IP address of the source host and the port number of the source
socketare also attached to the packet. However, attaching the source address to the
packet is typically not done by the UDP application code; instead it is automatically
done by the underlying operating system.
Well use the following simple client-server application to demonstrate socket
programming for both UDP and TCP:
1. The client reads a line of characters (data) from its keyboard and sends the data
to the server.
2. The server receives the data and converts the characters to uppercase.
3. The server sends the modified data to the client.
4. The client receives the modified data and displays the line on its screen.
Figure 2.28 highlights the main socket-related activity of the client and server that
communicate over the UDP transport service.
Now lets get our hands dirty and take a look at the client-server program pair
for a UDP implementation of this simple application. We also provide a detailed,
line-by-line analysis after each program. Well begin with the UDP client, which
will send a simple application-level message to the server. In order for the server to
be able to receive and reply to the clients message, it must be ready and running
that is, it must be running as a process before the client sends its message.
The client program is called UDPClient.py, and the server program is called
UDPServer.py. In order to emphasize the key issues, we intentionally provide code
that is minimal. Good code would certainly have a few more auxiliary lines, in
158 CHAPTER 2 APPLICATION LAYER
particular for handling error cases. For this application, we have arbitrarily chosen
12000 for the server port number.
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 159
Createsocket, port=x:
Server
serverSocket =
socket(AF_INET,SOCK_DGRAM)
(Running on serverIP)
Client
Read UDP segment from
serverSocket
Write reply to
specifying client address,
port number
serverSocket
Create datagram with serverIP
and port=x ;
send datagram via
clientSocket
Create socket:
clientSocket =
socket(AF_INET,SOCK_DGRAM)
Read datagram from
clientSocket
Close
clientSocket
Figure 2.28 The client-server application using UDP
UDPClient.py
Here is the code for the client side of the application:
from socket import *
serverName = hostname
serverPort = 12000
clientSocket = socket(socket.AF_INET, socket.SOCK_DGRAM)
message = raw_input(Input lowercase sentence:)
clientSocket.sendto(message,(serverName, serverPort))
modifiedMessage, serverAddress = clientSocket.recvfrom(2048)
print modifiedMessage
clientSocket.close()
Now lets take a look at the various lines of code in UDPClient.py.
from socket import *
The socket module forms the basis of all network communications in Python. By
including this line, we will be able to create sockets within our program.
serverName = hostname
serverPort = 12000
The first line sets the string serverName to hostname. Here, we provide a string
containing either the IP address of the server (e.g., 128.138.32.126) or the host-
name of the server (e.g., cis.poly.edu). If we use the hostname, then a DNS lookup
will automatically be performed to get the IP address.) The second line sets the inte-
ger variable serverPort to 12000.
clientSocket = socket(socket.AF_INET, socket.SOCK_DGRAM)
This line creates the clients socket, called clientSocket. The first parameter
indicates the address family; in particular, AF_INET indicates that the underlying
network is using IPv4. (Do not worry about this nowwe will discuss IPv4 in
Chapter 4.) The second parameter indicates that the socket is of type SOCK_DGRAM,
which means it is a UDP socket (rather than a TCP socket). Note that we are not
specifying the port number of the client socket when we create it; we are instead let-
ting the operating system do this for us. Now that the client processs door has been
created, we will want to create a message to send through the door.
160 CHAPTER 2 APPLICATION LAYER
message = raw_input(Input lowercase sentence:)
raw_input() is a built-in function in Python. When this command is executed,
the user at the client is prompted with the words Input data: The user then uses her
keyboard to input a line, which is put into the variable message. Now that we have
a socket and a message, we will want to send the message through the socket to the
destination host.
clientSocket.sendto(message,(serverName, serverPort))
In the above line, the method sendto() attaches the destination address
(serverName, serverPort) to the message and sends the resulting packet into
the processs socket, clientSocket. (As mentioned earlier, the source address is
also attached to the packet, although this is done automatically rather than explicitly
by the code.) Sending a client-to-server message via a UDP socket is that simple!
After sending the packet, the client waits to receive data from the server.
modifiedMessage, serverAddress = clientSocket.recvfrom(2048)
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 161
With the above line, when a packet arrives from the Internet at the clients socket,
the packets data is put into the variable modifiedMessage and the packets
source address is put into the variable serverAddress. The variable
serverAddress contains both the servers IP address and the servers port
number. The program UDPClient doesnt actually need this server address infor-
mation, since it already knows the server address from the outset; but this line of
Python provides the server address nevertheless. The method recvfrom also
takes the buffer size 2048 as input. (This buffer size works for most purposes.)
print modifiedMessage
This line prints out modifiedMessage on the users display. It should be the original
line that the user typed, but now capitalized.
clientSocket.close()
This line closes the socket. The process then terminates.
UDPServer.py
Lets now take a look at the server side of the application:
from socket import *
serverPort = 12000
serverSocket = socket(AF_INET, SOCK_DGRAM)
serverSocket.bind((, serverPort))
print The server is ready to receive
while 1:
message, clientAddress = serverSocket.recvfrom(2048)
modifiedMessage = message.upper()
serverSocket.sendto(modifiedMessage, clientAddress)
Note that the beginning of UDPServer is similar to UDPClient. It also imports the
socket module, also sets the integer variable serverPort to 12000, and also cre-
ates a socket of type SOCK_DGRAM (a UDP socket). The first line of code that is
significantly different from UDPClient is:
serverSocket.bind((, serverPort))
The above line binds (that is, assigns) the port number 12000 to the servers socket.
Thus in UDPServer, the code (written by the application developer) is explicitly
assigning a port number to the socket. In this manner, when anyone sends a packet
to port 12000 at the IP address of the server, that packet will be directed to this
socket. UDPServer then enters a while loop; the while loop will allow UDPServer
to receive and process packets from clients indefinitely. In the while loop,
UDPServer waits for a packet to arrive.
message, clientAddress = serverSocket.recvfrom(2048)
This line of code is similar to what we saw in UDPClient. When a packet arrives at
the servers socket, the packets data is put into the variable message and the
packets source address is put into the variable clientAddress. The variable
clientAddress contains both the clients IP address and the clients port number.
Here, UDPServer will make use of this address information, as it provides a return
address, similar to the return address with ordinary postal mail. With this source
address information, the server now knows to where it should direct its reply.
modifiedMessage = message.upper()
This line is the heart of our simple application. It takes the line sent by the client and
uses the method upper() to capitalize it.
serverSocket.sendto(modifiedMessage, clientAddress)
This last line attaches the clients address (IP address and port number) to the capi-
talized message, and sends the resulting packet into the servers socket. (As men-
tioned earlier, the server address is also attached to the packet, although this is done
automatically rather than explicitly by the code.) The Internet will then deliver the
packet to this client address. After the server sends the packet, it remains in the
while loop, waiting for another UDP packet to arrive (from any client running on
any host).
To test the pair of programs, you install and compile UDPClient.py in one host
and UDPServer.py in another host. Be sure to include the proper hostname or IP
address of the server in UDPClient.py. Next, you execute UDPServer.py, the com-
piled server program, in the server host. This creates a process in the server that
idles until it is contacted by some client. Then you execute UDPClient.py, the com-
piled client program, in the client. This creates a process in the client. Finally, to use
the application at the client, you type a sentence followed by a carriage return.
To develop your own UDP client-server application, you can begin by
slightly modifying the client or server programs. For example, instead of con-
verting all the letters to uppercase, the server could count the number of times the
letter s appears and return this number. Or you can modify the client so that after
receiving a capitalized sentence, the user can continue to send more sentences to
the server.
162 CHAPTER 2 APPLICATION LAYER
2.7.2 Socket Programming with TCP
Unlike UDP, TCP is a connection-oriented protocol. This means that before the client
and server can start to send data to each other, they first need to handshake and estab-
lish a TCP connection. One end of the TCP connection is attached to the client socket
and the other end is attached to a server socket. When creating the TCP connection,
we associate with it the client socket address (IP address and port number) and the
server socket address (IP address and port number). With the TCP connection estab-
lished, when one side wants to send data to the other side, it just drops the data into
the TCP connection via its socket. This is different from UDP, for which the server
must attach a destination address to the packet before dropping it into the socket.
Now lets take a closer look at the interaction of client and server programs in
TCP. The client has the job of initiating contact with the server. In order for the
server to be able to react to the clients initial contact, the server has to be ready.
This implies two things. First, as in the case of UDP, the TCP server must be run-
ning as a process before the client attempts to initiate contact. Second, the server
program must have a special doormore precisely, a special socketthat wel-
comes some initial contact from a client process running on an arbitrary host. Using
our house/door analogy for a process/socket, we will sometimes refer to the clients
initial contact as knocking on the welcoming door.
With the server process running, the client process can initiate a TCP connec-
tion to the server. This is done in the client program by creating a TCP socket. When
the client creates its TCP socket, it specifies the address of the welcoming socket in
the server, namely, the IP address of the server host and the port number of the
socket. After creating its socket, the client initiates a three-way handshake and
establishes a TCP connection with the server. The three-way handshake, which takes
place within the transport layer, is completely invisible to the client and server pro-
grams.
During the three-way handshake, the client process knocks on the welcoming door
of the server process. When the server hears the knocking, it creates a new door
more precisely, a new socket that is dedicated to that particular client. In our example
below, the welcoming door is a TCP socket object that we call serverSocket; the
newly created socket dedicated to the client making the connection is called connec-
tionSocket. Students who are encountering TCP sockets for the first time some-
times confuse the welcoming socket (which is the initial point of contact for all clients
wanting to communicate with the server), and each newly created server-side connec-
tion socket that is subsequently created for communicating with each client.
From the applications perspective, the clients socket and the servers connec-
tion socket are directly connected by a pipe. As shown in Figure 2.29, the client
process can send arbitrary bytes into its socket, and TCP guarantees that the server
process will receive (through the connection socket) each byte in the order sent. TCP
thus provides a reliable service between the client and server processes. Furthermore,
just as people can go in and out the same door, the client process not only sends bytes
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 163
into but also receives bytes from its socket; similarly, the server process not only
receives bytes from but also sends bytes into its connection socket.
We use the same simple client-server application to demonstrate socket program-
ming with TCP: The client sends one line of data to the server, the server capitalizes
the line and sends it back to the client. Figure 2.30 highlights the main socket-related
activity of the client and server that communicate over the TCP transport service.
TCPClient.py
Here is the code for the client side of the application:
164 CHAPTER 2 APPLICATION LAYER
Client process Server process
Client
socket
Welcoming
socket
Th
ree
-w
ay
ha
nd
sha
ke
Connection
socket
bytes
bytes
Figure 2.29 The TCPServer process has two sockets
from socket import *
serverName = servername
serverPort = 12000
clientSocket = socket(AF_INET, SOCK_STREAM)
clientSocket.connect((serverName,serverPort))
sentence = raw_input(Input lowercase sentence:)
clientSocket.send(sentence)
modifiedSentence = clientSocket.recv(1024)
print From Server:, modifiedSentence
clientSocket.close()
Lets now take a look at the various lines in the code that differ significantly
from the UDP implementation. The first such line is the creation of the client
socket.
clientSocket = socket(AF_INET, SOCK_STREAM)
This line creates the clients socket, called clientSocket. The first parameter
again indicates that the underlying network is using IPv4. The second parameter
indicates that the socket is of type SOCK_STREAM, which means it is a TCP socket
(rather than a UDP socket). Note that we are again not specifying the port number
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 165
Close
connectionSocket
Write reply to
connectionSocket
Read request from
connectionSocket
Createsocket, port=x,
for incoming request:
Server
serverSocket =
socket()
Wait for incoming
connection request:
connectionSocket =
serverSocket.accept()
(Running on serverIP)
Client
TCP
connection setup Create socket, connect
to serverIP, port=x:
clientSocket =
socket()
Read reply from
clientSocket
Send request using
clientSocket
Close
clientSocket
Figure 2.30 The client-server application using TCP
of the client socket when we create it; we are instead letting the operating system
do this for us. Now the next line of code is very different from what we saw in
UDPClient:
clientSocket.connect((serverName,serverPort))
Recall that before the client can send data to the server (or vice versa) using a TCP
socket, a TCP connection must first be established between the client and server.
The above line initiates the TCP connection between the client and server. The
parameter of the connect() method is the address of the server side of the con-
nection. After this line of code is executed, the three-way handshake is performed
and a TCP connection is established between the client and server.
166 CHAPTER 2 APPLICATION LAYER
sentence = raw_input(Input lowercase sentence:)
As with UDPClient, the above obtains a sentence from the user. The string sentence
continues to gather characters until the user ends the line by typing a carriage return.
The next line of code is also very different from UDPClient:
clientSocket.send(sentence)
The above line sends the string sentence through the clients socket and into the
TCP connection. Note that the program does not explicitly create a packet and attach
the destination address to the packet, as was the case with UDP sockets. Instead the
client program simply drops the bytes in the string sentence into the TCP con-
nection. The client then waits to receive bytes from the server.
modifiedSentence = clientSocket.recv(2048)
When characters arrive from the server, they get placed into the string modified-
Sentence. Characters continue to accumulate in modifiedSentence until the
line ends with a carriage return character. After printing the capitalized sentence, we
close the clients socket:
clientSocket.close()
This last line closes the socket and, hence, closes the TCP connection between the
client and the server. It causes TCP in the client to send a TCP message to TCP in
the server (see Section 3.5).
TCPServer.py
Now lets take a look at the server program.
from socket import *
serverPort = 12000
serverSocket = socket(AF_INET,SOCK_STREAM)
serverSocket.bind((,serverPort))
serverSocket.listen(1)
print The server is ready to receive
while 1:
connectionSocket, addr = serverSocket.accept()
sentence = connectionSocket.recv(1024)
capitalizedSentence = sentence.upper()
connectionSocket.send(capitalizedSentence)
connectionSocket.close()
Lets now take a look at the lines that differ significantly from UDPServer and TCP-
Client. As with TCPClient, the server creates a TCP socket with:
serverSocket=socket(AF_INET,SOCK_STREAM)
Similar to UDPServer, we associate the server port number, serverPort, with
this socket:
serverSocket.bind((,serverPort))
But with TCP, serverSocket will be our welcoming socket. After establish-
ing this welcoming door, we will wait and listen for some client to knock on the
door:
serverSocket.listen(1)
This line has the server listen for TCP connection requests from the client. The
parameter specifies the maximum number of queued connections (at least 1).
connectionSocket, addr = serverSocket.accept()
When a client knocks on this door, the program invokes the accept() method for
serverSocket, which creates a new socket in the server, called connec-
tionSocket, dedicated to this particular client. The client and server then complete
the handshaking, creating a TCP connection between the clients clientSocket
and the servers connectionSocket. With the TCP connection established, the
client and server can now send bytes to each other over the connection. With TCP, all
bytes sent from one side not are not only guaranteed to arrive at the other side but also
guaranteed arrive in order.
connectionSocket.close()
2.7 SOCKET PROGRAMMING: CREATING NETWORK APPLICATIONS 167
In this program, after sending the modified sentence to the client, we close the con-
nection socket. But since serverSocket remains open, another client can now
knock on the door and send the server a sentence to modify.
This completes our discussion of socket programming in TCP. You are encour-
aged to run the two programs in two separate hosts, and also to modify them to
achieve slightly different goals. You should compare the UDP program pair with the
TCP program pair and see how they differ. You should also do many of the socket
programming assignments described at the ends of Chapters 2, 4, and 7. Finally, we
hope someday, after mastering these and more advanced socket programs, you will
write your own popular network application, become very rich and famous, and
remember the authors of this textbook!
2.8 Summary
In this chapter, weve studied the conceptual and the implementation aspects of net-
work applications. Weve learned about the ubiquitous client-server architecture
adopted by many Internet applications and seen its use in the HTTP, FTP, SMTP,
POP3, and DNS protocols. Weve studied these important application-level proto-
cols, and their corresponding associated applications (the Web, file transfer, e-mail,
and DNS) in some detail. Weve also learned about the increasingly prevalent P2P
architecture and how it is used in many applications. Weve examined how the
socket API can be used to build network applications. Weve walked through the use
of sockets for connection-oriented (TCP) and connectionless (UDP) end-to-end
transport services. The first step in our journey down the layered network architec-
ture is now complete!
At the very beginning of this book, in Section 1.1, we gave a rather vague, bare-
bones definition of a protocol: the format and the order of messages exchanged
between two or more communicating entities, as well as the actions taken on the trans-
mission and/or receipt of a message or other event. The material in this chapter, and
in particular our detailed study of the HTTP, FTP, SMTP, POP3, and DNS protocols,
has now added considerable substance to this definition. Protocols are a key concept
in networking; our study of application protocols has now given us the opportunity to
develop a more intuitive feel for what protocols are all about.
In Section 2.1, we described the service models that TCP and UDP offer to
applications that invoke them. We took an even closer look at these service models
when we developed simple applications that run over TCP and UDP in Section 2.7.
However, we have said little about how TCP and UDP provide these service mod-
els. For example, we know that TCP provides a reliable data service, but we havent
said yet how it does so. In the next chapter well take a careful look at not only the
what, but also the how and why of transport protocols.
168 CHAPTER 2 APPLICATION LAYER
Equipped with knowledge about Internet application structure and application-
level protocols, were now ready to head further down the protocol stack and exam-
ine the transport layer in Chapter 3.
Homework Problems and Questions
Chapter 2 Review Questions
SECTION 2.1
R1. List five nonproprietary Internet applications and the application-layer
protocols that they use.
R2. What is the difference between network architecture and application
architecture?
R3. For a communication session between a pair of processes, which process is
the client and which is the server?
R4. For a P2P file-sharing application, do you agree with the statement, There is
no notion of client and server sides of a communication session? Why or
why not?
R5. What information is used by a process running on one host to identify a
process running on another host?
R6. Suppose you wanted to do a transaction from a remote client to a server as
fast as possible. Would you use UDP or TCP? Why?
R7. Referring to Figure 2.4, we see that none of the applications listed in Figure
2.4 requires both no data loss and timing. Can you conceive of an application
that requires no data loss and that is also highly time-sensitive?
R8. List the four broad classes of services that a transport protocol can provide.
For each of the service classes, indicate if either UDP or TCP (or both) pro-
vides such a service.
R9. Recall that TCP can be enhanced with SSL to provide process-to-process
security services, including encryption. Does SSL operate at the transport
layer or the application layer? If the application developer wants TCP to be
enhanced with SSL, what does the developer have to do?
SECTIONS 2.22.5
R10. What is meant by a handshaking protocol?
R11. Why do HTTP, FTP, SMTP, and POP3 run on top of TCP rather than on UDP?
R12. Consider an e-commerce site that wants to keep a purchase record for each of
its customers. Describe how this can be done with cookies.
HOMEWORK PROBLEMS AND QUESTIONS 169
R13. Describe how Web caching can reduce the delay in receiving a requested
object. Will Web caching reduce the delay for all objects requested by a user
or for only some of the objects? Why?
R14. Telnet into a Web server and send a multiline request message. Include in the
request message the If-modified-since: header line to force a
response message with the 304 Not Modified status code.
R15. Why is it said that FTP sends control information out-of-band?
R16. Suppose Alice, with a Web-based e-mail account (such as Hotmail or gmail),
sends a message to Bob, who accesses his mail from his mail server using
POP3. Discuss how the message gets from Alices host to Bobs host. Be sure
to list the series of application-layer protocols that are used to move the mes-
sage between the two hosts.
R17. Print out the header of an e-mail message you have recently received. How
many Received: header lines are there? Analyze each of the header lines
in the message.
R18. From a users perspective, what is the difference between the download-and-
delete mode and the download-and-keep mode in POP3?
R19. Is it possible for an organizations Web server and mail server to have exactly
the same alias for a hostname (for example, foo.com)? What would be the
type for the RR that contains the hostname of the mail server?
R20. Look over your received emails, and examine the header of a message sent
from a user with an .edu email address. Is it possible to determine from the
header the IP address of the host from which the message was sent? Do the
same for a message sent from a gmail account.
SECTION 2.6
R21. In BitTorrent, suppose Alice provides chunks to Bob throughout a 30-second
interval. Will Bob necessarily return the favor and provide chunks to Alice in
this same interval? Why or why not?
R22. Consider a new peer Alice that joins BitTorrent without possessing any
chunks. Without any chunks, she cannot become a top-four uploader for any
of the other peers, since she has nothing to upload. How then will Alice get
her first chunk?
R23. What is an overlay network? Does it include routers? What are the edges in
the overlay network?
R24. Consider a DHT with a mesh overlay topology (that is, every peer tracks all
peers in the system). What are the advantages and disadvantages of such a
design? What are the advantages and disadvantages of a circular DHT (with
no shortcuts)?
170 CHAPTER 2 APPLICATION LAYER
R25. List at least four different applications that are naturally suitable for P2P
architectures. (Hint: File distribution and instant messaging are two.)
SECTION 2.7
R26. In Section 2.7, the UDP server described needed only one socket, whereas the
TCP server needed two sockets. Why? If the TCP server were to support n
simultaneous connections, each from a different client host, how many sockets
would the TCP server need?
R27. For the client-server application over TCP described in Section 2.7, why must
the server program be executed before the client program? For the client-
server application over UDP, why may the client program be executed before
the server program?
Problems
P1. True or false?
a. A user requests a Web page that consists of some text and three images.
For this page, the client will send one request message and receive four
response messages.
b. Two distinct Web pages (for example, www.mit.edu/research.html
and www.mit.edu/students.html) can be sent over the same per-
sistent connection.
c. With nonpersistent connections between browser and origin server, it is pos-
sible for a single TCP segment to carry two distinct HTTP request messages.
d. The Date: header in the HTTP response message indicates when the
object in the response was last modified.
e. HTTP response messages never have an empty message body.
P2. Read RFC 959 for FTP. List all of the client commands that are supported by
the RFC.
P3. Consider an HTTP client that wants to retrieve a Web document at a given
URL. The IP address of the HTTP server is initially unknown. What transport
and application-layer protocols besides HTTP are needed in this scenario?
P4. Consider the following string of ASCII characters that were captured by
Wireshark when the browser sent an HTTP GET message (i.e., this is the actual
content of an HTTP GET message). The characters
return and line-feed characters (that is, the italized character string
the text below represents the single carriage-return character that was con-
tained at that point in the HTTP header). Answer the following questions,
indicating where in the HTTP GET message below you find the answer.
PROBLEMS 171
www.mit.edu/research.html
www.mit.edu/students.html
GET /cs453/index.html HTTP/1.1
a.cs.umass.edu
Windows;U; Windows NT 5.1; en-US; rv:1.7.2) Gec
ko/20040804 Netscape/7.2 (ax)
t/xml, application/xml, application/xhtml+xml, text
/html;q=0.9, text/plain;q=0.8,image/png,*/*;q=0.5
Encoding: zip,deflate
-8859-1,utf-8;q=0.7,*;q=0.7
a. What is the URL of the document requested by the browser?
b. What version of HTTP is the browser running?
c. Does the browser request a non-persistent or a persistent connection?
d. What is the IP address of the host on which the browser is running?
e. What type of browser initiates this message? Why is the browser type
needed in an HTTP request message?
P5. The text below shows the reply sent from the server in response to the HTTP
GET message in the question above. Answer the following questions, indicat-
ing where in the message below you find the answer.
172 CHAPTER 2 APPLICATION LAYER
HTTP/1.1 200 OK
12:39:45GMT
GMT
Ranges: bytes
Keep-Alive: timeout=max=100
Keep-Alive
ISO-8859-1
a. Was the server able to successfully find the document or not? What time
was the document reply provided?
b. When was the document last modified?
c. How many bytes are there in the document being returned?
d. What are the first 5 bytes of the document being returned? Did the server
agree to a persistent connection?
P6. Obtain the HTTP/1.1 specification (RFC 2616). Answer the following questions:
a. Explain the mechanism used for signaling between the client and server
to indicate that a persistent connection is being closed. Can the client, the
server, or both signal the close of a connection?
b. What encryption services are provided by HTTP?
c. Can a client open three or more simultaneous connections with a given
server?
d. Either a server or a client may close a transport connection between them
if either one detects the connection has been idle for some time. Is it pos-
sible that one side starts closing a connection while the other side is trans-
mitting data via this connection? Explain.
P7. Suppose within your Web browser you click on a link to obtain a Web page.
The IP address for the associated URL is not cached in your local host, so a
DNS lookup is necessary to obtain the IP address. Suppose that n DNS
servers are visited before your host receives the IP address from DNS; the
successive visits incur an RTT of RTT1, . . ., RTTn. Further suppose that the
Web page associated with the link contains exactly one object, consisting of a
small amount of HTML text. Let RTT0 denote the RTT between the local host
and the server containing the object. Assuming zero transmission time of the
object, how much time elapses from when the client clicks on the link until
the client receives the object?
P8. Referring to Problem P7, suppose the HTML file references eight very small
objects on the same server. Neglecting transmission times, how much time
elapses with
a. Non-persistent HTTP with no parallel TCP connections?
b. Non-persistent HTTP with the browser configured for 5 parallel connections?
c. Persistent HTTP?
P9. Consider Figure 2.12, for which there is an institutional network connected to
the Internet. Suppose that the average object size is 850,000 bits and that the
average request rate from the institutions browsers to the origin servers is 16
requests per second. Also suppose that the amount of time it takes from when
the router on the Internet side of the access link forwards an HTTP request
until it receives the response is three seconds on average (see Section 2.2.5).
Model the total average response time as the sum of the average access delay
(that is, the delay from Internet router to institution router) and the average
Internet delay. For the average access delay, use /(1 ), where is the
average time required to send an object over the access link and is the
arrival rate of objects to the access link.
a. Find the total average response time.
b. Now suppose a cache is installed in the institutional LAN. Suppose the
miss rate is 0.4. Find the total response time.
PROBLEMS 173
P10. Consider a short, 10-meter link, over which a sender can transmit at a rate of
150 bits/sec in both directions. Suppose that packets containing data are
100,000 bits long, and packets containing only control (e.g., ACK or hand-
shaking) are 200 bits long. Assume that N parallel connections each get 1/N
of the link bandwidth. Now consider the HTTP protocol, and suppose that
each downloaded object is 100 Kbits long, and that the initial downloaded
object contains 10 referenced objects from the same sender. Would parallel
downloads via parallel instances of non-persistent HTTP make sense in this
case? Now consider persistent HTTP. Do you expect significant gains over
the non-persistent case? Justify and explain your answer.
P11. Consider the scenario introduced in the previous problem. Now suppose that
the link is shared by Bob with four other users. Bob uses parallel instances of
non-persistent HTTP, and the other four users use non-persistent HTTP with-
out parallel downloads.
a. Do Bobs parallel connections help him get Web pages more quickly?
Why or why not?
b. If all five users open five parallel instances of non-persistent HTTP, then
would Bobs parallel connections still be beneficial? Why or why not?
P12. Write a simple TCP program for a server that accepts lines of input from a
client and prints the lines onto the servers standard output. (You can do this by
modifying the TCPServer.py program in the text.) Compile and execute your
program. On any other machine that contains a Web browser, set the proxy
server in the browser to the host that is running your server program; also con-
figure the port number appropriately. Your browser should now send its GET
request messages to your server, and your server should display the messages
on its standard output. Use this platform to determine whether your browser
generates conditional GET messages for objects that are locally cached.
P13. What is the difference between MAIL FROM: in SMTP and From: in the
mail message itself?
P14. How does SMTP mark the end of a message body? How about HTTP? Can
HTTP use the same method as SMTP to mark the end of a message body?
Explain.
P15. Read RFC 5321 for SMTP. What does MTA stand for? Consider the follow-
ing received spam email (modified from a real spam email). Assuming only
the originator of this spam email is malacious and all other hosts are honest,
identify the malacious host that has generated this spam email.
From Fri Nov 07 13:41:30 2008
Return-Path:
Received: from barmail.cs.umass.edu
(barmail.cs.umass.edu [128.119.240.3]) by cs.umass.edu
(8.13.1/8.12.6) for
13:27:10 -0500
174 CHAPTER 2 APPLICATION LAYER
Received: from asusus-4b96 (localhost [127.0.0.1]) by
barmail.cs.umass.edu (Spam Firewall) for
(EST)
Received: from asusus-4b96 ([58.88.21.177]) by
barmail.cs.umass.edu for
07 Nov 2008 13:27:07 -0500 (EST)
Received: from [58.88.21.177] by
inbnd55.exchangeddd.com; Sat, 8 Nov 2008 01:27:07 +0700
From: Jonny
To:
Subject: How to secure your savings
P16. Read the POP3 RFC, RFC 1939. What is the purpose of the UIDL POP3
command?
P17. Consider accessing your e-mail with POP3.
a. Suppose you have configured your POP mail client to operate in the
download-and-delete mode. Complete the following transaction:
C: list
S: 1 498
S: 2 912
S: .
C: retr 1
S: blah blah
S: .blah
S: .
?
?
b. Suppose you have configured your POP mail client to operate in the
download-and-keep mode. Complete the following transaction:
C: list
S: 1 498
S: 2 912
S: .
C: retr 1
S: blah blah
S: .blah
S: .
?
?
PROBLEMS 175
176 CHAPTER 2 APPLICATION LAYER
c. Suppose you have configured your POP mail client to operate in the
download-and-keep mode. Using your transcript in part (b), suppose you
retrieve messages 1 and 2, exit POP, and then five minutes later you again
access POP to retrieve new e-mail. Suppose that in the five-minute inter-
val no new messages have been sent to you. Provide a transcript of this
second POP session.
P18. a. What is a whois database?
b. Use various whois databases on the Internet to obtain the names of two
DNS servers. Indicate which whois databases you used.
c. Use nslookup on your local host to send DNS queries to three DNS servers:
your local DNS server and the two DNS servers you found in part (b). Try
querying for Type A, NS, and MX reports. Summarize your findings.
d. Use nslookup to find a Web server that has multiple IP addresses. Does
the Web server of your institution (school or company) have multiple IP
addresses?
e. Use the ARIN whois database to determine the IP address range used by
your university.
f. Describe how an attacker can use whois databases and the nslookup tool
to perform reconnaissance on an institution before launching an attack.
g. Discuss why whois databases should be publicly available.
P19. In this problem, we use the useful dig tool available on Unix and Linux hosts
to explore the hierarchy of DNS servers. Recall that in Figure 2.21, a DNS
server higher in the DNS hierarchy delegates a DNS query to a DNS server
lower in the hierarchy, by sending back to the DNS client the name of that
lower-level DNS server. First read the man page for dig, and then answer the
following questions.
a. Starting with a root DNS server (from one ofthe root servers [a-m].root-
servers.net), initiate a sequence of queries for the IP address for your
departments Web server by using dig. Show the list of the names of DNS
servers in the delegation chain in answering your query.
b. Repeat part a) for several popular Web sites, such as google.com,
yahoo.com, or amazon.com.
P20. Suppose you can access the caches in the local DNS servers of your department.
Can you propose a way to roughly determine the Web servers (outside your
department) that are most popular among the users in your department? Explain.
P21. Suppose that your department has a local DNS server for all computers in the
department. You are an ordinary user (i.e., not a network/system administra-
tor). Can you determine if an external Web site was likely accessed from a
computer in your department a couple of seconds ago? Explain.
PROBLEMS 177
P22. Consider distributing a file of F = 15 Gbits to N peers. The server has an upload
rate of us = 30 Mbps, and each peer has a download rate of di = 2 Mbps and an
upload rate of u. For N = 10, 100, and 1,000 and u = 300 Kbps, 700 Kbps, and
2 Mbps, prepare a chart giving the minimum distribution time for each of
the combinations of N and u for both client-server distribution and P2P
distribution.
P23. Consider distributing a file of F bits to N peers using a client-server architec-
ture. Assume a fluid model where the server can simultaneously transmit to
multiple peers, transmitting to each peer at different rates, as long as the com-
bined rate does not exceed us.
a. Suppose that us/N dmin. Specify a distribution scheme that has a distri-
bution time of NF/us.
b. Suppose that us/N dmin. Specify a distribution scheme that has a distri-
bution time of F/ dmin.
c. Conclude that the minimum distribution time is in general given by
max{NF/us, F/ dmin}.
P24. Consider distributing a file of F bits to N peers using a P2P architecture.
Assume a fluid model. For simplicity assume that dmin is very large, so that
peer download bandwidth is never a bottleneck.
a. Suppose that us (us + u1 + + uN)/N. Specify a distribution scheme
that has a distribution time of F/us.
b. Suppose that us (us + u1 + + uN)/N. Specify a distribution scheme
that has a distribution time of NF/(us + u1 + + uN).
c. Conclude that the minimum distribution time is in general given by
max{F/us, NF/(us + u1 + + uN)}.
P25. Consider an overlay network with N active peers, with each pair of peers hav-
ing an active TCP connection. Additionally, suppose that the TCP connections
pass through a total of M routers. How many nodes and edges are there in the
corresponding overlay network?
P26. Suppose Bob joins a BitTorrent torrent, but he does not want to upload any
data to any other peers (so called free-riding).
a. Bob claims that he can receive a complete copy of the file that is shared
by the swarm. Is Bobs claim possible? Why or why not?
b. Bob further claims that he can further make his free-riding more effi-
cient by using a collection of multiple computers (with distinct IP
addresses) in the computer lab in his department. How can he do that?
P27. In the circular DHT example in Section 2.6.2, suppose that peer 3 learns that
peer 5 has left. How does peer 3 update its successor state information?
Which peer is now its first successor? Its second successor?
VideoNote
Walking through
distributed hash tables
178 CHAPTER 2 APPLICATION LAYER
P28. In the circular DHT example in Section 2.6.2, suppose that a new peer 6
wants to join the DHT and peer 6 initially only knows peer 15s IP address.
What steps are taken?
P29. Because an integer in [0, 2n 1] can be expressed as an n-bit binary number in
a DHT, each key can be expressed as k = (k0, k1, . . . , kn1), and each peer iden-
tifier can be expressed p = (p0, p1, . . . , pn1). Lets now define the XOR dis-
tance between a key k and peer p as
Describe how this metric can be used to assign (key, value) pairs to peers.
(To learn about how to build an efficient DHT using this natural metric, see
[Maymounkov 2002] in which the Kademlia DHT is described.)
P30. As DHTs are overlay networks, they may not necessarily match the under-
lay physical network well in the sense that two neighboring peers might be
physically very far away; for example, one peer could be in Asia and its
neighbor could be in North America. If we randomly and uniformly assign
identifiers to newly joined peers, would this assignment scheme cause such
a mismatch? Explain. And how would such a mismatch affect the DHTs
performance?
P31. Install and compile the Python programs TCPClient and UDPClient on one
host and TCPServer and UDPServer on another host.
a. Suppose you run TCPClient before you run TCPServer. What happens?
Why?
b. Suppose you run UDPClient before you run UDPServer. What happens?
Why?
c. What happens if you use different port numbers for the client and server
sides?
P32. Suppose that in UDPClient.py, after we create the socket, we add the line:
clientSocket.bind((, 5432))
Will it become necessary to change UDPServer.py? What are the port num-
bers for the sockets in UDPClient and UDPServer? What were they before
making this change?
P33. Can you configure your browser to open multiple simultaneous connections
to a Web site? What are the advantages and disadvantages of having a large
number of simultaneous TCP connections?
P34 We have seen that Internet TCP sockets treat the data being sent as a byte
stream but UDP sockets recognize message boundaries. What are one
d1k, p2 = a
n-1
j=0
k j pj 2j
advantage and one disadvantage of byte-oriented API versus having the API
explicitly recognize and preserve application-defined message boundaries?
P35. What is the Apache Web server? How much does it cost? What functionality
does it currently have? You may want to look at Wikipedia to answer this
question.
P36. Many BitTorrent clients use DHTs to create a distributed tracker. For these
DHTs, what is the key and what is the value?
Socket Programming Assignments
The companion Web site includes six socket programming assignments. The first
four assignments are summarized below. The fifth assignment makes use of the
ICMP protocol and is summarized at the end of Chapter 4. The sixth assignment
employs multimedia protocols and is summarized at the end of Chapter 7. It is
highly recommended that students complete several, if not all, of these assignments.
Students can find full details of these assignments, as well as important snippets of
the Python code, at the Web site http://www.awl.com/kurose-ross.
Assignment 1: Web Server
In this assignment, you will develop a simple Web server in Python that is capable
of processing only one request. Specifically, your Web server will (i) create a con-
nection socket when contacted by a client (browser); (ii) receive the HTTP request
from this connection; (iii) parse the request to determine the specific file being
requested; (iv) get the requested file from the servers file system; (v) create an
HTTP response message consisting of the requested file preceded by header lines;
and (vi) send the response over the TCP connection to the requesting browser. If a
browser requests a file that is not present in your server, your server should return a
404 Not Found error message.
In the companion Web site, we provide the skeleton code for your server. Your
job is to complete the code, run your server, and then test your server by sending
requests from browsers running on different hosts. If you run your server on a host
that already has a Web server running on it, then you should use a different port than
port 80 for your Web server.
Assignment 2: UDP Pinger
In this programming assignment, you will write a client ping program in Python.
Your client will send a simple ping message to a server, receive a corresponding
pong message back from the server, and determine the delay between when the
client sent the ping message and received the pong message. This delay is called the
Round Trip Time (RTT). The functionality provided by the client and server is
SOCKET PROGRAMMING ASSIGNMENTS 179
http://www.awl.com/kurose-ross
similar to the functionality provided by standard ping program available in modern
operating systems. However, standard ping programs use the Internet Control Mes-
sage Protocol (ICMP) (which we will study in Chapter 4). Here we will create a
nonstandard (but simple!) UDP-based ping program.
Your ping program is to send 10 ping messages to the target server over UDP.
For each message, your client is to determine and print the RTT when the correspon-
ding pong message is returned. Because UDP is an unreliable protocol, a packet sent
by the client or server may be lost. For this reason, the client cannot wait indefinitely
for a reply to a ping message. You should have the client wait up to one second for a
reply from the server; if no reply is received, the client should assume that the
packet was lost and print a message accordingly.
In this assignment, you will be given the complete code for the server (avail-
able in the companion Web site). Your job is to write the client code, which will be
very similar to the server code. It is recommended that you first study carefully the
server code. You can then write your client code, liberally cutting and pasting lines
from the server code.
Assignment 3: Mail Client
The goal of this programming assignment is to create a simple mail client that sends
email to any recipient. Your client will need to establish a TCP connection with a
mail server (e.g., a Google mail server), dialogue with the mail server using the
SMTP protocol, send an email message to a recipient (e.g., your friend) via the mail
server, and finally close the TCP connection with the mail server.
For this assignment, the companion Web site provides the skeleton code for
your client. Your job is to complete the code and test your client by sending
email to different user accounts. You may also try sending through different
servers (for example, through a Google mail server and through your university
mail server).
Assignment 4: Multi-Threaded Web Proxy
In this assignment, you will develop a Web proxy. When your proxy receives an
HTTP request for an object from a browser, it generates a new HTTP request for
the same object and sends it to the origin server. When the proxy receives the
corresponding HTTP response with the object from the origin server, it creates a
new HTTP response, including the object, and sends it to the client. This proxy
will be multi-threaded, so that it will be able to handle multiple requests at the
same time.
For this assignment, the companion Web site provides the skeleton code for the
proxy server. Your job is to complete the code, and then test it by having different
browsers request Web objects via your proxy.
180 CHAPTER 2 APPLICATION LAYER
Wireshark Lab: HTTP
Having gotten our feet wet with the Wireshark packet sniffer in Lab 1, were now
ready to use Wireshark to investigate protocols in operation. In this lab, well explore
several aspects of the HTTP protocol: the basic GET/reply interaction, HTTP message
formats, retrieving large HTML files, retrieving HTML files with embedded URLs,
persistent and non-persistent connections, and HTTP authentication and security.
As is the case with all Wireshark labs, the full description of this lab is available
at this books Web site, http://www.awl.com/kurose-ross.
Wireshark Lab: DNS
In this lab, we take a closer look at the client side of the DNS, the protocol that trans-
lates Internet hostnames to IP addresses. Recall from Section 2.5 that the clients role in
the DNS is relatively simplea client sends a query to its local DNS server and
receives a response back. Much can go on under the covers, invisible to the DNS
clients, as the hierarchical DNS servers communicate with each other to either recur-
sively or iteratively resolve the clients DNS query. From the DNS clients standpoint,
however, the protocol is quite simplea query is formulated to the local DNS server
and a response is received from that server. We observe DNS in action in this lab.
As is the case with all Wireshark labs, the full description of this lab is available at
this books Web site, http://www.awl.com/kurose-ross.
WIRESHARK LABS 181
VideoNote
Using Wireshark to
investigate the
HTTP protocol
http://www.awl.com/kurose-ross
http://www.awl.com/kurose-ross
182
How did you become interested in computing? Did you always know that you wanted to
work in information technology?
The video game and personal computing revolutions hit right when I was growing up
personal computing was the new technology frontier in the late 70s and early 80s. And it
wasnt just Apple and the IBM PC, but hundreds of new companies like Commodore and
Atari as well. I taught myself to program out of a book called Instant Freeze-Dried BASIC
at age 10, and got my first computer (a TRS-80 Color Computerlook it up!) at age 12.
Please describe one or two of the most exciting projects you have worked on during your
career. What were the biggest challenges?
Undoubtedly the most exciting project was the original Mosaic web browser in 9293
and the biggest challenge was getting anyone to take it seriously back then. At the time,
everyone thought the interactive future would be delivered as interactive television by
huge companies, not as the Internet by startups.
What excites you about the future of networking and the Internet? What are your biggest
concerns?
The most exciting thing is the huge unexplored frontier of applications and services that
programmers and entrepreneurs are able to explorethe Internet has unleashed creativity at
Marc Andreessen
Marc Andreessen is the co-creator of Mosaic, the Web browser that
popularized the World Wide Web in 1993. Mosaic had a clean,
easily understood interface and was the first browser to display
images in-line with text. In 1994, Marc Andreessen and Jim Clark
founded Netscape, whose browser was by far the most popular
browser through the mid-1990s. Netscape also developed the Secure
Sockets Layer (SSL) protocol and many Internet server products, includ-
ing mail servers and SSL-based Web servers. He is now a co-founder
and general partner of venture capital firm Andreessen Horowitz, over-
seeing portfolio development with holdings that include Facebook,
Foursquare, Groupon, Jawbone, Twitter, and Zynga. He serves on
numerous boards, including Bump, eBay, Glam Media, Facebook,
and Hewlett-Packard. He holds a BS in Computer Science from the
University of Illinois at Urbana-Champaign.
AN INTERVIEW WITH
183
a level that I dont think weve ever seen before. My biggest concern is the principle of
unintended consequenceswe dont always know the implications of what we do, such as
the Internet being used by governments to run a new level of surveillance on citizens.
Is there anything in particular students should be aware of as Web technology advances?
The rate of changethe most important thing to learn is how to learnhow to flexibly
adapt to changes in the specific technologies, and how to keep an open mind on the new
opportunities and possibilities as you move through your career.
What people inspired you professionally?
Vannevar Bush, Ted Nelson, Doug Engelbart, Nolan Bushnell, Bill Hewlett and Dave
Packard, Ken Olsen, Steve Jobs, Steve Wozniak, Andy Grove, Grace Hopper, Hedy Lamarr,
Alan Turing, Richard Stallman.
What are your recommendations for students who want to pursue careers in computing
and information technology?
Go as deep as you possibly can on understanding how technology is created, and then com-
plement with learning how business works.
Can technology solve the worlds problems?
No, but we advance the standard of living of people through economic growth, and most
economic growth throughout history has come from technologyso thats as good as it gets.
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CHAPTER 3
Transport
Layer
Residing between the application and network layers, the transport layer is a central
piece of the layered network architecture. It has the critical role of providing com-
munication services directly to the application processes running on different hosts.
The pedagogic approach we take in this chapter is to alternate between discussions
of transport-layer principles and discussions of how these principles are imple-
mented in existing protocols; as usual, particular emphasis will be given to Internet
protocols, in particular the TCP and UDP transport-layer protocols.
Well begin by discussing the relationship between the transport and network
layers. This sets the stage for examining the first critical function of the transport
layerextending the network layers delivery service between two end systems to a
delivery service between two application-layer processes running on the end sys-
tems. Well illustrate this function in our coverage of the Internets connectionless
transport protocol, UDP.
Well then return to principles and confront one of the most fundamental prob-
lems in computer networkinghow two entities can communicate reliably over a
medium that may lose and corrupt data. Through a series of increasingly compli-
cated (and realistic!) scenarios, well build up an array of techniques that transport
protocols use to solve this problem. Well then show how these principles are
embodied in TCP, the Internets connection-oriented transport protocol.
Well next move on to a second fundamentally important problem in networking
controlling the transmission rate of transport-layer entities in order to avoid, or 185
recover from, congestion within the network. Well consider the causes and conse-
quences of congestion, as well as commonly used congestion-control techniques.
After obtaining a solid understanding of the issues behind congestion control, well
study TCPs approach to congestion control.
3.1 Introduction and Transport-Layer Services
In the previous two chapters we touched on the role of the transport layer and the
services that it provides. Lets quickly review what we have already learned about
the transport layer.
A transport-layer protocol provides for logical communication between appli-
cation processes running on different hosts. By logical communication, we mean
that from an applications perspective, it is as if the hosts running the processes were
directly connected; in reality, the hosts may be on opposite sides of the planet, con-
nected via numerous routers and a wide range of link types. Application processes
use the logical communication provided by the transport layer to send messages to
each other, free from the worry of the details of the physical infrastructure used to
carry these messages. Figure 3.1 illustrates the notion of logical communication.
As shown in Figure 3.1, transport-layer protocols are implemented in the end
systems but not in network routers. On the sending side, the transport layer converts
the application-layer messages it receives from a sending application process into
transport-layer packets, known as transport-layer segments in Internet terminology.
This is done by (possibly) breaking the application messages into smaller chunks and
adding a transport-layer header to each chunk to create the transport-layer segment.
The transport layer then passes the segment to the network layer at the sending end
system, where the segment is encapsulated within a network-layer packet (a data-
gram) and sent to the destination. Its important to note that network routers act only
on the network-layer fields of the datagram; that is, they do not examine the fields of
the transport-layer segment encapsulated with the datagram. On the receiving side,
the network layer extracts the transport-layer segment from the datagram and passes
the segment up to the transport layer. The transport layer then processes the received
segment, making the data in the segment available to the receiving application.
More than one transport-layer protocol may be available to network applications.
For example, the Internet has two protocolsTCP and UDP. Each of these protocols
provides a different set of transport-layer services to the invoking application.
3.1.1 Relationship Between Transport and Network Layers
Recall that the transport layer lies just above the network layer in the protocol stack.
Whereas a transport-layer protocol provides logical communication between
processes running on different hosts, a network-layer protocol provides logical
186 CHAPTER 3 TRANSPORT LAYER
3.1 INTRODUCTION AND TRANSPORT-LAYER SERVICES 187
Mobile Network
National or
Global ISP
Local or
Regional ISP
Enterprise Network
Home NetworkNetwork
Data link
Physical
Application
Transport
Network
Data link
Physical
Application
Transport
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Logical end-to-end transport
Figure 3.1 The transport layer provides logical rather than physical
communication between application processes
communication between hosts. This distinction is subtle but important. Lets exam-
ine this distinction with the aid of a household analogy.
Consider two houses, one on the East Coast and the other on the West Coast, with
each house being home to a dozen kids. The kids in the East Coast household are
cousins of the kids in the West Coast household. The kids in the two households love
to write to each othereach kid writes each cousin every week, with each letter deliv-
ered by the traditional postal service in a separate envelope. Thus, each household
sends 144 letters to the other household every week. (These kids would save a lot of
money if they had e-mail!) In each of the households there is one kidAnn in the
West Coast house and Bill in the East Coast houseresponsible for mail collection
and mail distribution. Each week Ann visits all her brothers and sisters, collects the
mail, and gives the mail to a postal-service mail carrier, who makes daily visits to the
house. When letters arrive at the West Coast house, Ann also has the job of distribut-
ing the mail to her brothers and sisters. Bill has a similar job on the East Coast.
In this example, the postal service provides logical communication between the
two housesthe postal service moves mail from house to house, not from person to
person. On the other hand, Ann and Bill provide logical communication among the
cousinsAnn and Bill pick up mail from, and deliver mail to, their brothers and sis-
ters. Note that from the cousins perspective, Ann and Bill are the mail service, even
though Ann and Bill are only a part (the end-system part) of the end-to-end delivery
process. This household example serves as a nice analogy for explaining how the
transport layer relates to the network layer:
application messages = letters in envelopes
processes = cousins
hosts (also called end systems) = houses
transport-layer protocol = Ann and Bill
network-layer protocol = postal service (including mail carriers)
Continuing with this analogy, note that Ann and Bill do all their work within
their respective homes; they are not involved, for example, in sorting mail in any
intermediate mail center or in moving mail from one mail center to another. Simi-
larly, transport-layer protocols live in the end systems. Within an end system, a
transport protocol moves messages from application processes to the network edge
(that is, the network layer) and vice versa, but it doesnt have any say about how the
messages are moved within the network core. In fact, as illustrated in Figure 3.1,
intermediate routers neither act on, nor recognize, any information that the transport
layer may have added to the application messages.
Continuing with our family saga, suppose now that when Ann and Bill go on
vacation, another cousin pairsay, Susan and Harveysubstitute for them and pro-
vide the household-internal collection and delivery of mail. Unfortunately for the
two families, Susan and Harvey do not do the collection and delivery in exactly the
same way as Ann and Bill. Being younger kids, Susan and Harvey pick up and drop
off the mail less frequently and occasionally lose letters (which are sometimes
188 CHAPTER 3 TRANSPORT LAYER
chewed up by the family dog). Thus, the cousin-pair Susan and Harvey do not pro-
vide the same set of services (that is, the same service model) as Ann and Bill. In an
analogous manner, a computer network may make available multiple transport pro-
tocols, with each protocol offering a different service model to applications.
The possible services that Ann and Bill can provide are clearly constrained by
the possible services that the postal service provides. For example, if the postal serv-
ice doesnt provide a maximum bound on how long it can take to deliver mail
between the two houses (for example, three days), then there is no way that Ann and
Bill can guarantee a maximum delay for mail delivery between any of the cousin
pairs. In a similar manner, the services that a transport protocol can provide are often
constrained by the service model of the underlying network-layer protocol. If the
network-layer protocol cannot provide delay or bandwidth guarantees for transport-
layer segments sent between hosts, then the transport-layer protocol cannot provide
delay or bandwidth guarantees for application messages sent between processes.
Nevertheless, certain services can be offered by a transport protocol even when
the underlying network protocol doesnt offer the corresponding service at the net-
work layer. For example, as well see in this chapter, a transport protocol can offer
reliable data transfer service to an application even when the underlying network
protocol is unreliable, that is, even when the network protocol loses, garbles, or
duplicates packets. As another example (which well explore in Chapter 8 when we
discuss network security), a transport protocol can use encryption to guarantee that
application messages are not read by intruders, even when the network layer cannot
guarantee the confidentiality of transport-layer segments.
3.1.2 Overview of the Transport Layer in the Internet
Recall that the Internet, and more generally a TCP/IP network, makes two distinct
transport-layer protocols available to the application layer. One of these protocols is
UDP (User Datagram Protocol), which provides an unreliable, connectionless service
to the invoking application. The second of these protocols is TCP (Transmission Con-
trol Protocol), which provides a reliable, connection-oriented service to the invoking
application. When designing a network application, the application developer must
specify one of these two transport protocols. As we saw in Section 2.7, the application
developer selects between UDP and TCP when creating sockets.
To simplify terminology, when in an Internet context, we refer to the transport-
layer packet as a segment. We mention, however, that the Internet literature (for exam-
ple, the RFCs) also refers to the transport-layer packet for TCP as a segment but often
refers to the packet for UDP as a datagram. But this same Internet literature also uses
the term datagram for the network-layer packet! For an introductory book on computer
networking such as this, we believe that it is less confusing to refer to both TCP and
UDP packets as segments, and reserve the term datagram for the network-layer packet.
Before proceeding with our brief introduction of UDP and TCP, it will be use-
ful to say a few words about the Internets network layer. (Well learn about the net-
work layer in detail in Chapter 4.) The Internets network-layer protocol has a
3.1 INTRODUCTION AND TRANSPORT-LAYER SERVICES 189
nameIP, for Internet Protocol. IP provides logical communication between hosts.
The IP service model is a best-effort delivery service. This means that IP makes its
best effort to deliver segments between communicating hosts, but it makes no
guarantees. In particular, it does not guarantee segment delivery, it does not guaran-
tee orderly delivery of segments, and it does not guarantee the integrity of the data
in the segments. For these reasons, IP is said to be an unreliable service. We also
mention here that every host has at least one network-layer address, a so-called IP
address. Well examine IP addressing in detail in Chapter 4; for this chapter we need
only keep in mind that each host has an IP address.
Having taken a glimpse at the IP service model, lets now summarize the serv-
ice models provided by UDP and TCP. The most fundamental responsibility of UDP
and TCP is to extend IPs delivery service between two end systems to a delivery
service between two processes running on the end systems. Extending host-to-host
delivery to process-to-process delivery is called transport-layer multiplexing and
demultiplexing. Well discuss transport-layer multiplexing and demultiplexing in
the next section. UDP and TCP also provide integrity checking by including error-
detection fields in their segments headers. These two minimal transport-layer serv-
icesprocess-to-process data delivery and error checkingare the only two
services that UDP provides! In particular, like IP, UDP is an unreliable serviceit
does not guarantee that data sent by one process will arrive intact (or at all!) to the
destination process. UDP is discussed in detail in Section 3.3.
TCP, on the other hand, offers several additional services to applications. First
and foremost, it provides reliable data transfer. Using flow control, sequence num-
bers, acknowledgments, and timers (techniques well explore in detail in this chap-
ter), TCP ensures that data is delivered from sending process to receiving process,
correctly and in order. TCP thus converts IPs unreliable service between end sys-
tems into a reliable data transport service between processes. TCP also provides
congestion control. Congestion control is not so much a service provided to the
invoking application as it is a service for the Internet as a whole, a service for the
general good. Loosely speaking, TCP congestion control prevents any one TCP con-
nection from swamping the links and routers between communicating hosts with an
excessive amount of traffic. TCP strives to give each connection traversing a con-
gested link an equal share of the link bandwidth. This is done by regulating the rate
at which the sending sides of TCP connections can send traffic into the network.
UDP traffic, on the other hand, is unregulated. An application using UDP transport
can send at any rate it pleases, for as long as it pleases.
A protocol that provides reliable data transfer and congestion control is neces-
sarily complex. Well need several sections to cover the principles of reliable data
transfer and congestion control, and additional sections to cover the TCP protocol
itself. These topics are investigated in Sections 3.4 through 3.8. The approach taken
in this chapter is to alternate between basic principles and the TCP protocol. For
example, well first discuss reliable data transfer in a general setting and then dis-
cuss how TCP specifically provides reliable data transfer. Similarly, well first
190 CHAPTER 3 TRANSPORT LAYER
discuss congestion control in a general setting and then discuss how TCP performs
congestion control. But before getting into all this good stuff, lets first look at
transport-layer multiplexing and demultiplexing.
3.2 Multiplexing and Demultiplexing
In this section, we discuss transport-layer multiplexing and demultiplexing, that is,
extending the host-to-host delivery service provided by the network layer to a
process-to-process delivery service for applications running on the hosts. In order to
keep the discussion concrete, well discuss this basic transport-layer service in the
context of the Internet. We emphasize, however, that a multiplexing/demultiplexing
service is needed for all computer networks.
At the destination host, the transport layer receives segments from the network
layer just below. The transport layer has the responsibility of delivering the data in
these segments to the appropriate application process running in the host. Lets take
a look at an example. Suppose you are sitting in front of your computer, and you are
downloading Web pages while running one FTP session and two Telnet sessions.
You therefore have four network application processes runningtwo Telnet
processes, one FTP process, and one HTTP process. When the transport layer in
your computer receives data from the network layer below, it needs to direct the
received data to one of these four processes. Lets now examine how this is done.
First recall from Section 2.7 that a process (as part of a network application) can
have one or more sockets, doors through which data passes from the network to the
process and through which data passes from the process to the network. Thus, as
shown in Figure 3.2, the transport layer in the receiving host does not actually
deliver data directly to a process, but instead to an intermediary socket. Because at
any given time there can be more than one socket in the receiving host, each socket
has a unique identifier. The format of the identifier depends on whether the socket is
a UDP or a TCP socket, as well discuss shortly.
Now lets consider how a receiving host directs an incoming transport-layer seg-
ment to the appropriate socket. Each transport-layer segment has a set of fields in the
segment for this purpose. At the receiving end, the transport layer examines these
fields to identify the receiving socket and then directs the segment to that socket. This
job of delivering the data in a transport-layer segment to the correct socket is called
demultiplexing. The job of gathering data chunks at the source host from different
sockets, encapsulating each data chunk with header information (that will later be
used in demultiplexing) to create segments, and passing the segments to the network
layer is called multiplexing. Note that the transport layer in the middle host in Fig-
ure 3.2 must demultiplex segments arriving from the network layer below to either
process P1 or P2 above; this is done by directing the arriving segments data to the
corresponding processs socket. The transport layer in the middle host must also
3.2 MULTIPLEXING AND DEMULTIPLEXING 191
gather outgoing data from these sockets, form transport-layer segments, and pass
these segments down to the network layer. Although we have introduced multiplex-
ing and demultiplexing in the context of the Internet transport protocols, its impor-
tant to realize that they are concerns whenever a single protocol at one layer (at the
transport layer or elsewhere) is used by multiple protocols at the next higher layer.
To illustrate the demultiplexing job, recall the household analogy in the previ-
ous section. Each of the kids is identified by his or her name. When Bill receives a
batch of mail from the mail carrier, he performs a demultiplexing operation by
observing to whom the letters are addressed and then hand delivering the mail to his
brothers and sisters. Ann performs a multiplexing operation when she collects let-
ters from her brothers and sisters and gives the collected mail to the mail person.
Now that we understand the roles of transport-layer multiplexing and demulti-
plexing, let us examine how it is actually done in a host. From the discussion above,
we know that transport-layer multiplexing requires (1) that sockets have unique
identifiers, and (2) that each segment have special fields that indicate the socket to
which the segment is to be delivered. These special fields, illustrated in Figure 3.3,
are the source port number field and the destination port number field. (The
UDP and TCP segments have other fields as well, as discussed in the subsequent
sections of this chapter.) Each port number is a 16-bit number, ranging from 0 to
65535. The port numbers ranging from 0 to 1023 are called well-known port num-
bers and are restricted, which means that they are reserved for use by well-known
application protocols such as HTTP (which uses port number 80) and FTP (which
uses port number 21). The list of well-known port numbers is given in RFC 1700
and is updated at http://www.iana.org [RFC 3232]. When we develop a new
192 CHAPTER 3 TRANSPORT LAYER
Network
Key:
Process Socket
Data link
Physical
Transport
Application
Network
Application
Data link
Physical
Transport
Network
Data link
Physical
Transport
P3 P2P1 P4 Application
Figure 3.2 Transport-layer multiplexing and demultiplexing
http://www.iana.org
application (such as the simple application developed in Section 2.7), we must
assign the application a port number.
It should now be clear how the transport layer could implement the demultiplex-
ing service: Each socket in the host could be assigned a port number, and when a seg-
ment arrives at the host, the transport layer examines the destination port number in
the segment and directs the segment to the corresponding socket. The segments data
then passes through the socket into the attached process. As well see, this is basi-
cally how UDP does it. However, well also see that multiplexing/demultiplexing in
TCP is yet more subtle.
Connectionless Multiplexing and Demultiplexing
Recall from Section 2.7.1 that the Python program running in a host can create a
UDP socket with the line
clientSocket = socket(socket.AF_INET, socket.SOCK_DGRAM)
When a UDP socket is created in this manner, the transport layer automatically
assigns a port number to the socket. In particular, the transport layer assigns a port
number in the range 1024 to 65535 that is currently not being used by any other UDP
port in the host. Alternatively, we can add a line into our Python program after we
create the socket to associate a specific port number (say, 19157) to this UDP socket
via the socket bind() method:
clientSocket.bind((, 19157))
If the application developer writing the code were implementing the server side of a
well-known protocol, then the developer would have to assign the corresponding
3.2 MULTIPLEXING AND DEMULTIPLEXING 193
Source port #
32 bits
Dest. port #
Other header fields
Application
data
(message)
Figure 3.3 Source and destination port-number fields in a transport-layer
segment
well-known port number. Typically, the client side of the application lets the trans-
port layer automatically (and transparently) assign the port number, whereas the
server side of the application assigns a specific port number.
With port numbers assigned to UDP sockets, we can now precisely describe
UDP multiplexing/demultiplexing. Suppose a process in Host A, with UDP port
19157, wants to send a chunk of application data to a process with UDP port 46428
in Host B. The transport layer in Host A creates a transport-layer segment that
includes the application data, the source port number (19157), the destination port
number (46428), and two other values (which will be discussed later, but are unim-
portant for the current discussion). The transport layer then passes the resulting seg-
ment to the network layer. The network layer encapsulates the segment in an IP
datagram and makes a best-effort attempt to deliver the segment to the receiving host.
If the segment arrives at the receiving Host B, the transport layer at the receiving
host examines the destination port number in the segment (46428) and delivers the
segment to its socket identified by port 46428. Note that Host B could be running
multiple processes, each with its own UDP socket and associated port number. As
UDP segments arrive from the network, Host B directs (demultiplexes) each segment
to the appropriate socket by examining the segments destination port number.
It is important to note that a UDP socket is fully identified by a two-tuple consist-
ing of a destination IP address and a destination port number. As a consequence, if two
UDP segments have different source IP addresses and/or source port numbers, but have
the same destination IP address and destination port number, then the two segments
will be directed to the same destination process via the same destination socket.
You may be wondering now, what is the purpose of the source port number? As
shown in Figure 3.4, in the A-to-B segment the source port number serves as part of
a return addresswhen B wants to send a segment back to A, the destination port
in the B-to-A segment will take its value from the source port value of the A-to-B
segment. (The complete return address is As IP address and the source port num-
ber.) As an example, recall the UDP server program studied in Section 2.7. In
UDPServer.py, the server uses the recvfrom() method to extract the client-
side (source) port number from the segment it receives from the client; it then sends
a new segment to the client, with the extracted source port number serving as the
destination port number in this new segment.
Connection-Oriented Multiplexing and Demultiplexing
In order to understand TCP demultiplexing, we have to take a close look at TCP
sockets and TCP connection establishment. One subtle difference between a TCP
socket and a UDP socket is that a TCP socket is identified by a four-tuple: (source
IP address, source port number, destination IP address, destination port number).
Thus, when a TCP segment arrives from the network to a host, the host uses all four
values to direct (demultiplex) the segment to the appropriate socket. In particular,
and in contrast with UDP, two arriving TCP segments with different source IP
194 CHAPTER 3 TRANSPORT LAYER
addresses or source port numbers will (with the exception of a TCP segment carry-
ing the original connection-establishment request) be directed to two different sock-
ets. To gain further insight, lets reconsider the TCP client-server programming
example in Section 2.7.2:
The TCP server application has a welcoming socket, that waits for connection-
establishment requests from TCP clients (see Figure 2.29) on port number 12000.
The TCP client creates a socket and sends a connection establishment request
segment with the lines:
clientSocket = socket(AF_INET, SOCK_STREAM)
clientSocket.connect((serverName,12000))
A connection-establishment request is nothing more than a TCP segment with desti-
nation port number 12000 and a special connection-establishment bit set in the TCP
header (discussed in Section 3.5). The segment also includes a source port number
that was chosen by the client.
When the host operating system of the computer running the server process
receives the incoming connection-request segment with destination port 12000,
it locates the server process that is waiting to accept a connection on port num-
ber 12000. The server process then creates a new socket:
connectionSocket, addr = serverSocket.accept()
3.2 MULTIPLEXING AND DEMULTIPLEXING 195
Host A
Client process
Socket
Server B
source port:
19157
dest. port:
46428
source port:
46428
dest. port:
19157
Figure 3.4 The inversion of source and destination port numbers
Also, the transport layer at the server notes the following four values in the con-
nection-request segment: (1) the source port number in the segment, (2) the IP
address of the source host, (3) the destination port number in the segment, and
(4) its own IP address. The newly created connection socket is identified by these
four values; all subsequently arriving segments whose source port, source IP
address, destination port, and destination IP address match these four values will
be demultiplexed to this socket. With the TCP connection now in place, the client
and server can now send data to each other.
The server host may support many simultaneous TCP connection sockets, with
each socket attached to a process, and with each socket identified by its own four-
tuple. When a TCP segment arrives at the host, all four fields (source IP address,
source port, destination IP address, destination port) are used to direct (demultiplex)
the segment to the appropriate socket.
196 CHAPTER 3 TRANSPORT LAYER
PORT SCANNING
Weve seen that a server process waits patiently on an open port for contact by a
remote client. Some ports are reserved for well-known applications (e.g., Web, FTP,
DNS, and SMTP servers); other ports are used by convention by popular applications
(e.g., the Microsoft 2000 SQL server listens for requests on UDP port 1434). Thus, if
we determine that a port is open on a host, we may be able to map that port to a
specific application running on the host. This is very useful for system administrators,
who are often interested in knowing which network applications are running on the
hosts in their networks. But attackers, in order to case the joint, also want to know
which ports are open on target hosts. If a host is found to be running an application
with a known security flaw (e.g., a SQL server listening on port 1434 was subject to
a buffer overflow, allowing a remote user to execute arbitrary code on the vulnerable
host, a flaw exploited by the Slammer worm [CERT 200304]), then that host is ripe
for attack.
Determining which applications are listening on which ports is a relatively easy
task. Indeed there are a number of public domain programs, called port scanners,
that do just that. Perhaps the most widely used of these is nmap, freely available at
http://nmap.org and included in most Linux distributions. For TCP, nmap sequentially
scans ports, looking for ports that are accepting TCP connections. For UDP, nmap
again sequentially scans ports, looking for UDP ports that respond to transmitted
UDP segments. In both cases, nmap returns a list of open, closed, or unreachable
ports. A host running nmap can attempt to scan any target host anywhere in the
Internet. Well revisit nmap in Section 3.5.6, when we discuss TCP connection
management.
FOCUS ON SECURITY
http://nmap.org
The situation is illustrated in Figure 3.5, in which Host C initiates two HTTP ses-
sions to server B, and Host A initiates one HTTP session to B. Hosts A and C and server
B each have their own unique IP addressA, C, and B, respectively. Host C assigns
two different source port numbers (26145 and 7532) to its two HTTP connections.
Because Host A is choosing source port numbers independently of C, it might also
assign a source port of 26145 to its HTTP connection. But this is not a problemserver
B will still be able to correctly demultiplex the two connections having the same source
port number, since the two connections have different source IP addresses.
Web Servers and TCP
Before closing this discussion, its instructive to say a few additional words about
Web servers and how they use port numbers. Consider a host running a Web server,
such as an Apache Web server, on port 80. When clients (for example, browsers)
send segments to the server, all segments will have destination port 80. In particu-
lar, both the initial connection-establishment segments and the segments carrying
HTTP request messages will have destination port 80. As we have just described,
3.2 MULTIPLEXING AND DEMULTIPLEXING 197
source port:
7532
dest. port:
80
source IP:
C
dest. IP:
B
source port:
26145
dest. port:
80
source IP:
C
dest. IP:
B
source port:
26145
dest. port:
80
source IP:
A
dest. IP:
B
Per-connection
HTTP
processes
Transport-
layer
demultiplexing
Web
server B
Web client
host C
Web client
host A
Figure 3.5 Two clients, using the same destination port number (80) to
communicate with the same Web server application
the server distinguishes the segments from the different clients using source IP
addresses and source port numbers.
Figure 3.5 shows a Web server that spawns a new process for each connection.
As shown in Figure 3.5, each of these processes has its own connection socket
through which HTTP requests arrive and HTTP responses are sent. We mention,
however, that there is not always a one-to-one correspondence between connection
sockets and processes. In fact, todays high-performing Web servers often use only
one process, and create a new thread with a new connection socket for each new
client connection. (A thread can be viewed as a lightweight subprocess.) If you did
the first programming assignment in Chapter 2, you built a Web server that does just
this. For such a server, at any given time there may be many connection sockets
(with different identifiers) attached to the same process.
If the client and server are using persistent HTTP, then throughout the duration
of the persistent connection the client and server exchange HTTP messages via the
same server socket. However, if the client and server use non-persistent HTTP, then
a new TCP connection is created and closed for every request/response, and hence
a new socket is created and later closed for every request/response. This frequent
creating and closing of sockets can severely impact the performance of a busy Web
server (although a number of operating system tricks can be used to mitigate
the problem). Readers interested in the operating system issues surrounding per-
sistent and non-persistent HTTP are encouraged to see [Nielsen 1997; Nahum
2002].
Now that weve discussed transport-layer multiplexing and demultiplexing,
lets move on and discuss one of the Internets transport protocols, UDP. In the next
section well see that UDP adds little more to the network-layer protocol than a mul-
tiplexing/demultiplexing service.
3.3 Connectionless Transport: UDP
In this section, well take a close look at UDP, how it works, and what it does.
We encourage you to refer back to Section 2.1, which includes an overview of
the UDP service model, and to Section 2.7.1, which discusses socket program-
ming using UDP.
To motivate our discussion about UDP, suppose you were interested in design-
ing a no-frills, bare-bones transport protocol. How might you go about doing this?
You might first consider using a vacuous transport protocol. In particular, on the
sending side, you might consider taking the messages from the application process
and passing them directly to the network layer; and on the receiving side, you might
consider taking the messages arriving from the network layer and passing them
directly to the application process. But as we learned in the previous section, we
have to do a little more than nothing! At the very least, the transport layer has to
198 CHAPTER 3 TRANSPORT LAYER
provide a multiplexing/demultiplexing service in order to pass data between the
network layer and the correct application-level process.
UDP, defined in [RFC 768], does just about as little as a transport protocol can
do. Aside from the multiplexing/demultiplexing function and some light error
checking, it adds nothing to IP. In fact, if the application developer chooses UDP
instead of TCP, then the application is almost directly talking with IP. UDP takes
messages from the application process, attaches source and destination port number
fields for the multiplexing/demultiplexing service, adds two other small fields, and
passes the resulting segment to the network layer. The network layer encapsulates
the transport-layer segment into an IP datagram and then makes a best-effort attempt
to deliver the segment to the receiving host. If the segment arrives at the receiving
host, UDP uses the destination port number to deliver the segments data to the cor-
rect application process. Note that with UDP there is no handshaking between send-
ing and receiving transport-layer entities before sending a segment. For this reason,
UDP is said to be connectionless.
DNS is an example of an application-layer protocol that typically uses UDP.
When the DNS application in a host wants to make a query, it constructs a DNS
query message and passes the message to UDP. Without performing any handshak-
ing with the UDP entity running on the destination end system, the host-side UDP
adds header fields to the message and passes the resulting segment to the network
layer. The network layer encapsulates the UDP segment into a datagram and sends
the datagram to a name server. The DNS application at the querying host then waits
for a reply to its query. If it doesnt receive a reply (possibly because the underlying
network lost the query or the reply), either it tries sending the query to another name
server, or it informs the invoking application that it cant get a reply.
Now you might be wondering why an application developer would ever choose
to build an application over UDP rather than over TCP. Isnt TCP always preferable,
since TCP provides a reliable data transfer service, while UDP does not? The answer
is no, as many applications are better suited for UDP for the following reasons:
Finer application-level control over what data is sent, and when. Under UDP, as
soon as an application process passes data to UDP, UDP will package the data
inside a UDP segment and immediately pass the segment to the network layer.
TCP, on the other hand, has a congestion-control mechanism that throttles the
transport-layer TCP sender when one or more links between the source and des-
tination hosts become excessively congested. TCP will also continue to resend a
segment until the receipt of the segment has been acknowledged by the destina-
tion, regardless of how long reliable delivery takes. Since real-time applications
often require a minimum sending rate, do not want to overly delay segment
transmission, and can tolerate some data loss, TCPs service model is not partic-
ularly well matched to these applications needs. As discussed below, these appli-
cations can use UDP and implement, as part of the application, any additional
functionality that is needed beyond UDPs no-frills segment-delivery service.
3.3 CONNECTIONLESS TRANSPORT: UDP 199
No connection establishment. As well discuss later, TCP uses a three-way hand-
shake before it starts to transfer data. UDP just blasts away without any formal pre-
liminaries. Thus UDP does not introduce any delay to establish a connection. This
is probably the principal reason why DNS runs over UDP rather than TCPDNS
would be much slower if it ran over TCP. HTTP uses TCP rather than UDP, since
reliability is critical for Web pages with text. But, as we briefly discussed in Sec-
tion 2.2, the TCP connection-establishment delay in HTTP is an important contrib-
utor to the delays associated with downloading Web documents.
No connection state. TCP maintains connection state in the end systems. This
connection state includes receive and send buffers, congestion-control parame-
ters, and sequence and acknowledgment number parameters. We will see in Sec-
tion 3.5 that this state information is needed to implement TCPs reliable data
transfer service and to provide congestion control. UDP, on the other hand, does
not maintain connection state and does not track any of these parameters. For this
reason, a server devoted to a particular application can typically support many
more active clients when the application runs over UDP rather than TCP.
Small packet header overhead. The TCP segment has 20 bytes of header over-
head in every segment, whereas UDP has only 8 bytes of overhead.
Figure 3.6 lists popular Internet applications and the transport protocols that they
use. As we expect, e-mail, remote terminal access, the Web, and file transfer run over
TCPall these applications need the reliable data transfer service of TCP. Neverthe-
less, many important applications run over UDP rather than TCP. UDP is used for RIP
routing table updates (see Section 4.6.1). Since RIP updates are sent periodically (typi-
cally every five minutes), lost updates will be replaced by more recent updates, thus
making the lost, out-of-date update useless. UDP is also used to carry network manage-
ment (SNMP; see Chapter 9) data. UDP is preferred to TCP in this case, since network
management applications must often run when the network is in a stressed statepre-
cisely when reliable, congestion-controlled data transfer is difficult to achieve. Also,
as we mentioned earlier, DNS runs over UDP, thereby avoiding TCPs connection-
establishment delays.
As shown in Figure 3.6, both UDP and TCP are used today with multimedia
applications, such as Internet phone, real-time video conferencing, and streaming of
stored audio and video. Well take a close look at these applications in Chapter 7. We
just mention now that all of these applications can tolerate a small amount of packet
loss, so that reliable data transfer is not absolutely critical for the applications suc-
cess. Furthermore, real-time applications, like Internet phone and video conferenc-
ing, react very poorly to TCPs congestion control. For these reasons, developers of
multimedia applications may choose to run their applications over UDP instead of
TCP. However, TCP is increasingly being used for streaming media transport. For
example, [Sripanidkulchai 2004] found that nearly 75% of on-demand and live
streaming used TCP. When packet loss rates are low, and with some organizations
200 CHAPTER 3 TRANSPORT LAYER
blocking UDP traffic for security reasons (see Chapter 8), TCP becomes an increas-
ingly attractive protocol for streaming media transport.
Although commonly done today, running multimedia applications over UDP is
controversial. As we mentioned above, UDP has no congestion control. But conges-
tion control is needed to prevent the network from entering a congested state in
which very little useful work is done. If everyone were to start streaming high-bit-
rate video without using any congestion control, there would be so much packet
overflow at routers that very few UDP packets would successfully traverse the
source-to-destination path. Moreover, the high loss rates induced by the uncon-
trolled UDP senders would cause the TCP senders (which, as well see, do decrease
their sending rates in the face of congestion) to dramatically decrease their rates.
Thus, the lack of congestion control in UDP can result in high loss rates between a
UDP sender and receiver, and the crowding out of TCP sessionsa potentially seri-
ous problem [Floyd 1999]. Many researchers have proposed new mechanisms to
force all sources, including UDP sources, to perform adaptive congestion control
[Mahdavi 1997; Floyd 2000; Kohler 2006: RFC 4340].
Before discussing the UDP segment structure, we mention that it is possible for
an application to have reliable data transfer when using UDP. This can be done if reli-
ability is built into the application itself (for example, by adding acknowledgment
and retransmission mechanisms, such as those well study in the next section). But
this is a nontrivial task that would keep an application developer busy debugging for
3.3 CONNECTIONLESS TRANSPORT: UDP 201
Application-Layer Underlying Transport
Application Protocol Protocol
Electronic mail SMTP TCP
Remote terminal access Telnet TCP
Web HTTP TCP
File transfer FTP TCP
Remote file server NFS Typically UDP
Streaming multimedia typically proprietary UDP or TCP
Internet telephony typically proprietary UDP or TCP
Network management SNMP Typically UDP
Routing protocol RIP Typically UDP
Name translation DNS Typically UDP
Figure 3.6 Popular Internet applications and their underlying transport
protocols
a long time. Nevertheless, building reliability directly into the application allows the
application to have its cake and eat it too. That is, application processes can com-
municate reliably without being subjected to the transmission-rate constraints
imposed by TCPs congestion-control mechanism.
3.3.1 UDP Segment Structure
The UDP segment structure, shown in Figure 3.7, is defined in RFC 768. The applica-
tion data occupies the data field of the UDP segment. For example, for DNS, the data
field contains either a query message or a response message. For a streaming audio
application, audio samples fill the data field. The UDP header has only four fields,
each consisting of two bytes. As discussed in the previous section, the port numbers
allow the destination host to pass the application data to the correct process running
on the destination end system (that is, to perform the demultiplexing function). The
length field specifies the number of bytes in the UDP segment (header plus data). An
explicit length value is needed since the size of the data field may differ from one UDP
segment to the next. The checksum is used by the receiving host to check whether
errors have been introduced into the segment. In truth, the checksum is also calculated
over a few of the fields in the IP header in addition to the UDP segment. But we ignore
this detail in order to see the forest through the trees. Well discuss the checksum cal-
culation below. Basic principles of error detection are described in Section 5.2. The
length field specifies the length of the UDP segment, including the header, in bytes.
3.3.2 UDP Checksum
The UDP checksum provides for error detection. That is, the checksum is used to
determine whether bits within the UDP segment have been altered (for example, by
noise in the links or while stored in a router) as it moved from source to destination.
UDP at the sender side performs the 1s complement of the sum of all the 16-bit
words in the segment, with any overflow encountered during the sum being
202 CHAPTER 3 TRANSPORT LAYER
Source port #
32 bits
Dest. port #
Length Checksum
Application
data
(message)
Figure 3.7 UDP segment structure
wrapped around. This result is put in the checksum field of the UDP segment. Here
we give a simple example of the checksum calculation. You can find details about
efficient implementation of the calculation in RFC 1071 and performance over real
data in [Stone 1998; Stone 2000]. As an example, suppose that we have the follow-
ing three 16-bit words:
0110011001100000
0101010101010101
1000111100001100
The sum of first two of these 16-bit words is
0110011001100000
0101010101010101
1011101110110101
Adding the third word to the above sum gives
1011101110110101
1000111100001100
0100101011000010
Note that this last addition had overflow, which was wrapped around. The 1s com-
plement is obtained by converting all the 0s to 1s and converting all the 1s to 0s.
Thus the 1s complement of the sum 0100101011000010 is 1011010100111101,
which becomes the checksum. At the receiver, all four 16-bit words are added,
including the checksum. If no errors are introduced into the packet, then clearly the
sum at the receiver will be 1111111111111111. If one of the bits is a 0, then we know
that errors have been introduced into the packet.
You may wonder why UDP provides a checksum in the first place, as many link-
layer protocols (including the popular Ethernet protocol) also provide error checking.
The reason is that there is no guarantee that all the links between source and destination
provide error checking; that is, one of the links may use a link-layer protocol that does
not provide error checking. Furthermore, even if segments are correctly transferred
across a link, its possible that bit errors could be introduced when a segment is stored
in a routers memory. Given that neither link-by-link reliability nor in-memory error
detection is guaranteed, UDP must provide error detection at the transport layer, on an
end-end basis, if the end-end data transfer service is to provide error detection. This is
an example of the celebrated end-end principle in system design [Saltzer 1984], which
states that since certain functionality (error detection, in this case) must be implemented
on an end-end basis: functions placed at the lower levels may be redundant or of little
value when compared to the cost of providing them at the higher level.
Because IP is supposed to run over just about any layer-2 protocol, it is useful
for the transport layer to provide error checking as a safety measure. Although UDP
3.3 CONNECTIONLESS TRANSPORT: UDP 203
provides error checking, it does not do anything to recover from an error. Some
implementations of UDP simply discard the damaged segment; others pass the dam-
aged segment to the application with a warning.
That wraps up our discussion of UDP. We will soon see that TCP offers reliable
data transfer to its applications as well as other services that UDP doesnt offer. Natu-
rally, TCP is also more complex than UDP. Before discussing TCP, however, it will be
useful to step back and first discuss the underlying principles of reliable data transfer.
3.4 Principles of Reliable Data Transfer
In this section, we consider the problem of reliable data transfer in a general con-
text. This is appropriate since the problem of implementing reliable data transfer
occurs not only at the transport layer, but also at the link layer and the application
layer as well. The general problem is thus of central importance to networking.
Indeed, if one had to identify a top-ten list of fundamentally important problems
in all of networking, this would be a candidate to lead the list. In the next section
well examine TCP and show, in particular, that TCP exploits many of the principles
that we are about to describe.
Figure 3.8 illustrates the framework for our study of reliable data transfer. The
service abstraction provided to the upper-layer entities is that of a reliable channel
through which data can be transferred. With a reliable channel, no transferred data
bits are corrupted (flipped from 0 to 1, or vice versa) or lost, and all are delivered in
the order in which they were sent. This is precisely the service model offered by
TCP to the Internet applications that invoke it.
It is the responsibility of a reliable data transfer protocol to implement this
service abstraction. This task is made difficult by the fact that the layer below the
reliable data transfer protocol may be unreliable. For example, TCP is a reliable data
transfer protocol that is implemented on top of an unreliable (IP) end-to-end net-
work layer. More generally, the layer beneath the two reliably communicating end
points might consist of a single physical link (as in the case of a link-level data
transfer protocol) or a global internetwork (as in the case of a transport-level proto-
col). For our purposes, however, we can view this lower layer simply as an unreli-
able point-to-point channel.
In this section, we will incrementally develop the sender and receiver sides of a
reliable data transfer protocol, considering increasingly complex models of the under-
lying channel. For example, well consider what protocol mechanisms are needed when
the underlying channel can corrupt bits or lose entire packets. One assumption well
adopt throughout our discussion here is that packets will be delivered in the order in
which they were sent, with some packets possibly being lost; that is, the underlying
channel will not reorder packets. Figure 3.8(b) illustrates the interfaces for our data
transfer protocol. The sending side of the data transfer protocol will be invoked from
above by a call to rdt_send(). It will pass the data to be delivered to the upper layer
at the receiving side. (Here rdt stands for reliable data transfer protocol and _send
204 CHAPTER 3 TRANSPORT LAYER
indicates that the sending side of rdt is being called. The first step in developing any
protocol is to choose a good name!) On the receiving side, rdt_rcv() will be called
when a packet arrives from the receiving side of the channel. When the rdt protocol
wants to deliver data to the upper layer, it will do so by calling deliver_data(). In
the following we use the terminology packet rather than transport-layer segment.
Because the theory developed in this section applies to computer networks in general
and not just to the Internet transport layer, the generic term packet is perhaps more
appropriate here.
In this section we consider only the case of unidirectional data transfer, that is,
data transfer from the sending to the receiving side. The case of reliable bidirectional
(that is, full-duplex) data transfer is conceptually no more difficult but considerably
more tedious to explain. Although we consider only unidirectional data transfer, it is
important to note that the sending and receiving sides of our protocol will nonetheless
need to transmit packets in both directions, as indicated in Figure 3.8. We will see
shortly that, in addition to exchanging packets containing the data to be transferred, the
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 205
Reliable channel
Unreliable channel
rdt_send()
udt_send()
Sending
process
Receiver
process
deliver_data
Application
layer
Transport
layer
a. Provided service
Network
layer
Key:
Data Packet
b. Service implementation
Reliable data
transfer protocol
(sending side)
Reliable data
transfer protocol
(receiving side)
rdt_rcv()
Figure 3.8 Reliable data transfer: Service model and service
implementation
sending and receiving sides of rdtwill also need to exchange control packets back and
forth. Both the send and receive sides of rdt send packets to the other side by a call to
udt_send() (where udt stands for unreliable data transfer).
3.4.1 Building a Reliable Data Transfer Protocol
We now step through a series of protocols, each one becoming more complex, arriv-
ing at a flawless, reliable data transfer protocol.
Reliable Data Transfer over a Perfectly Reliable Channel: rdt1.0
We first consider the simplest case, in which the underlying channel is completely
reliable. The protocol itself, which well call rdt1.0, is trivial. The finite-state
machine (FSM) definitions for the rdt1.0 sender and receiver are shown in
Figure 3.9. The FSM in Figure 3.9(a) defines the operation of the sender, while the
FSM in Figure 3.9(b) defines the operation of the receiver. It is important to note
that there are separate FSMs for the sender and for the receiver. The sender and
receiver FSMs in Figure 3.9 each have just one state. The arrows in the FSM
description indicate the transition of the protocol from one state to another. (Since
each FSM in Figure 3.9 has just one state, a transition is necessarily from the one
state back to itself; well see more complicated state diagrams shortly.) The event
causing the transition is shown above the horizontal line labeling the transition, and
206 CHAPTER 3 TRANSPORT LAYER
Wait for
call from
above
a.rdt1.0: sending side
rdt_send(data)
packet=make_pkt(data)
udt_send(packet)
Wait for
call from
below
b.rdt1.0: receiving side
rdt_rcv(packet)
extract(packet,data)
deliver_data(data)
Figure 3.9 rdt1.0 A protocol for a completely reliable channel
the actions taken when the event occurs are shown below the horizontal line. When
no action is taken on an event, or no event occurs and an action is taken, well use
the symbol below or above the horizontal, respectively, to explicitly denote the
lack of an action or event. The initial state of the FSM is indicated by the dashed
arrow. Although the FSMs in Figure 3.9 have but one state, the FSMs we will see
shortly have multiple states, so it will be important to identify the initial state of
each FSM.
The sending side of rdt simply accepts data from the upper layer via the
rdt_send(data) event, creates a packet containing the data (via the action
make_pkt(data)) and sends the packet into the channel. In practice, the
rdt_send(data) event would result from a procedure call (for example, to
rdt_send()) by the upper-layer application.
On the receiving side, rdt receives a packet from the underlying channel via
the rdt_rcv(packet) event, removes the data from the packet (via the action
extract (packet, data)) and passes the data up to the upper layer (via the
action deliver_data(data)). In practice, the rdt_rcv(packet) event
would result from a procedure call (for example, to rdt_rcv()) from the lower-
layer protocol.
In this simple protocol, there is no difference between a unit of data and a
packet. Also, all packet flow is from the sender to receiver; with a perfectly reliable
channel there is no need for the receiver side to provide any feedback to the sender
since nothing can go wrong! Note that we have also assumed that the receiver is able
to receive data as fast as the sender happens to send data. Thus, there is no need for
the receiver to ask the sender to slow down!
Reliable Data Transfer over a Channel with Bit Errors: rdt2.0
A more realistic model of the underlying channel is one in which bits in a packet
may be corrupted. Such bit errors typically occur in the physical components of a
network as a packet is transmitted, propagates, or is buffered. Well continue to
assume for the moment that all transmitted packets are received (although their bits
may be corrupted) in the order in which they were sent.
Before developing a protocol for reliably communicating over such a channel,
first consider how people might deal with such a situation. Consider how you your-
self might dictate a long message over the phone. In a typical scenario, the message
taker might say OK after each sentence has been heard, understood, and recorded.
If the message taker hears a garbled sentence, youre asked to repeat the garbled
sentence. This message-dictation protocol uses both positive acknowledgments
(OK) and negative acknowledgments (Please repeat that.). These control mes-
sages allow the receiver to let the sender know what has been received correctly, and
what has been received in error and thus requires repeating. In a computer network
setting, reliable data transfer protocols based on such retransmission are known as
ARQ (Automatic Repeat reQuest) protocols.
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 207
Fundamentally, three additional protocol capabilities are required in ARQ
protocols to handle the presence of bit errors:
Error detection. First, a mechanism is needed to allow the receiver to detect
when bit errors have occurred. Recall from the previous section that UDP uses
the Internet checksum field for exactly this purpose. In Chapter 5 well exam-
ine error-detection and -correction techniques in greater detail; these tech-
niques allow the receiver to detect and possibly correct packet bit errors. For
now, we need only know that these techniques require that extra bits (beyond
the bits of original data to be transferred) be sent from the sender to the
receiver; these bits will be gathered into the packet checksum field of the
rdt2.0 data packet.
Receiver feedback. Since the sender and receiver are typically executing on differ-
ent end systems, possibly separated by thousands of miles, the only way for the
sender to learn of the receivers view of the world (in this case, whether or not a
packet was received correctly) is for the receiver to provide explicit feedback to the
sender. The positive (ACK) and negative (NAK) acknowledgment replies in the
message-dictation scenario are examples of such feedback. Our rdt2.0 protocol
will similarly send ACK and NAK packets back from the receiver to the sender. In
principle, these packets need only be one bit long; for example, a 0 value could indi-
cate a NAK and a value of 1 could indicate an ACK.
Retransmission. A packet that is received in error at the receiver will be retrans-
mitted by the sender.
Figure 3.10 shows the FSM representation of rdt2.0, a data transfer protocol
employing error detection, positive acknowledgments, and negative acknowledgments.
The send side of rdt2.0 has two states. In the leftmost state, the send-side proto-
col is waiting for data to be passed down from the upper layer. When the
rdt_send(data) event occurs, the sender will create a packet (sndpkt)
containing the data to be sent, along with a packet checksum (for example, as discussed
in Section 3.3.2 for the case of a UDP segment), and then send the packet via the
udt_send(sndpkt) operation. In the rightmost state, the sender protocol is wait-
ing for an ACK or a NAK packet from the receiver. If an ACK packet is received (the
notation rdt_rcv(rcvpkt) && isACK (rcvpkt) in Figure 3.10 corresponds
to this event), the sender knows that the most recently transmitted packet has been
received correctly and thus the protocol returns to the state of waiting for data from the
upper layer. If a NAK is received, the protocol retransmits the last packet and waits for
an ACK or NAK to be returned by the receiver in response to the retransmitted data
packet. It is important to note that when the sender is in the wait-for-ACK-or-NAK
state, it cannot get more data from the upper layer; that is, the rdt_send() event can
not occur; that will happen only after the sender receives an ACK and leaves this state.
Thus, the sender will not send a new piece of data until it is sure that the receiver has
208 CHAPTER 3 TRANSPORT LAYER
correctly received the current packet. Because of this behavior, protocols such as
rdt2.0 are known as stop-and-wait protocols.
The receiver-side FSM for rdt2.0 still has a single state. On packet arrival,
the receiver replies with either an ACK or a NAK, depending on whether or not the
received packet is corrupted. In Figure 3.10, the notation rdt_rcv(rcvpkt) &&
corrupt(rcvpkt) corresponds to the event in which a packet is received and is
found to be in error.
Protocol rdt2.0 may look as if it works but, unfortunately, it has a fatal
flaw. In particular, we havent accounted for the possibility that the ACK or NAK
packet could be corrupted! (Before proceeding on, you should think about how this
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 209
Wait for
call from
above
a.rdt2.0: sending side
b.rdt2.0: receiving side
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt=make_pkt(NAK)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
rdt_send(data)
sndpkt=make_pkt(data,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt=make_pkt(ACK)
udt_send(sndpkt)
Wait for
call from
below
Wait for
ACK or
NAK
Figure 3.10 rdt2.0A protocol for a channel with bit errors
problem may be fixed.) Unfortunately, our slight oversight is not as innocuous as it
may seem. Minimally, we will need to add checksum bits to ACK/NAK packets in
order to detect such errors. The more difficult question is how the protocol should
recover from errors in ACK or NAK packets. The difficulty here is that if an ACK
or NAK is corrupted, the sender has no way of knowing whether or not the receiver
has correctly received the last piece of transmitted data.
Consider three possibilities for handling corrupted ACKs or NAKs:
For the first possibility, consider what a human might do in the message-
dictation scenario. If the speaker didnt understand the OK or Please repeat
that reply from the receiver, the speaker would probably ask, What did you
say? (thus introducing a new type of sender-to-receiver packet to our protocol).
The receiver would then repeat the reply. But what if the speakers What did
you say? is corrupted? The receiver, having no idea whether the garbled sen-
tence was part of the dictation or a request to repeat the last reply, would proba-
bly then respond with What did you say? And then, of course, that response
might be garbled. Clearly, were heading down a difficult path.
A second alternative is to add enough checksum bits to allow the sender not only
to detect, but also to recover from, bit errors. This solves the immediate problem
for a channel that can corrupt packets but not lose them.
A third approach is for the sender simply to resend the current data packet when
it receives a garbled ACK or NAK packet. This approach, however, introduces
duplicate packets into the sender-to-receiver channel. The fundamental diffi-
culty with duplicate packets is that the receiver doesnt know whether the ACK
or NAK it last sent was received correctly at the sender. Thus, it cannot know a
priori whether an arriving packet contains new data or is a retransmission!
A simple solution to this new problem (and one adopted in almost all existing
data transfer protocols, including TCP) is to add a new field to the data packet and
have the sender number its data packets by putting a sequence number into this
field. The receiver then need only check this sequence number to determine whether
or not the received packet is a retransmission. For this simple case of a stop-and-
wait protocol, a 1-bit sequence number will suffice, since it will allow the receiver
to know whether the sender is resending the previously transmitted packet (the
sequence number of the received packet has the same sequence number as the most
recently received packet) or a new packet (the sequence number changes, moving
forward in modulo-2 arithmetic). Since we are currently assuming a channel that
does not lose packets, ACK and NAK packets do not themselves need to indicate
the sequence number of the packet they are acknowledging. The sender knows that
a received ACK or NAK packet (whether garbled or not) was generated in response
to its most recently transmitted data packet.
210 CHAPTER 3 TRANSPORT LAYER
Figures 3.11 and 3.12 show the FSM description for rdt2.1, our fixed version
of rdt2.0. The rdt2.1 sender and receiver FSMs each now have twice as many
states as before. This is because the protocol state must now reflect whether the packet
currently being sent (by the sender) or expected (at the receiver) should have a
sequence number of 0 or 1. Note that the actions in those states where a 0-numbered
packet is being sent or expected are mirror images of those where a 1-numbered
packet is being sent or expected; the only differences have to do with the handling of
the sequence number.
Protocol rdt2.1 uses both positive and negative acknowledgments from the
receiver to the sender. When an out-of-order packet is received, the receiver sends a
positive acknowledgment for the packet it has received. When a corrupted packet is
received, the receiver sends a negative acknowledgment. We can accomplish the
same effect as a NAK if, instead of sending a NAK, we send an ACK for the last
correctly received packet. A sender that receives two ACKs for the same packet (that
is, receives duplicate ACKs) knows that the receiver did not correctly receive the
packet following the packet that is being ACKed twice. Our NAK-free reliable data
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 211
Wait for
call 0 from
above
rdt_rcv(rcvpkt)&&
(corrupt(rcvpkt)||
isNAK(rcvpkt))
udt_send(sndpkt)
rdt_rcv(rcvpkt)&&
(corrupt(rcvpkt)||
isNAK(rcvpkt))
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
rdt_send(data)
sndpkt=make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
sndpkt=make_pkt(1,data,checksum)
udt_send(sndpkt)
Wait for
ACK or
NAK 0
Wait for
ACK or
NAK 1
Wait for
call 1 from
above
Figure 3.11 rdt2.1 sender
transfer protocol for a channel with bit errors is rdt2.2, shown in Figures 3.13 and
3.14. One subtle change between rtdt2.1 and rdt2.2 is that the receiver must
now include the sequence number of the packet being acknowledged by an ACK
message (this is done by including the ACK,0 or ACK,1 argument in make_pkt()
in the receiver FSM), and the sender must now check the sequence number of the
packet being acknowledged by a received ACK message (this is done by including
the 0 or 1 argument in isACK()in the sender FSM).
Reliable Data Transfer over a Lossy Channel with Bit Errors: rdt3.0
Suppose now that in addition to corrupting bits, the underlying channel can lose
packets as well, a not-uncommon event in todays computer networks (including the
Internet). Two additional concerns must now be addressed by the protocol: how to
detect packet loss and what to do when packet loss occurs. The use of checksum-
ming, sequence numbers, ACK packets, and retransmissionsthe techniques
already developed in rdt2.2will allow us to answer the latter concern. Han-
dling the first concern will require adding a new protocol mechanism.
There are many possible approaches toward dealing with packet loss (several
more of which are explored in the exercises at the end of the chapter). Here, well
put the burden of detecting and recovering from lost packets on the sender. Suppose
212 CHAPTER 3 TRANSPORT LAYER
rdt_rcv(rcvpkt)&& notcorrupt
(rcvpkt)&&has_seq0(rcvpkt)
sndpkt=make_pkt(ACK,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt=make_pkt(NAK,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
sndpkt=make_pkt(NAK,checksum)
udt_send(sndpkt)
sndpkt=make_pkt(ACK,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt=make_pkt(ACK,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt)&& notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt=make_pkt(ACK,checksum)
udt_send(sndpkt)
Wait for
0 from
below
Wait for
1 from
belowrdt_rcv(rcvpkt)&& notcorrupt
(rcvpkt)&&has_seq1(rcvpkt)
Figure 3.12 rdt2.1 receiver
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 213
that the sender transmits a data packet and either that packet, or the receivers ACK
of that packet, gets lost. In either case, no reply is forthcoming at the sender from
the receiver. If the sender is willing to wait long enough so that it is certain that a
packet has been lost, it can simply retransmit the data packet. You should convince
yourself that this protocol does indeed work.
But how long must the sender wait to be certain that something has been lost?
The sender must clearly wait at least as long as a round-trip delay between the
sender and receiver (which may include buffering at intermediate routers) plus
whatever amount of time is needed to process a packet at the receiver. In many net-
works, this worst-case maximum delay is very difficult even to estimate, much less
know with certainty. Moreover, the protocol should ideally recover from packet
loss as soon as possible; waiting for a worst-case delay could mean a long wait
until error recovery is initiated. The approach thus adopted in practice is for the
sender to judiciously choose a time value such that packet loss is likely, although
not guaranteed, to have happened. If an ACK is not received within this time, the
packet is retransmitted. Note that if a packet experiences a particularly large delay,
the sender may retransmit the packet even though neither the data packet nor its
ACK have been lost. This introduces the possibility of duplicate data packets in
Wait for
call 0 from
above
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
isACK(rcvpkt,1))
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
isACK(rcvpkt,0))
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
rdt_send(data)
sndpkt=make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
sndpkt=make_pkt(1,data,checksum)
udt_send(sndpkt)
Wait for
ACK 0
Wait for
ACK 1
Wait for
call 1 from
above
Figure 3.13 rdt2.2 sender
the sender-to-receiver channel. Happily, protocol rdt2.2 already has enough
functionality (that is, sequence numbers) to handle the case of duplicate packets.
From the senders viewpoint, retransmission is a panacea. The sender does not
know whether a data packet was lost, an ACK was lost, or if the packet or ACK was
simply overly delayed. In all cases, the action is the same: retransmit. Implementing
a time-based retransmission mechanism requires a countdown timer that can
interrupt the sender after a given amount of time has expired. The sender will thus
need to be able to (1) start the timer each time a packet (either a first-time packet or
a retransmission) is sent, (2) respond to a timer interrupt (taking appropriate
actions), and (3) stop the timer.
Figure 3.15 shows the sender FSM for rdt3.0, a protocol that reliably transfers
data over a channel that can corrupt or lose packets; in the homework problems, youll
be asked to provide the receiver FSM for rdt3.0. Figure 3.16 shows how the proto-
col operates with no lost or delayed packets and how it handles lost data packets. In
Figure 3.16, time moves forward from the top of the diagram toward the bottom of the
diagram; note that a receive time for a packet is necessarily later than the send time
for a packet as a result of transmission and propagation delays. In Figures 3.16(b)(d),
the send-side brackets indicate the times at which a timer is set and later times out.
Several of the more subtle aspects of this protocol are explored in the exercises at the
end of this chapter. Because packet sequence numbers alternate between 0 and 1, pro-
tocol rdt3.0 is sometimes known as the alternating-bit protocol.
214 CHAPTER 3 TRANSPORT LAYER
Wait for
0 from
below
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
has_seq0(rcvpkt))
sndpkt=make_pkt(ACK,0,che
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
has_seq1(rcvpkt))
sndpkt=make_pkt(ACK,1,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt=make_pkt(ACK,1,checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt=make_pkt(ACK,0,checksum)
udt_send(sndpkt)
Wait for
1 from
below
Figure 3.14 rdt2.2 receiver
We have now assembled the key elements of a data transfer protocol. Check-
sums, sequence numbers, timers, and positive and negative acknowledgment pack-
ets each play a crucial and necessary role in the operation of the protocol. We now
have a working reliable data transfer protocol!
3.4.2 Pipelined Reliable Data Transfer Protocols
Protocol rdt3.0 is a functionally correct protocol, but it is unlikely that anyone would
be happy with its performance, particularly in todays high-speed networks. At the heart
of rdt3.0s performance problem is the fact that it is a stop-and-wait protocol.
To appreciate the performance impact of this stop-and-wait behavior, consider
an idealized case of two hosts, one located on the West Coast of the United States
and the other located on the East Coast, as shown in Figure 3.17. The speed-of-light
round-trip propagation delay between these two end systems, RTT, is approxi-
mately 30 milliseconds. Suppose that they are connected by a channel with a trans-
mission rate, R, of 1 Gbps (109 bits per second). With a packet size, L, of 1,000 bytes
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 215
Wait for
call 0 from
above
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
isACK(rcvpkt,1))
timeout
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
isACK(rcvpkt,0))
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
stop_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
stop_timer
timeout
udt_send(sndpkt)
start_timer
rdt_send(data)
sndpkt=make_pkt(0,data,checksum)
udt_send(sndpkt)
start_timer
rdt_send(data)
sndpkt=make_pkt(1,data,checksum)
udt_send(sndpkt)
start_timer
Wait for
ACK 0
Wait for
ACK 1
Wait for
call 1 from
above
rdt_rcv(rcvpkt)
Figure 3.15 rdt3.0 sender
VideoNote
Developing a protocol
and FSM representation
for a simple application-
layer protocol
216 CHAPTER 3 TRANSPORT LAYER
rcv pkt0
send ACK0
rcv pkt1
send ACK1
rcv pkt0
send ACK0
Sender Receiver
a. Operation with no loss
pkt0
ACK
0
pkt1
pkt0
ACK
1
ACK
0
(loss) X
b. Lost packet
rcv pkt0
send ACK0
rcv pkt1
send ACK1
c. Lost ACK
send pkt0
rcv ACK0
send pkt1
rcv ACK1
send pkt0
send pkt0
rcv ACK0
send pkt1
timeout
resend pkt1
rcv ACK1
send pkt0
rcv pkt0
send ACK0
rcv pkt1
(detect
duplicate)
send ACK1
send pkt0
rcv ACK0
send pkt1
rcv pkt0
send ACK0
timeout
resend pkt1
rcv pkt1
send ACK1
d. Premature timeout
rcv ACK1
send pkt0
rcv ACK1
do nothing
rcv pkt0
send ACK0
rcv pkt 1
(detect duplicate)
send ACK1
Sender Receiver ReceiverSender
pkt0
ACK
0
pkt1
ACK
1
ACK
1
ACK
0
ACK
1
ACK
0
pkt1
pkt0
pkt0
pkt1
pkt1
pkt0
ACK
1
ACK
0
X (loss)
pkt1
rcv pkt0
send ACK0
send pkt0
rcv ACK0
send pkt1
timeout
resend pkt1
rcv ACK1
send pkt0
rcv pkt0
send ACK0
rcv pkt1
send ACK1
Sender Receiver
pkt0
ACK
0
pkt1
pkt0
ACK
1
ACK
0
Figure 3.16 Operation of rdt3.0, the alternating-bit protocol
(8,000 bits) per packet, including both header fields and data, the time needed to
actually transmit the packet into the 1 Gbps link is
dtrans
Figure 3.18(a) shows that with our stop-and-wait protocol, if the sender begins
sending the packet at t = 0, then at t = L/R = 8 microseconds, the last bit enters the
channel at the sender side. The packet then makes its 15-msec cross-country jour-
ney, with the last bit of the packet emerging at the receiver at t = RTT/2 + L/R =
15.008 msec. Assuming for simplicity that ACK packets are extremely small (so that
we can ignore their transmission time) and that the receiver can send an ACK as
soon as the last bit of a data packet is received, the ACK emerges back at the sender
at t = RTT + L/R = 30.008 msec. At this point, the sender can now transmit the next
message. Thus, in 30.008 msec, the sender was sending for only 0.008 msec. If we
define the utilization of the sender (or the channel) as the fraction of time the sender
is actually busy sending bits into the channel, the analysis in Figure 3.18(a) shows
that the stop-and-wait protocol has a rather dismal sender utilization, Usender, of
That is, the sender was busy only 2.7 hundredths of one percent of the time!
Viewed another way, the sender was able to send only 1,000 bytes in 30.008 mil-
liseconds, an effective throughput of only 267 kbpseven though a 1 Gbps link was
available! Imagine the unhappy network manager who just paid a fortune for a giga-
bit capacity link but manages to get a throughput of only 267 kilobits per second!
This is a graphic example of how network protocols can limit the capabilities
Usender =
L>R
RTT + L>R
=
.008
30.008
= 0.00027
=
L
R
=
8000 bits>packet
109 bits/sec
= 8 microseconds
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 217
Data packetsData packet
ACK packets
a. A stop-and-wait protocol in operation b. A pipelined protocol in operation
Figure 3.17 Stop-and-wait versus pipelined protocol
provided by the underlying network hardware. Also, we have neglected lower-layer
protocol-processing times at the sender and receiver, as well as the processing and
queuing delays that would occur at any intermediate routers between the sender
and receiver. Including these effects would serve only to further increase the delay
and further accentuate the poor performance.
The solution to this particular performance problem is simple: Rather than oper-
ate in a stop-and-wait manner, the sender is allowed to send multiple packets with-
out waiting for acknowledgments, as illustrated in Figure 3.17(b). Figure 3.18(b)
shows that if the sender is allowed to transmit three packets before having to wait
for acknowledgments, the utilization of the sender is essentially tripled. Since the
many in-transit sender-to-receiver packets can be visualized as filling a pipeline, this
technique is known as pipelining. Pipelining has the following consequences for
reliable data transfer protocols:
The range of sequence numbers must be increased, since each in-transit packet
(not counting retransmissions) must have a unique sequence number and there
may be multiple, in-transit, unacknowledged packets.
The sender and receiver sides of the protocols may have to buffer more than one
packet. Minimally, the sender will have to buffer packets that have been trans-
mitted but not yet acknowledged. Buffering of correctly received packets may
also be needed at the receiver, as discussed below.
The range of sequence numbers needed and the buffering requirements will
depend on the manner in which a data transfer protocol responds to lost, cor-
rupted, and overly delayed packets. Two basic approaches toward pipelined error
recovery can be identified: Go-Back-N and selective repeat.
3.4.3 Go-Back-N (GBN)
In a Go-Back-N (GBN) protocol, the sender is allowed to transmit multiple packets
(when available) without waiting for an acknowledgment, but is constrained to have no
more than some maximum allowable number, N, of unacknowledged packets in the
pipeline. We describe the GBN protocol in some detail in this section. But before read-
ing on, you are encouraged to play with the GBN applet (an awesome applet!) at the
companion Web site.
Figure 3.19 shows the senders view of the range of sequence numbers in a GBN
protocol. If we define base to be the sequence number of the oldest unacknowledged
packet and nextseqnum to be the smallest unused sequence number (that is, the
sequence number of the next packet to be sent), then four intervals in the range of
sequence numbers can be identified. Sequence numbers in the interval [0,base-1]
correspond to packets that have already been transmitted and acknowledged. The inter-
val [base,nextseqnum-1] corresponds to packets that have been sent but not yet
acknowledged. Sequence numbers in the interval [nextseqnum,base+N-1] can
218 CHAPTER 3 TRANSPORT LAYER
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 219
First bit of first packet
transmitted, t = 0
Last bit of first packet
transmitted, t = L/R
First bit of first packet
transmitted, t = 0
Last bit of first packet
transmitted, t = L/R
ACK arrives, send next packet,
t = RTT + L/R
a. Stop-and-wait operation
Sender Receiver
RTT
First bit of first packet arrives
Last bit of first packet arrives, send ACK
First bit of first packet arrives
Last bit of first packet arrives, send ACK
ACK arrives, send next packet,
t = RTT + L/R
b. Pipelined operation
Sender Receiver
RTT
Last bit of 2nd packet arrives, send ACK
Last bit of 3rd packet arrives, send ACK
Figure 3.18 Stop-and-wait and pipelined sending
be used for packets that can be sent immediately, should data arrive from the upper
layer. Finally, sequence numbers greater than or equal to base+N cannot be used until
an unacknowledged packet currently in the pipeline (specifically, the packet with
sequence number base) has been acknowledged.
As suggested by Figure 3.19, the range of permissible sequence numbers for
transmitted but not yet acknowledged packets can be viewed as a window of size N
over the range of sequence numbers. As the protocol operates, this window slides
forward over the sequence number space. For this reason, N is often referred to as
the window size and the GBN protocol itself as a sliding-window protocol. You
might be wondering why we would even limit the number of outstanding, unac-
knowledged packets to a value of N in the first place. Why not allow an unlimited
number of such packets? Well see in Section 3.5 that flow control is one reason to
impose a limit on the sender. Well examine another reason to do so in Section 3.7,
when we study TCP congestion control.
In practice, a packets sequence number is carried in a fixed-length field in the
packet header. If k is the number of bits in the packet sequence number field, the range
of sequence numbers is thus [0,2k 1]. With a finite range of sequence numbers, all
arithmetic involving sequence numbers must then be done using modulo 2k arithmetic.
(That is, the sequence number space can be thought of as a ring of size 2k, where
sequence number 2k 1 is immediately followed by sequence number 0.) Recall that
rdt3.0 had a 1-bit sequence number and a range of sequence numbers of [0,1]. Sev-
eral of the problems at the end of this chapter explore the consequences of a finite range
of sequence numbers. We will see in Section 3.5 that TCP has a 32-bit sequence number
field, where TCP sequence numbers count bytes in the byte stream rather than packets.
Figures 3.20 and 3.21 give an extended FSM description of the sender and
receiver sides of an ACK-based, NAK-free, GBN protocol. We refer to this FSM
description as an extended FSM because we have added variables (similar to pro-
gramming-language variables) for base and nextseqnum, and added operations
on these variables and conditional actions involving these variables. Note that the
extended FSM specification is now beginning to look somewhat like a programming-
language specification. [Bochman 1984] provides an excellent survey of additional
extensions to FSM techniques as well as other programming-language-based tech-
niques for specifying protocols.
220 CHAPTER 3 TRANSPORT LAYER
base nextseqnum
Window size
N
Key:
Already
ACKd
Sent, not
yet ACKd
Usable,
not yet sent
Not usable
Figure 3.19 Senders view of sequence numbers in Go-Back-N
3.4 PRINCIPLES OF RELIABLE DATA TRANSFER 221
rdt_send(data)
if(nextseqnum
/* Assume sender is not constrained by TCP flow or congestion control, that data from above is less
than MSS in size, and that data transfer is in one direction only. */
NextSeqNum=InitialSeqNumber
SendBase=InitialSeqNumber
loop (forever) {
switch(event)
event: data received from application above
create TCP segment with sequence number NextSeqNum
if (timer currently not running)
start timer
pass segment to IP
NextSeqNum=NextSeqNum+length(data)
break;
event: timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number
start timer
break;
event: ACK received, with ACK field value of y
if (y > SendBase) {
SendBase=y
if (there are currently any not-yet-acknowledged segments)
start timer
}
break;
} /* end of loop forever */
244 CHAPTER 3 TRANSPORT LAYER
then the ACK is acknowledging one or more previously unacknowledged segments.
Thus the sender updates its SendBase variable; it also restarts the timer if there cur-
rently are any not-yet-acknowledged segments.
A Few Interesting Scenarios
We have just described a highly simplified version of how TCP provides reliable
data transfer. But even this highly simplified version has many subtleties. To get a
good feeling for how this protocol works, lets now walk through a few simple
scenarios. Figure 3.34 depicts the first scenario, in which Host A sends one seg-
ment to Host B. Suppose that this segment has sequence number 92 and contains 8
bytes of data. After sending this segment, Host A waits for a segment from B with
acknowledgment number 100. Although the segment from A is received at B, the
acknowledgment from B to A gets lost. In this case, the timeout event occurs, and
Host A retransmits the same segment. Of course, when Host B receives the
retransmission, it observes from the sequence number that the segment contains
data that has already been received. Thus, TCP in Host B will discard the bytes in
the retransmitted segment.
Time Time
Host A Host B
Timeout
Seq=92, 8 bytes data
Seq=92, 8 bytes data
ACK=
100
ACK=
100
X
(loss)
Figure 3.34 Retransmission due to a lost acknowledgment
3.5 CONNECTION-ORIENTED TRANSPORT: TCP 245
In a second scenario, shown in Figure 3.35, Host A sends two segments back to
back. The first segment has sequence number 92 and 8 bytes of data, and the second
segment has sequence number 100 and 20 bytes of data. Suppose that both segments
arrive intact at B, and B sends two separate acknowledgments for each of these seg-
ments. The first of these acknowledgments has acknowledgment number 100; the
second has acknowledgment number 120. Suppose now that neither of the acknowl-
edgments arrives at Host A before the timeout. When the timeout event occurs, Host
A resends the first segment with sequence number 92 and restarts the timer. As long
as the ACK for the second segment arrives before the new timeout, the second seg-
ment will not be retransmitted.
In a third and final scenario, suppose Host A sends the two segments, exactly as
in the second example. The acknowledgment of the first segment is lost in the
network, but just before the timeout event, Host A receives an acknowledgment with
acknowledgment number 120. Host A therefore knows that Host B has received
everything up through byte 119; so Host A does not resend either of the two
segments. This scenario is illustrated in Figure 3.36.
Time Time
Host A Host B
seq=92 timeout interval
Seq=92, 8 bytes data
Seq=100, 20 bytes data
AC
K=
10
0
ACK=
120
AC
K=
12
0
seq=92 timeout interval
Seq=92, 8 bytes data
Figure 3.35 Segment 100 not retransmitted
246 CHAPTER 3 TRANSPORT LAYER
Doubling the Timeout Interval
We now discuss a few modifications that most TCP implementations employ. The
first concerns the length of the timeout interval after a timer expiration. In this mod-
ification, whenever the timeout event occurs, TCP retransmits the not-yet-
acknowledged segment with the smallest sequence number, as described above. But
each time TCP retransmits, it sets the next timeout interval to twice the previous
value, rather than deriving it from the last EstimatedRTT and DevRTT (as
described in Section 3.5.3). For example, suppose TimeoutInterval associated
with the oldest not yet acknowledged segment is .75 sec when the timer first expires.
TCP will then retransmit this segment and set the new expiration time to 1.5 sec. If
the timer expires again 1.5 sec later, TCP will again retransmit this segment, now
setting the expiration time to 3.0 sec. Thus the intervals grow exponentially after
each retransmission. However, whenever the timer is started after either of the two
other events (that is, data received from application above, and ACK received), the
Time Time
Host A Host B
Seq=92 timeout interval
Seq=92, 8 bytes data
Seq=100,20 bytes data
ACK=
100
ACK=
120
X
(loss)
Figure 3.36 A cumulative acknowledgment avoids retransmission of the
first segment
3.5 CONNECTION-ORIENTED TRANSPORT: TCP 247
TimeoutInterval is derived from the most recent values of EstimatedRTT
and DevRTT.
This modification provides a limited form of congestion control. (More com-
prehensive forms of TCP congestion control will be studied in Section 3.7.) The
timer expiration is most likely caused by congestion in the network, that is, too
many packets arriving at one (or more) router queues in the path between the source
and destination, causing packets to be dropped and/or long queuing delays. In times
of congestion, if the sources continue to retransmit packets persistently, the conges-
tion may get worse. Instead, TCP acts more politely, with each sender retransmitting
after longer and longer intervals. We will see that a similar idea is used by Ethernet
when we study CSMA/CD in Chapter 5.
Fast Retransmit
One of the problems with timeout-triggered retransmissions is that the timeout
period can be relatively long. When a segment is lost, this long timeout period
forces the sender to delay resending the lost packet, thereby increasing the end-to-
end delay. Fortunately, the sender can often detect packet loss well before the time-
out event occurs by noting so-called duplicate ACKs. A duplicate ACK is an ACK
that reacknowledges a segment for which the sender has already received an earlier
acknowledgment. To understand the senders response to a duplicate ACK, we must
look at why the receiver sends a duplicate ACK in the first place. Table 3.2 summa-
rizes the TCP receivers ACK generation policy [RFC 5681]. When a TCP receiver
receives a segment with a sequence number that is larger than the next, expected,
in-order sequence number, it detects a gap in the data streamthat is, a missing seg-
ment. This gap could be the result of lost or reordered segments within the network.
Event TCP Receiver Action
Arrival of in-order segment with expected sequence number. All Delayed ACK. Wait up to 500 msec for arrival of another in-order seg-
data up to expected sequence number already acknowledged. ment. If next in-order segment does not arrive in this interval, send an ACK.
Arrival of in-order segment with expected sequence number. One Immediately send single cumulative ACK, ACKing both in-order segments.
other in-order segment waiting for ACK transmission.
Arrival of out-of-order segment with higher-than-expected sequence Immediately send duplicate ACK, indicating sequence number of next
number. Gap detected. expected byte (which is the lower end of the gap).
Arrival of segment that partially or completely fills in gap in Immediately send ACK, provided that segment starts at the lower end
received data. of gap.
Table 3.2 TCP ACK Generation Recommendation [RFC 5681]
248 CHAPTER 3 TRANSPORT LAYER
Since TCP does not use negative acknowledgments, the receiver cannot send an
explicit negative acknowledgment back to the sender. Instead, it simply reacknowl-
edges (that is, generates a duplicate ACK for) the last in-order byte of data it has
received. (Note that Table 3.2 allows for the case that the receiver does not discard
out-of-order segments.)
Because a sender often sends a large number of segments back to back, if one seg-
ment is lost, there will likely be many back-to-back duplicate ACKs. If the TCP sender
receives three duplicate ACKs for the same data, it takes this as an indication that the
segment following the segment that has been ACKed three times has been lost. (In the
homework problems, we consider the question of why the sender waits for three dupli-
cate ACKs, rather than just a single duplicate ACK.) In the case that three duplicate
ACKs are received, the TCP sender performs a fast retransmit [RFC 5681], retrans-
mitting the missing segment before that segments timer expires. This is shown in
Figure 3.37, where the second segment is lost, then retransmitted before its timer
expires. For TCP with fast retransmit, the following code snippet replaces the ACK
received event in Figure 3.33:
event: ACK received, with ACK field value of y
if (y > SendBase) {
SendBase=y
if (there are currently any not yet
acknowledged segments)
start timer
}
else { /* a duplicate ACK for already ACKed
segment */
increment number of duplicate ACKs
received for y
if (number of duplicate ACKS received
for y==3)
/* TCP fast retransmit */
resend segment with sequence number y
}
break;
We noted earlier that many subtle issues arise when a timeout/retransmit mech-
anism is implemented in an actual protocol such as TCP. The procedures above,
which have evolved as a result of more than 20 years of experience with TCP timers,
should convince you that this is indeed the case!
Go-Back-N or Selective Repeat?
Let us close our study of TCPs error-recovery mechanism by considering the follow-
ing question: Is TCP a GBN or an SR protocol? Recall that TCP acknowledgments are
cumulative and correctly received but out-of-order segments are not individually
3.5 CONNECTION-ORIENTED TRANSPORT: TCP 249
ACKed by the receiver. Consequently, as shown in Figure 3.33 (see also Figure 3.19),
the TCP sender need only maintain the smallest sequence number of a transmitted but
unacknowledged byte (SendBase) and the sequence number of the next byte to be
sent (NextSeqNum). In this sense, TCP looks a lot like a GBN-style protocol. But
there are some striking differences between TCP and Go-Back-N. Many TCP imple-
mentations will buffer correctly received but out-of-order segments [Stevens 1994].
Consider also what happens when the sender sends a sequence of segments 1, 2, . . . ,
N, and all of the segments arrive in order without error at the receiver. Further suppose
that the acknowledgment for packet n < N gets lost, but the remaining N 1 acknowl-edgments arrive at the sender before their respective timeouts. In this example, GBNwould retransmit not only packet n, but also all of the subsequent packets n + 1, n + 2,. . . , N. TCP, on the other hand, would retransmit at most one segment, namely, seg-ment n. Moreover, TCP would not even retransmit segment n if the acknowledgmentfor segment n + 1 arrived before the timeout for segment n.Host A Host Bseq=100, 20 bytes of dataTimeoutTime TimeXseq=100, 20 bytes of dataseq=92, 8 bytes of dataseq=120, 15 bytes of dataseq=135, 6 bytes of dataseq=141, 16 bytes of dataack=100ack=100ack=100ack=100Figure 3.37 Fast retransmit: retransmitting the missing segment beforethe segments timer expiresA proposed modification to TCP, the so-called selective acknowledgment[RFC 2018], allows a TCP receiver to acknowledge out-of-order segments selec-tively rather than just cumulatively acknowledging the last correctly received, in-order segment. When combined with selective retransmissionskipping theretransmission of segments that have already been selectively acknowledged by thereceiverTCP looks a lot like our generic SR protocol. Thus, TCPs error-recoverymechanism is probably best categorized as a hybrid of GBN and SR protocols.3.5.5 Flow ControlRecall that the hosts on each side of a TCP connection set aside a receive buffer forthe connection. When the TCP connection receives bytes that are correct and insequence, it places the data in the receive buffer. The associated application processwill read data from this buffer, but not necessarily at the instant the data arrives.Indeed, the receiving application may be busy with some other task and may noteven attempt to read the data until long after it has arrived. If the application is rela-tively slow at reading the data, the sender can very easily overflow the connectionsreceive buffer by sending too much data too quickly.TCP provides a flow-control service to its applications to eliminate the possibilityof the sender overflowing the receivers buffer. Flow control is thus a speed-matchingservicematching the rate at which the sender is sending against the rate at which thereceiving application is reading. As noted earlier, a TCP sender can also be throttleddue to congestion within the IP network; this form of sender control is referred to ascongestion control, a topic we will explore in detail in Sections 3.6 and 3.7. Eventhough the actions taken by flow and congestion control are similar (the throttling ofthe sender), they are obviously taken for very different reasons. Unfortunately, manyauthors use the terms interchangeably, and the savvy reader would be wise to distin-guish between them. Lets now discuss how TCP provides its flow-control service. Inorder to see the forest for the trees, we suppose throughout this section that the TCPimplementation is such that the TCP receiver discards out-of-order segments.TCP provides flow control by having the sender maintain a variable called thereceive window. Informally, the receive window is used to give the sender an idea ofhow much free buffer space is available at the receiver. Because TCP is full-duplex, thesender at each side of the connection maintains a distinct receive window. Lets investi-gate the receive window in the context of a file transfer. Suppose that Host A is sendinga large file to Host B over a TCP connection. Host B allocates a receive buffer to thisconnection; denote its size by RcvBuffer. From time to time, the application processin Host B reads from the buffer. Define the following variables: LastByteRead: the number of the last byte in the data stream read from thebuffer by the application process in B LastByteRcvd: the number of the last byte in the data stream that has arrivedfrom the network and has been placed in the receive buffer at B250 CHAPTER 3 TRANSPORT LAYER3.5 CONNECTION-ORIENTED TRANSPORT: TCP 251Because TCP is not permitted to overflow the allocated buffer, we must haveLastByteRcvd LastByteRead RcvBufferThe receive window, denoted rwnd is set to the amount of spare room in the buffer:rwnd = RcvBuffer [LastByteRcvd LastByteRead]Because the spare room changes with time, rwnd is dynamic. The variable rwnd isillustrated in Figure 3.38.How does the connection use the variable rwnd to provide the flow-controlservice? Host B tells Host A how much spare room it has in the connection bufferby placing its current value of rwnd in the receive window field of every segment itsends to A. Initially, Host B sets rwnd = RcvBuffer. Note that to pull this off,Host B must keep track of several connection-specific variables.Host A in turn keeps track of two variables, LastByteSent and Last-ByteAcked, which have obvious meanings. Note that the difference between thesetwo variables, LastByteSent LastByteAcked, is the amount of unac-knowledged data that A has sent into the connection. By keeping the amount ofunacknowledged data less than the value of rwnd, Host A is assured that it is notoverflowing the receive buffer at Host B. Thus, Host A makes sure throughout theconnections life thatLastByteSent LastByteAcked rwndApplicationprocessDatafrom IP TCP datain bufferrwndRcvBufferSpare roomFigure 3.38 The receive window (rwnd) and the receive buffer(RcvBuffer)There is one minor technical problem with this scheme. To see this, supposeHost Bs receive buffer becomes full so that rwnd = 0. After advertising rwnd = 0to Host A, also suppose that B has nothing to send to A. Now consider what hap-pens. As the application process at B empties the buffer, TCP does not send new seg-ments with new rwnd values to Host A; indeed, TCP sends a segment to Host Aonly if it has data to send or if it has an acknowledgment to send. Therefore, Host Ais never informed that some space has opened up in Host Bs receive bufferHostA is blocked and can transmit no more data! To solve this problem, the TCP specifi-cation requires Host A to continue to send segments with one data byte when Bsreceive window is zero. These segments will be acknowledged by the receiver.Eventually the buffer will begin to empty and the acknowledgments will contain anonzero rwnd value.The online site at http://www.awl.com/kurose-ross for this book provides aninteractive Java applet that illustrates the operation of the TCP receive window.Having described TCPs flow-control service, we briefly mention here that UDPdoes not provide flow control. To understand the issue, consider sending a series ofUDP segments from a process on Host A to a process on Host B. For a typical UDPimplementation, UDP will append the segments in a finite-sized buffer that precedesthe corresponding socket (that is, the door to the process). The process reads one entiresegment at a time from the buffer. If the process does not read the segments fastenough from the buffer, the buffer will overflow and segments will get dropped.3.5.6 TCP Connection ManagementIn this subsection we take a closer look at how a TCP connection is established andtorn down. Although this topic may not seem particularly thrilling, it is importantbecause TCP connection establishment can significantly add to perceived delays(for example, when surfing the Web). Furthermore, many of the most common net-work attacksincluding the incredibly popular SYN flood attackexploit vulnera-bilities in TCP connection management. Lets first take a look at how a TCPconnection is established. Suppose a process running in one host (client) wants toinitiate a connection with another process in another host (server). The client appli-cation process first informs the client TCP that it wants to establish a connection toa process in the server. The TCP in the client then proceeds to establish a TCP con-nection with the TCP in the server in the following manner: Step 1. The client-side TCP first sends a special TCP segment to the server-sideTCP. This special segment contains no application-layer data. But one of the flagbits in the segments header (see Figure 3.29), the SYN bit, is set to 1. For thisreason, this special segment is referred to as a SYN segment. In addition, the client randomly chooses an initial sequence number (client_isn) and putsthis number in the sequence number field of the initial TCP SYN segment. Thissegment is encapsulated within an IP datagram and sent to the server. There has252 CHAPTER 3 TRANSPORT LAYERhttp://www.awl.com/kurose-ross3.5 CONNECTION-ORIENTED TRANSPORT: TCP 253been considerable interest in properly randomizing the choice of theclient_isn in order to avoid certain security attacks [CERT 200109]. Step 2. Once the IP datagram containing the TCP SYN segment arrives at theserver host (assuming it does arrive!), the server extracts the TCP SYN segmentfrom the datagram, allocates the TCP buffers and variables to the connection, andsends a connection-granted segment to the client TCP. (Well see in Chapter 8 thatthe allocation of these buffers and variables before completing the third step of thethree-way handshake makes TCP vulnerable to a denial-of-service attack knownas SYN flooding.) This connection-granted segment also contains no application-layer data. However, it does contain three important pieces of information in thesegment header. First, the SYN bit is set to 1. Second, the acknowledgment fieldof the TCP segment header is set to client_isn+1. Finally, the serverchooses its own initial sequence number (server_isn) and puts this value inthe sequence number field of the TCP segment header. This connection-grantedsegment is saying, in effect, I received your SYN packet to start a connectionwith your initial sequence number, client_isn. I agree to establish this con-nection. My own initial sequence number is server_isn. The connection-granted segment is referred to as a SYNACK segment. Step 3. Upon receiving the SYNACK segment, the client also allocates buffersand variables to the connection. The client host then sends the server yet anothersegment; this last segment acknowledges the servers connection-granted seg-ment (the client does so by putting the value server_isn+1 in the acknowl-edgment field of the TCP segment header). The SYN bit is set to zero, since theconnection is established. This third stage of the three-way handshake may carryclient-to-server data in the segment payload.Once these three steps have been completed, the client and server hosts can sendsegments containing data to each other. In each of these future segments, the SYN bitwill be set to zero. Note that in order to establish the connection, three packets are sentbetween the two hosts, as illustrated in Figure 3.39. For this reason, this connection-establishment procedure is often referred to as a three-way handshake. Severalaspects of the TCP three-way handshake are explored in the homework problems(Why are initial sequence numbers needed? Why is a three-way handshake, asopposed to a two-way handshake, needed?). Its interesting to note that a rock climberand a belayer (who is stationed below the rock climber and whose job it is to handlethe climbers safety rope) use a three-way-handshake communication protocol that isidentical to TCPs to ensure that both sides are ready before the climber begins ascent.All good things must come to an end, and the same is true with a TCP connec-tion. Either of the two processes participating in a TCP connection can end the con-nection. When a connection ends, the resources (that is, the buffers and variables)in the hosts are deallocated. As an example, suppose the client decides to close theconnection, as shown in Figure 3.40. The client application process issues a close254 CHAPTER 3 TRANSPORT LAYERcommand. This causes the client TCP to send a special TCP segment to the serverprocess. This special segment has a flag bit in the segments header, the FIN bit(see Figure 3.29), set to 1. When the server receives this segment, it sends the clientan acknowledgment segment in return. The server then sends its own shutdownsegment, which has the FIN bit set to 1. Finally, the client acknowledges theservers shutdown segment. At this point, all the resources in the two hosts are nowdeallocated.During the life of a TCP connection, the TCP protocol running in each hostmakes transitions through various TCP states. Figure 3.41 illustrates a typicalsequence of TCP states that are visited by the client TCP. The client TCP begins inthe CLOSED state. The application on the client side initiates a new TCP connec-tion (by creating a Socket object in our Java examples as in the Python examplesfrom Chapter 2). This causes TCP in the client to send a SYN segment to TCP in theserver. After having sent the SYN segment, the client TCP enters the SYN_SENTstate. While in the SYN_SENT state, the client TCP waits for a segment from theserver TCP that includes an acknowledgment for the clients previous segment andhas the SYN bit set to 1. Having received such a segment, the client TCP enters theESTABLISHED state. While in the ESTABLISHED state, the TCP client can sendand receive TCP segments containing payload (that is, application-generated) data.Time TimeClient hostConnectionrequestConnectiongrantedServer hostSYN=1, seq=client_isnSYN=1, seq=server_isn,ack=client_isn+1SYN=0, seq=client_isn+1,ack=server_isn+1ACKFigure 3.39 TCP three-way handshake: segment exchange3.5 CONNECTION-ORIENTED TRANSPORT: TCP 255Suppose that the client application decides it wants to close the connection.(Note that the server could also choose to close the connection.) This causes theclient TCP to send a TCP segment with the FIN bit set to 1 and to enter theFIN_WAIT_1 state. While in the FIN_WAIT_1 state, the client TCP waits for a TCPsegment from the server with an acknowledgment. When it receives this segment,the client TCP enters the FIN_WAIT_2 state. While in the FIN_WAIT_2 state, theclient waits for another segment from the server with the FIN bit set to 1; afterreceiving this segment, the client TCP acknowledges the servers segment andenters the TIME_WAIT state. The TIME_WAIT state lets the TCP client resend thefinal acknowledgment in case the ACK is lost. The time spent in the TIME_WAITstate is implementation-dependent, but typical values are 30 seconds, 1 minute, and2 minutes. After the wait, the connection formally closes and all resources on theclient side (including port numbers) are released.Figure 3.42 illustrates the series of states typically visited by the server-sideTCP, assuming the client begins connection teardown. The transitions are self-explanatory. In these two state-transition diagrams, we have only shown how a TCPconnection is normally established and shut down. We have not described whatTime TimeClientCloseCloseServerFINACKACKFINClosedTimed waitFigure 3.40 Closing a TCP connection256 CHAPTER 3 TRANSPORT LAYERCLOSEDSYN_SENTESTABLISHEDFIN_WAIT_1FIN_WAIT_2TIME_WAITSend SYNSend FINReceive ACK, send nothingWait 30 secondsReceive FIN, send ACKReceive SYN & ACK, send ACKClient applicationinitiates a TCP connectionClient applicationinitiates close connectionFigure 3.41 A typical sequence of TCP states visited by a client TCPCLOSEDLISTENSYN_RCVDESTABLISHEDCLOSE_WAITLAST_ACKReceive FIN,send ACKReceive ACK, send nothingSend FINReceive SYN send SYN & ACKServer applicationcreates a listen socketReceive ACK, send nothingFigure 3.42 A typical sequence of TCP states visited by a server-side TCP3.5 CONNECTION-ORIENTED TRANSPORT: TCP 257THE SYN FLOOD ATTACKWeve seen in our discussion of TCPs three-way handshake that a server allocatesand initializes connection variables and buffers in response to a received SYN. Theserver then sends a SYNACK in response, and awaits an ACK segment from theclient. If the client does not send an ACK to complete the third step of this 3-wayhandshake, eventually (often after a minute or more) the server will terminate the half-open connection and reclaim the allocated resources. This TCP connection management protocol sets the stage for a classic Denial ofService (DoS) attack known as the SYN flood attack. In this attack, the attacker(s)send a large number of TCP SYN segments, without completing the third handshakestep. With this deluge of SYN segments, the servers connection resources becomeexhausted as they are allocated (but never used!) for half-open connections; legiti-mate clients are then denied service. Such SYN flooding attacks were among the firstdocumented DoS attacks [CERT SYN 1996]. Fortunately, an effective defense knownas SYN cookies [RFC 4987] are now deployed in most major operating systems.SYN cookies work as follows:oWhen the server receives a SYN segment, it does not know if the segment is com-ing from a legitimate user or is part of a SYN flood attack. So, instead of creatinga half-open TCP connection for this SYN, the server creates an initial TCPsequence number that is a complicated function (hash function) of source and des-tination IP addresses and port numbers of the SYN segment, as well as a secretnumber only known to the server. This carefully crafted initial sequence number isthe so-called cookie. The server then sends the client a SYNACK packet with thisspecial initial sequence number. Importantly, the server does not remember thecookie or any other state information corresponding to the SYN.oA legitimate client will return an ACK segment. When the server receives thisACK, it must verify that the ACK corresponds to some SYN sent earlier. But how isthis done if the server maintains no memory about SYN segments? As you mayhave guessed, it is done with the cookie. Recall that for a legitimate ACK, thevalue in the acknowledgment field is equal to the initial sequence number in theSYNACK (the cookie value in this case) plus one (see Figure 3.39). The server canthen run the same hash function using the source and destination IP address andport numbers in the SYNACK (which are the same as in the original SYN) and thesecret number. If the result of the function plus one is the same as the acknowledg-ment (cookie) value in the clients SYNACK, the server concludes that the ACK cor-responds to an earlier SYN segment and is hence valid. The server then creates afully open connection along with a socket.oOn the other hand, if the client does not return an ACK segment, then the originalSYN has done no harm at the server, since the server hasnt yet allocated anyresources in response to the original bogus SYN.FOCUS ON SECURITY258 CHAPTER 3 TRANSPORT LAYERhappens in certain pathological scenarios, for example, when both sides of a con-nection want to initiate or shut down at the same time. If you are interested in learn-ing about this and other advanced issues concerning TCP, you are encouraged to seeStevens comprehensive book [Stevens 1994].Our discussion above has assumed that both the client and server are preparedto communicate, i.e., that the server is listening on the port to which the client sendsits SYN segment. Lets consider what happens when a host receives a TCP segmentwhose port numbers or source IP address do not match with any of the ongoingsockets in the host. For example, suppose a host receives a TCP SYN packet withdestination port 80, but the host is not accepting connections on port 80 (that is, it isnot running a Web server on port 80). Then the host will send a special reset seg-ment to the source. This TCP segment has the RST flag bit (see Section 3.5.2) set to1. Thus, when a host sends a reset segment, it is telling the source I dont have asocket for that segment. Please do not resend the segment. When a host receives aUDP packet whose destination port number doesnt match with an ongoing UDPsocket, the host sends a special ICMP datagram, as discussed in Chapter 4.Now that we have a good understanding of TCP connection management, letsrevisit the nmap port-scanning tool and examine more closely how it works. To explorea specific TCP port, say port 6789, on a target host, nmap will send a TCP SYN seg-ment with destination port 6789 to that host. There are three possible outcomes: The source host receives a TCP SYNACK segment from the target host. Since thismeans that an application is running with TCP port 6789 on the target post, nmapreturns open. The source host receives a TCP RST segment from the target host. This means thatthe SYN segment reached the target host, but the target host is not running an appli-cation with TCP port 6789. But the attacker at least knows that the segments des-tined to the host at port 6789 are not blocked by any firewall on the path betweensource and target hosts. (Firewalls are discussed in Chapter 8.) The source receives nothing. This likely means that the SYN segment was blockedby an intervening firewall and never reached the target host.Nmap is a powerful tool, which can case the joint not only for open TCP ports,but also for open UDP ports, for firewalls and their configurations, and even for the ver-sions of applications and operating systems. Most of this is done by manipulating TCPconnection-management segments [Skoudis 2006].You can download nmap fromwww.nmap.org.This completes our introduction to error control and flow control in TCP. InSection 3.7 well return to TCP and look at TCP congestion control in some depth.Before doing so, however, we first step back and examine congestion-control issuesin a broader context.www.nmap.org3.6 PRINCIPLES OF CONGESTION CONTROL 2593.6 Principles of Congestion ControlIn the previous sections, we examined both the general principles and specificTCP mechanisms used to provide for a reliable data transfer service in the face ofpacket loss. We mentioned earlier that, in practice, such loss typically results fromthe overflowing of router buffers as the network becomes congested. Packetretransmission thus treats a symptom of network congestion (the loss of a specifictransport-layer segment) but does not treat the cause of network congestiontoomany sources attempting to send data at too high a rate. To treat the cause of net-work congestion, mechanisms are needed to throttle senders in the face of networkcongestion.In this section, we consider the problem of congestion control in a general con-text, seeking to understand why congestion is a bad thing, how network congestionis manifested in the performance received by upper-layer applications, and variousapproaches that can be taken to avoid, or react to, network congestion. This moregeneral study of congestion control is appropriate since, as with reliable data trans-fer, it is high on our top-ten list of fundamentally important problems in network-ing. We conclude this section with a discussion of congestion control in theavailable bit-rate (ABR) service in asynchronous transfer mode (ATM)networks. The following section contains a detailed study of TCPs congestion-control algorithm.3.6.1 The Causes and the Costs of CongestionLets begin our general study of congestion control by examining three increasinglycomplex scenarios in which congestion occurs. In each case, well look at why con-gestion occurs in the first place and at the cost of congestion (in terms of resourcesnot fully utilized and poor performance received by the end systems). Well not (yet)focus on how to react to, or avoid, congestion but rather focus on the simpler issueof understanding what happens as hosts increase their transmission rate and the net-work becomes congested.Scenario 1: Two Senders, a Router with Infinite BuffersWe begin by considering perhaps the simplest congestion scenario possible: Twohosts (A and B) each have a connection that shares a single hop between source anddestination, as shown in Figure 3.43.Lets assume that the application in Host A is sending data into the connection(for example, passing data to the transport-level protocol via a socket) at an aver-age rate of in bytes/sec. These data are original in the sense that each unit of datais sent into the socket only once. The underlying transport-level protocol is a260 CHAPTER 3 TRANSPORT LAYERsimple one. Data is encapsulated and sent; no error recovery (for example, retrans-mission), flow control, or congestion control is performed. Ignoring the additionaloverhead due to adding transport- and lower-layer header information, the rate atwhich Host A offers traffic to the router in this first scenario is thus in bytes/sec.Host B operates in a similar manner, and we assume for simplicity that it too issending at a rate of in bytes/sec. Packets from Hosts A and B pass through arouter and over a shared outgoing link of capacity R. The router has buffers thatallow it to store incoming packets when the packet-arrival rate exceeds the outgo-ing links capacity. In this first scenario, we assume that the router has an infiniteamount of buffer space.Figure 3.44 plots the performance of Host As connection under this firstscenario. The left graph plots the per-connection throughput (number of bytes persecond at the receiver) as a function of the connection-sending rate. For a sendingrate between 0 and R/2, the throughput at the receiver equals the senders sendingrateeverything sent by the sender is received at the receiver with a finite delay.When the sending rate is above R/2, however, the throughput is only R/2. This upperlimit on throughput is a consequence of the sharing of link capacity between twoconnections. The link simply cannot deliver packets to a receiver at a steady-staterate that exceeds R/2. No matter how high Hosts A and B set their sending rates,they will each never see a throughput higher than R/2.Achieving a per-connection throughput of R/2 might actually appear to be agood thing, because the link is fully utilized in delivering packets to their destina-tions. The right-hand graph in Figure 3.44, however, shows the consequence ofoperating near link capacity. As the sending rate approaches R/2 (from the left), theaverage delay becomes larger and larger. When the sending rate exceeds R/2, theHost BUnlimited sharedoutput link buffersin: original dataHost A Host DHost CoutFigure 3.43 Congestion scenario 1: Two connections sharing a singlehop with infinite buffers3.6 PRINCIPLES OF CONGESTION CONTROL 261average number of queued packets in the router is unbounded, and the average delaybetween source and destination becomes infinite (assuming that the connectionsoperate at these sending rates for an infinite period of time and there is an infiniteamount of buffering available). Thus, while operating at an aggregate throughput ofnear R may be ideal from a throughput standpoint, it is far from ideal from a delaystandpoint. Even in this (extremely) idealized scenario, weve already found onecost of a congested networklarge queuing delays are experienced as the packet-arrival rate nears the link capacity.Scenario 2: Two Senders and a Router with Finite BuffersLet us now slightly modify scenario 1 in the following two ways (see Figure 3.45).First, the amount of router buffering is assumed to be finite. A consequence of thisreal-world assumption is that packets will be dropped when arriving to an already-full buffer. Second, we assume that each connection is reliable. If a packet contain-ing a transport-level segment is dropped at the router, the sender will eventuallyretransmit it. Because packets can be retransmitted, we must now be more carefulwith our use of the term sending rate. Specifically, let us again denote the rate atwhich the application sends original data into the socket by in bytes/sec. The rate atwhich the transport layer sends segments (containing original data and retransmit-ted data) into the network will be denoted in bytes/sec. in is sometimes referred toas the offered load to the network.The performance realized under scenario 2 will now depend strongly onhow retransmission is performed. First, consider the unrealistic case that Host A isable to somehow (magically!) determine whether or not a buffer is free in the routerand thus sends a packet only when a buffer is free. In this case, no loss would occur,R/2R/2DelayR/2in in outa. b.Figure 3.44 Congestion scenario 1: Throughput and delay as a functionof host sending rate262 CHAPTER 3 TRANSPORT LAYERin would be equal to in, and the throughput of the connection would be equal toin. This case is shown in Figure 3.46(a). From a throughput standpoint, perform-ance is idealeverything that is sent is received. Note that the average host sendingrate cannot exceed R/2 under this scenario, since packet loss is assumed neverto occur.Consider next the slightly more realistic case that the sender retransmits onlywhen a packet is known for certain to be lost. (Again, this assumption is a bit of astretch. However, it is possible that the sending host might set its timeout largeenough to be virtually assured that a packet that has not been acknowledged hasbeen lost.) In this case, the performance might look something like that shown inFigure 3.46(b). To appreciate what is happening here, consider the case that theoffered load, in (the rate of original data transmission plus retransmissions), equalsR/2. According to Figure 3.46(b), at this value of the offered load, the rate at whichdata are delivered to the receiver application is R/3. Thus, out of the 0.5R units ofdata transmitted, 0.333R bytes/sec (on average) are original data and 0.166R bytes/sec (on average) are retransmitted data. We see here another cost of a congested net-workthe sender must perform retransmissions in order to compensate for dropped(lost) packets due to buffer overflow.Finally, let us consider the case that the sender may time out prematurely andretransmit a packet that has been delayed in the queue but not yet lost. In this case,both the original data packet and the retransmission may reach the receiver. OfFinite shared outputlink buffersHost BHost A Host DHost Coutin: original datain: original data, plusretransmitted dataFigure 3.45 Scenario 2: Two hosts (with retransmissions) and a routerwith finite buffers3.6 PRINCIPLES OF CONGESTION CONTROL 263course, the receiver needs but one copy of this packet and will discard the retrans-mission. In this case, the work done by the router in forwarding the retransmittedcopy of the original packet was wasted, as the receiver will have already receivedthe original copy of this packet. The router would have better used the link trans-mission capacity to send a different packet instead. Here then is yet another cost ofa congested networkunneeded retransmissions by the sender in the face of largedelays may cause a router to use its link bandwidth to forward unneeded copies of apacket. Figure 3.46 (c) shows the throughput versus offered load when each packetis assumed to be forwarded (on average) twice by the router. Since each packet isforwarded twice, the throughput will have an asymptotic value of R/4 as the offeredload approaches R/2.Scenario 3: Four Senders, Routers with Finite Buffers, and Multihop PathsIn our final congestion scenario, four hosts transmit packets, each over overlap-ping two-hop paths, as shown in Figure 3.47. We again assume that each hostuses a timeout/retransmission mechanism to implement a reliable data transferservice, that all hosts have the same value of in, and that all router links havecapacity R bytes/sec.Lets consider the connection from Host A to Host C, passing through routersR1 and R2. The AC connection shares router R1 with the DB connection andshares router R2 with the BD connection. For extremely small values of in, bufferoverflows are rare (as in congestion scenarios 1 and 2), and the throughput approxi-mately equals the offered load. For slightly larger values of in, the correspondingthroughput is also larger, since more original data is being transmitted into theR/2R/2 R/2 outa. b.R/2 outR/3R/2R/2 outR/4c.in in inFigure 3.46 Scenario 2 performance with finite buffers264 CHAPTER 3 TRANSPORT LAYERnetwork and delivered to the destination, and overflows are still rare. Thus, for smallvalues of in, an increase in in results in an increase in out.Having considered the case of extremely low traffic, lets next examine thecase that in (and hence in) is extremely large. Consider router R2. The ACtraffic arriving to router R2 (which arrives at R2 after being forwarded from R1)can have an arrival rate at R2 that is at most R, the capacity of the link from R1to R2, regardless of the value of in. If in is extremely large for all connections(including the BD connection), then the arrival rate of BD traffic at R2 can bemuch larger than that of the AC traffic. Because the AC and BD traffic mustcompete at router R2 for the limited amount of buffer space, the amount of ACtraffic that successfully gets through R2 (that is, is not lost due to buffer over-flow) becomes smaller and smaller as the offered load from BD gets larger andlarger. In the limit, as the offered load approaches infinity, an empty buffer at R2Host BHost AR1R4 R2R3Host CHost DFinite shared outputlink buffersin: original datain : originaldata, plusretransmitteddataoutFigure 3.47 Four senders, routers with finite buffers, and multihop paths3.6 PRINCIPLES OF CONGESTION CONTROL 265is immediately filled by a BD packet, and the throughput of the AC connectionat R2 goes to zero. This, in turn, implies that the AC end-to-end throughput goesto zero in the limit of heavy traffic. These considerations give rise to the offeredload versus throughput tradeoff shown in Figure 3.48.The reason for the eventual decrease in throughput with increasing offeredload is evident when one considers the amount of wasted work done by the net-work. In the high-traffic scenario outlined above, whenever a packet is droppedat a second-hop router, the work done by the first-hop router in forwarding apacket to the second-hop router ends up being wasted. The network wouldhave been equally well off (more accurately, equally bad off) if the first routerhad simply discarded that packet and remained idle. More to the point, the trans-mission capacity used at the first router to forward the packet to the second routercould have been much more profitably used to transmit a different packet. (Forexample, when selecting a packet for transmission, it might be better for a routerto give priority to packets that have already traversed some number of upstreamrouters.) So here we see yet another cost of dropping a packet due to conges-tionwhen a packet is dropped along a path, the transmission capacity that wasused at each of the upstream links to forward that packet to the point at which itis dropped ends up having been wasted.3.6.2 Approaches to Congestion ControlIn Section 3.7, well examine TCPs specific approach to congestion control in greatdetail. Here, we identify the two broad approaches to congestion control that aretaken in practice and discuss specific network architectures and congestion-controlprotocols embodying these approaches.R/2 outinFigure 3.48 Scenario 3 performance with finite buffers and multihoppaths266 CHAPTER 3 TRANSPORT LAYERAt the broadest level, we can distinguish among congestion-control approachesby whether the network layer provides any explicit assistance to the transport layerfor congestion-control purposes: End-to-end congestion control. In an end-to-end approach to congestion control,the network layer provides no explicit support to the transport layer for congestion-control purposes. Even the presence of congestion in the network must be inferredby the end systems based only on observed network behavior (for example, packetloss and delay). We will see in Section 3.7 that TCP must necessarily take this end-to-end approach toward congestion control, since the IP layer provides no feedbackto the end systems regarding network congestion. TCP segment loss (as indicatedby a timeout or a triple duplicate acknowledgment) is taken as an indication of net-work congestion and TCP decreases its window size accordingly. We will also seea more recent proposal for TCP congestion control that uses increasing round-tripdelay values as indicators of increased network congestion. Network-assisted congestion control. With network-assisted congestion control,network-layer components (that is, routers) provide explicit feedback to thesender regarding the congestion state in the network. This feedback may be assimple as a single bit indicating congestion at a link. This approach was taken inthe early IBM SNA [Schwartz 1982] and DEC DECnet [Jain 1989; Ramakrish-nan 1990] architectures, was recently proposed for TCP/IP networks [Floyd TCP1994; RFC 3168], and is used in ATM available bit-rate (ABR) congestion con-trol as well, as discussed below. More sophisticated network feedback is also pos-sible. For example, one form of ATM ABR congestion control that we will studyshortly allows a router to inform the sender explicitly of the transmission rate it(the router) can support on an outgoing link. The XCP protocol [Katabi 2002] pro-vides router-computed feedback to each source, carried in the packet header,regarding how that source should increase or decrease its transmission rate.For network-assisted congestion control, congestion information is typically fedback from the network to the sender in one of two ways, as shown in Figure 3.49.Direct feedback may be sent from a network router to the sender. This form of notifi-cation typically takes the form of a choke packet (essentially saying, Im con-gested!). The second form of notification occurs when a router marks/updates a fieldin a packet flowing from sender to receiver to indicate congestion. Upon receipt of amarked packet, the receiver then notifies the sender of the congestion indication.Note that this latter form of notification takes at least a full round-trip time.3.6.3 Network-Assisted Congestion-Control Example:ATM ABR Congestion ControlWe conclude this section with a brief case study of the congestion-control algorithmin ATM ABRa protocol that takes a network-assisted approach toward congestioncontrol. We stress that our goal here is not to describe aspects of the ATM architecturein great detail, but rather to illustrate a protocol that takes a markedly differentapproach toward congestion control from that of the Internets TCP protocol. Indeed,we only present below those few aspects of the ATM architecture that are needed tounderstand ABR congestion control.Fundamentally ATM takes a virtual-circuit (VC) oriented approach towardpacket switching. Recall from our discussion in Chapter 1, this means that eachswitch on the source-to-destination path will maintain state about the source-to-destination VC. This per-VC state allows a switch to track the behavior of indi-vidual senders (e.g., tracking their average transmission rate) and to takesource-specific congestion-control actions (such as explicitly signaling to thesender to reduce its rate when the switch becomes congested). This per-VC stateat network switches makes ATM ideally suited to perform network-assisted con-gestion control.ABR has been designed as an elastic data transfer service in a manner reminis-cent of TCP. When the network is underloaded, ABR service should be able to takeadvantage of the spare available bandwidth; when the network is congested, ABRservice should throttle its transmission rate to some predetermined minimum trans-mission rate. A detailed tutorial on ATM ABR congestion control and traffic man-agement is provided in [Jain 1996].Figure 3.50 shows the framework for ATM ABR congestion control. In ourdiscussion we adopt ATM terminology (for example, using the term switch ratherthan router, and the term cell rather than packet). With ATM ABR service, datacells are transmitted from a source to a destination through a series of intermedi-ate switches. Interspersed with the data cells are resource-management cells3.6 PRINCIPLES OF CONGESTION CONTROL 267Host ANetwork feedback via receiverDirect networkfeedbackHost BFigure 3.49 Two feedback pathways for network-indicated congestioninformation268 CHAPTER 3 TRANSPORT LAYER(RM cells); these RM cells can be used to convey congestion-related informationamong the hosts and switches. When an RM cell arrives at a destination, it willbe turned around and sent back to the sender (possibly after the destination hasmodified the contents of the RM cell). It is also possible for a switch to generatean RM cell itself and send this RM cell directly to a source. RM cells can thus beused to provide both direct network feedback and network feedback via thereceiver, as shown in Figure 3.50.ATM ABR congestion control is a rate-based approach. That is, the senderexplicitly computes a maximum rate at which it can send and regulates itself accord-ingly. ABR provides three mechanisms for signaling congestion-related informationfrom the switches to the receiver: EFCI bit. Each data cell contains an explicit forward congestion indication(EFCI) bit. A congested network switch can set the EFCI bit in a data cell to1 to signal congestion to the destination host. The destination must check theEFCI bit in all received data cells. When an RM cell arrives at the destination,if the most recently received data cell had the EFCI bit set to 1, then the desti-nation sets the congestion indication bit (the CI bit) of the RM cell to 1 andsends the RM cell back to the sender. Using the EFCI in data cells and the CIbit in RM cells, a sender can thus be notified about congestion at a networkswitch. CI and NI bits. As noted above, sender-to-receiver RM cells are interspersedwith data cells. The rate of RM cell interspersion is a tunable parameter, withthe default value being one RM cell every 32 data cells. These RM cells have acongestion indication (CI) bit and a no increase (NI) bit that can be set by aSource DestinationSwitch SwitchKey:RM cells Data cellsFigure 3.50 Congestion-control framework for ATM ABR servicecongested network switch. Specifically, a switch can set the NI bit in a passingRM cell to 1 under mild congestion and can set the CI bit to 1 under severecongestion conditions. When a destination host receives an RM cell, it willsend the RM cell back to the sender with its CI and NI bits intact (except thatCI may be set to 1 by the destination as a result of the EFCI mechanismdescribed above). ER setting. Each RM cell also contains a 2-byte explicit rate (ER) field. A con-gested switch may lower the value contained in the ER field in a passing RMcell. In this manner, the ER field will be set to the minimum supportable rate ofall switches on the source-to-destination path.An ATM ABR source adjusts the rate at which it can send cells as a function ofthe CI, NI, and ER values in a returned RM cell. The rules for making this rateadjustment are rather complicated and a bit tedious. The interested reader is referredto [Jain 1996] for details.3.7 TCP Congestion ControlIn this section we return to our study of TCP. As we learned in Section 3.5, TCP pro-vides a reliable transport service between two processes running on different hosts.Another key component of TCP is its congestion-control mechanism. As indicatedin the previous section, TCP must use end-to-end congestion control rather than net-work-assisted congestion control, since the IP layer provides no explicit feedback tothe end systems regarding network congestion.The approach taken by TCP is to have each sender limit the rate at which itsends traffic into its connection as a function of perceived network congestion. If aTCP sender perceives that there is little congestion on the path between itself andthe destination, then the TCP sender increases its send rate; if the sender perceivesthat there is congestion along the path, then the sender reduces its send rate. But thisapproach raises three questions. First, how does a TCP sender limit the rate at whichit sends traffic into its connection? Second, how does a TCP sender perceive thatthere is congestion on the path between itself and the destination? And third, whatalgorithm should the sender use to change its send rate as a function of perceivedend-to-end congestion?Lets first examine how a TCP sender limits the rate at which it sends trafficinto its connection. In Section 3.5 we saw that each side of a TCP connection consistsof a receive buffer, a send buffer, and several variables (LastByteRead, rwnd,and so on). The TCP congestion-control mechanism operating at the sender keepstrack of an additional variable, the congestion window. The congestion window,denoted cwnd, imposes a constraint on the rate at which a TCP sender can send traffic3.7 TCP CONGESTION CONTROL 269270 CHAPTER 3 TRANSPORT LAYERinto the network. Specifically, the amount of unacknowledged data at a sender maynot exceed the minimum of cwnd and rwnd, that is:LastByteSent LastByteAcked min{cwnd, rwnd}In order to focus on congestion control (as opposed to flow control), let us hence-forth assume that the TCP receive buffer is so large that the receive-window con-straint can be ignored; thus, the amount of unacknowledged data at the sender issolely limited by cwnd. We will also assume that the sender always has data tosend, i.e., that all segments in the congestion window are sent.The constraint above limits the amount of unacknowledged data at the senderand therefore indirectly limits the senders send rate. To see this, consider a connec-tion for which loss and packet transmission delays are negligible. Then, roughly, atthe beginning of every RTT, the constraint permits the sender to send cwnd bytes ofdata into the connection; at the end of the RTT the sender receives acknowledg-ments for the data. Thus the senders send rate is roughly cwnd/RTT bytes/sec. Byadjusting the value of cwnd, the sender can therefore adjust the rate at which itsends data into its connection.Lets next consider how a TCP sender perceives that there is congestion onthe path between itself and the destination. Let us define a loss event at a TCPsender as the occurrence of either a timeout or the receipt of three duplicateACKs from the receiver. (Recall our discussion in Section 3.5.4 of the timeoutevent in Figure 3.33 and the subsequent modification to include fast retransmiton receipt of three duplicate ACKs.) When there is excessive congestion, thenone (or more) router buffers along the path overflows, causing a datagram (con-taining a TCP segment) to be dropped. The dropped datagram, in turn, results ina loss event at the sendereither a timeout or the receipt of three duplicateACKswhich is taken by the sender to be an indication of congestion on thesender-to-receiver path.Having considered how congestion is detected, lets next consider the moreoptimistic case when the network is congestion-free, that is, when a loss eventdoesnt occur. In this case, acknowledgments for previously unacknowledgedsegments will be received at the TCP sender. As well see, TCP will take thearrival of these acknowledgments as an indication that all is wellthat segmentsbeing transmitted into the network are being successfully delivered to thedestinationand will use acknowledgments to increase its congestion windowsize (and hence its transmission rate). Note that if acknowledgments arrive at arelatively slow rate (e.g., if the end-end path has high delay or contains a low-bandwidth link), then the congestion window will be increased at a relativelyslow rate. On the other hand, if acknowledgments arrive at a high rate, then thecongestion window will be increased more quickly. Because TCP usesacknowledgments to trigger (or clock) its increase in congestion window size,TCP is said to be self-clocking.Given the mechanism of adjusting the value of cwnd to control the sending rate,the critical question remains: How should a TCP sender determine the rate at whichit should send? If TCP senders collectively send too fast, they can congest the net-work, leading to the type of congestion collapse that we saw in Figure 3.48. Indeed,the version of TCP that well study shortly was developed in response to observedInternet congestion collapse [Jacobson 1988] under earlier versions of TCP. How-ever, if TCP senders are too cautious and send too slowly, they could under utilizethe bandwidth in the network; that is, the TCP senders could send at a higher ratewithout congesting the network. How then do the TCP senders determine their send-ing rates such that they dont congest the network but at the same time make use ofall the available bandwidth? Are TCP senders explicitly coordinated, or is there adistributed approach in which the TCP senders can set their sending rates based onlyon local information? TCP answers these questions using the following guidingprinciples: A lost segment implies congestion, and hence, the TCP senders rate should bedecreased when a segment is lost. Recall from our discussion in Section 3.5.4,that a timeout event or the receipt of four acknowledgments for a given seg-ment (one original ACK and then three duplicate ACKs) is interpreted as animplicit loss event indication of the segment following the quadruply ACKedsegment, triggering a retransmission of the lost segment. From a congestion-control standpoint, the question is how the TCP sender should decrease its con-gestion window size, and hence its sending rate, in response to this inferredloss event. An acknowledged segment indicates that the network is delivering the senderssegments to the receiver, and hence, the senders rate can be increased when anACK arrives for a previously unacknowledged segment. The arrival of acknowl-edgments is taken as an implicit indication that all is wellsegments are beingsuccessfully delivered from sender to receiver, and the network is thus not con-gested. The congestion window size can thus be increased. Bandwidth probing. Given ACKs indicating a congestion-free source-to-destinationpath and loss events indicating a congested path, TCPs strategy for adjusting itstransmission rate is to increase its rate in response to arriving ACKs until a lossevent occurs, at which point, the transmission rate is decreased. The TCP senderthus increases its transmission rate to probe for the rate that at which congestiononset begins, backs off from that rate, and then to begins probing again to see ifthe congestion onset rate has changed. The TCP senders behavior is perhaps anal-ogous to the child who requests (and gets) more and more goodies until finallyhe/she is finally told No!, backs off a bit, but then begins making requests3.7 TCP CONGESTION CONTROL 271272 CHAPTER 3 TRANSPORT LAYERagain shortly afterwards. Note that there is no explicit signaling of congestionstate by the networkACKs and loss events serve as implicit signalsand thateach TCP sender acts on local information asynchronously from other TCPsenders.Given this overview of TCP congestion control, were now in a position to considerthe details of the celebrated TCP congestion-control algorithm, which was firstdescribed in [Jacobson 1988] and is standardized in [RFC 5681]. The algorithm hasthree major components: (1) slow start, (2) congestion avoidance, and (3) fast recov-ery. Slow start and congestion avoidance are mandatory components of TCP, differ-ing in how they increase the size of cwnd in response to received ACKs. Well seeshortly that slow start increases the size of cwnd more rapidly (despite its name!)than congestion avoidance. Fast recovery is recommended, but not required, forTCP senders.Slow StartWhen a TCP connection begins, the value of cwnd is typically initialized to asmall value of 1 MSS [RFC 3390], resulting in an initial sending rate of roughlyMSS/RTT. For example, if MSS = 500 bytes and RTT = 200 msec, the resultinginitial sending rate is only about 20 kbps. Since the available bandwidth to theTCP sender may be much larger than MSS/RTT, the TCP sender would like tofind the amount of available bandwidth quickly. Thus, in the slow-start state, thevalue of cwnd begins at 1 MSS and increases by 1 MSS every time a transmittedsegment is first acknowledged. In the example of Figure 3.51, TCP sends the firstsegment into the network and waits for an acknowledgment. When this acknowl-edgment arrives, the TCP sender increases the congestion window by one MSSand sends out two maximum-sized segments. These segments are then acknowl-edged, with the sender increasing the congestion window by 1 MSS for each ofthe acknowledged segments, giving a congestion window of 4 MSS, and so on.This process results in a doubling of the sending rate every RTT. Thus, the TCPsend rate starts slow but grows exponentially during the slow start phase.But when should this exponential growth end? Slow start provides severalanswers to this question. First, if there is a loss event (i.e., congestion) indicatedby a timeout, the TCP sender sets the value of cwnd to 1 and begins the slowstart process anew. It also sets the value of a second state variable, ssthresh(shorthand for slow start threshold) to cwnd/2half of the value of the con-gestion window value when congestion was detected. The second way in whichslow start may end is directly tied to the value of ssthresh. Since ssthreshis half the value of cwnd when congestion was last detected, it might be a bitreckless to keep doubling cwnd when it reaches or surpasses the value ofssthresh. Thus, when the value of cwnd equals ssthresh, slow start endsand TCP transitions into congestion avoidance mode. As well see, TCP increases3.7 TCP CONGESTION CONTROL 273TCP SPLITTING: OPTIMIZING THE PERFORMANCE OF CLOUD SERVICESFor cloud services such as search, e-mail, and social networks, it is desirable to provide ahigh-level of responsiveness, ideally giving users the illusion that the services are runningwithin their own end systems (including their smartphones). This can be a major challenge,as users are often located far away from the data centers that are responsible for servingthe dynamic content associated with the cloud services. Indeed, if the end system is farfrom a data center, then the RTT will be large, potentially leading to poor response timeperformance due to TCP slow start. As a case study, consider the delay in receiving a response for a search query.Typically, the server requires three TCP windows during slow start to deliver theresponse [Pathak 2010]. Thus the time from when an end system initiates a TCP con-nection until the time when it receives the last packet of the response is roughly 4 RTT(one RTT to set up the TCP connection plus three RTTs for the three windows of data)plus the processing time in the data center. These RTT delays can lead to a noticeabledelay in returning search results for a significant fraction of queries. Moreover, therecan be significant packet loss in access networks, leading to TCP retransmissions andeven larger delays.One way to mitigate this problem and improve user-perceived performance is to (1)deploy front-end servers closer to the users, and (2) utilize TCP splitting by breaking theTCP connection at the front-end server. With TCP splitting, the client establishes a TCP con-nection to the nearby front-end, and the front-end maintains a persistent TCP connection tothe data center with a very large TCP congestion window [Tariq 2008, Pathak 2010,Chen 2011]. With this approach, the response time roughly becomes 4 RTTFE RTTBEprocessing time, where RTTFE is the round-trip time between client and front-end server, andRTTBE is the round-trip time between the front-end server and the data center (back-end server).If the front-end server is close to client, then this response time approximately becomesRTT plus processing time, since RTTFE is negligibly small and RTTBE is approximately RTT. Insummary, TCP splitting can reduce the networking delay roughly from 4 RTT to RTT, signifi-cantly improving user-perceived performance, particularly for users who are far from thenearest data center. TCP splitting also helps reduce TCP retransmission delays caused bylosses in access networks. Today, Google and Akamai make extensive use of their CDNservers in access networks (see Section 7.2) to perform TCP splitting for the cloud servicesthey support [Chen 2011].++PRINCIPLES IN PRACTICEcwnd more cautiously when in congestion-avoidance mode. The final way inwhich slow start can end is if three duplicate ACKs are detected, in which caseTCP performs a fast retransmit (see Section 3.5.4) and enters the fast recoverystate, as discussed below. TCPs behavior in slow start is summarized in the FSM274 CHAPTER 3 TRANSPORT LAYERdescription of TCP congestion control in Figure 3.52. The slow-start algorithmtraces it roots to [Jacobson 1988]; an approach similar to slow start was also pro-posed independently in [Jain 1986].Congestion AvoidanceOn entry to the congestion-avoidance state, the value of cwnd is approximately halfits value when congestion was last encounteredcongestion could be just aroundthe corner! Thus, rather than doubling the value of cwnd every RTT, TCP adopts amore conservative approach and increases the value of cwnd by just a single MSSevery RTT [RFC 5681]. This can be accomplished in several ways. A commonapproach is for the TCP sender to increase cwnd by MSS bytes (MSS/cwnd) when-ever a new acknowledgment arrives. For example, if MSS is 1,460 bytes and cwndis 14,600 bytes, then 10 segments are being sent within an RTT. Each arriving ACK(assuming one ACK per segment) increases the congestion window size by 1/10Host A Host Bone segmenttwo segmentsfour segmentsRTTTime TimeFigure 3.51 TCP slow start3.7 TCP CONGESTION CONTROL 275MSS, and thus, the value of the congestion window will have increased by one MSSafter ACKs when all 10 segments have been received.But when should congestion avoidances linear increase (of 1 MSS per RTT)end? TCPs congestion-avoidance algorithm behaves the same when a timeoutoccurs. As in the case of slow start: The value of cwnd is set to 1 MSS, and thevalue of ssthresh is updated to half the value of cwnd when the loss eventoccurred. Recall, however, that a loss event also can be triggered by a triple dupli-cate ACK event. In this case, the network is continuing to deliver segments fromsender to receiver (as indicated by the receipt of duplicate ACKs). So TCPs behav-ior to this type of loss event should be less drastic than with a timeout-indicated loss:TCP halves the value of cwnd (adding in 3 MSS for good measure to account forSlowstartduplicate ACKdupACKcount++duplicate ACKdupACKcount++timeoutssthresh=cwnd/2cwnd=1 MSSdupACKcount=0cwnd=1 MSSssthresh=64 KBdupACKcount=0timeoutssthresh=cwnd/2cwnd=1dupACKcount=0timeoutssthresh=cwnd/2cwnd=1 MSSdupACKcount=0cwnd ssthreshCongestionavoidanceFastrecoverynew ACKcwnd=cwnd+MSS (MSS/cwnd)dupACKcount=0transmit new segment(s), as allowednew ACKcwnd=cwnd+MSSdupACKcount=0transmit new segment(s), as allowedretransmit missing segmentretransmit missing segment dupACKcount==3ssthresh=cwnd/2cwnd=ssthresh+3MSSretransmit missing segmentduplicate ACKcwnd=cwnd+MSStransmit new segment(s), as alloweddupACKcount==3ssthresh=cwnd/2cwnd=ssthresh+3MSSretransmit missing segmentretransmit missing segmentnew ACKcwnd=ssthreshdupACKcount=0Figure 3.52 FSM description of TCP congestion control276 CHAPTER 3 TRANSPORT LAYERthe triple duplicate ACKs received) and records the value of ssthresh to be halfthe value of cwnd when the triple duplicate ACKs were received. The fast-recoverystate is then entered. Fast RecoveryIn fast recovery, the value of cwnd is increased by 1 MSS for every duplicate ACKreceived for the missing segment that caused TCP to enter the fast-recovery state.Eventually, when an ACK arrives for the missing segment, TCP enters thecongestion-avoidance state after deflating cwnd. If a timeout event occurs, fastrecovery transitions to the slow-start state after performing the same actions as inslow start and congestion avoidance: The value of cwnd is set to 1 MSS, and thevalue of ssthresh is set to half the value of cwnd when the loss event occurred.Fast recovery is a recommended, but not required, component of TCP [RFC5681]. It is interesting that an early version of TCP, known as TCP Tahoe, uncondi-tionally cut its congestion window to 1 MSS and entered the slow-start phase aftereither a timeout-indicated or triple-duplicate-ACK-indicated loss event. The newerversion of TCP, TCP Reno, incorporated fast recovery. Figure 3.53 illustrates the evolution of TCPs congestion window for both Renoand Tahoe. In this figure, the threshold is initially equal to 8 MSS. For the first eighttransmission rounds, Tahoe and Reno take identical actions. The congestion windowclimbs exponentially fast during slow start and hits the threshold at the fourth roundof transmission. The congestion window then climbs linearly until a triple duplicate-ACK event occurs, just after transmission round 8. Note that the congestion windowis 12 MSS when this loss event occurs. The value of ssthresh is then set to 010 2 3 4 5 6 7 8Transmission roundTCP TahoessthreshssthreshCongestion window(in segments)9 10 11 12 13 14 15246810121416TCP RenoFigure 3.53 Evolution of TCPs congestion window (Tahoe and Reno)VideoNoteExamining the behavior of TCP3.7 TCP CONGESTION CONTROL 2770.5 cwnd = 6 MSS. Under TCP Reno, the congestion window is set to cwnd =6 MSS and then grows linearly. Under TCP Tahoe, the congestion window is set to1 MSS and grows exponentially until it reaches the value of ssthresh, at whichpoint it grows linearly. Figure 3.52 presents the complete FSM description of TCPs congestion-control algorithmsslow start, congestion avoidance, and fast recovery. The figurealso indicates where transmission of new segments or retransmitted segments canoccur. Although it is important to distinguish between TCP error control/retransmis-sion and TCP congestion control, its also important to appreciate how these twoaspects of TCP are inextricably linked.TCP Congestion Control: RetrospectiveHaving delved into the details of slow start, congestion avoidance, and fast recov-ery, its worthwhile to now step back and view the forest from the trees. Ignoring theinitial slow-start period when a connection begins and assuming that losses are indi-cated by triple duplicate ACKs rather than timeouts, TCPs congestion control con-sists of linear (additive) increase in cwnd of 1 MSS per RTT and then a halving(multiplicative decrease) of cwnd on a triple duplicate-ACK event. For this reason,TCP congestion control is often referred to as an additive-increase, multiplicative-decrease (AIMD) form of congestion control. AIMD congestion control gives riseto the saw tooth behavior shown in Figure 3.54, which also nicely illustrates ourearlier intuition of TCP probing for bandwidthTCP linearly increases its con-gestion window size (and hence its transmission rate) until a triple duplicate-ACKevent occurs. It then decreases its congestion window size by a factor of two butthen again begins increasing it linearly, probing to see if there is additional availablebandwidth.24 K16 K8 KTimeCongestion windowFigure 3.54 Additive-increase, multiplicative-decrease congestion control278 CHAPTER 3 TRANSPORT LAYERAs noted previously, many TCP implementations use the Reno algorithm [Padhye2001]. Many variations of the Reno algorithm have been proposed [RFC 3782; RFC2018]. The TCP Vegas algorithm [Brakmo 1995; Ahn 1995] attempts to avoid conges-tion while maintaining good throughput. The basic idea of Vegas is to (1) detect con-gestion in the routers between source and destination before packet loss occurs, and (2)lower the rate linearly when this imminent packet loss is detected. Imminent packet lossis predicted by observing the RTT. The longer the RTT of the packets, the greater thecongestion in the routers. Linux supports a number of congestion-control algorithms(including TCP Reno and TCP Vegas) and allows a system administrator to configurewhich version of TCP will be used. The default version of TCP in Linux version 2.6.18was set to CUBIC [Ha 2008], a version of TCP developed for high-bandwidth applica-tions. For a recent survey of the many flavors of TCP, see [Afanasyev 2010].TCPs AIMD algorithm was developed based on a tremendous amount of engi-neering insight and experimentation with congestion control in operational net-works. Ten years after TCPs development, theoretical analyses showed that TCPscongestion-control algorithm serves as a distributed asynchronous-optimizationalgorithm that results in several important aspects of user and network performancebeing simultaneously optimized [Kelly 1998]. A rich theory of congestion controlhas since been developed [Srikant 2004].Macroscopic Description of TCP ThroughputGiven the saw-toothed behavior of TCP, its natural to consider what the averagethroughput (that is, the average rate) of a long-lived TCP connection might be. In thisanalysis well ignore the slow-start phases that occur after timeout events. (Thesephases are typically very short, since the sender grows out of the phase exponentiallyfast.) During a particular round-trip interval, the rate at which TCP sends data is afunction of the congestion window and the current RTT. When the window size is wbytes and the current round-trip time is RTT seconds, then TCPs transmission rate isroughly w/RTT. TCP then probes for additional bandwidth by increasing w by 1 MSSeach RTT until a loss event occurs. Denote by W the value of w when a loss eventoccurs. Assuming that RTT and W are approximately constant over the duration ofthe connection, the TCP transmission rate ranges from W/(2 RTT) to W/RTT.These assumptions lead to a highly simplified macroscopic model for thesteady-state behavior of TCP. The network drops a packet from the connection whenthe rate increases to W/RTT; the rate is then cut in half and then increases byMSS/RTT every RTT until it again reaches W/RTT. This process repeats itself overand over again. Because TCPs throughput (that is, rate) increases linearly betweenthe two extreme values, we haveaverage throughput of a connection =0.75 WRTT3.7 TCP CONGESTION CONTROL 279Using this highly idealized model for the steady-state dynamics of TCP, we canalso derive an interesting expression that relates a connections loss rate to its avail-able bandwidth [Mahdavi 1997]. This derivation is outlined in the homework prob-lems. A more sophisticated model that has been found empirically to agree withmeasured data is [Padhye 2000].TCP Over High-Bandwidth PathsIt is important to realize that TCP congestion control has evolved over the years andindeed continues to evolve. For a summary ofcurrent TCP variants and discussionof TCP evolution, see [Floyd 2001, RFC 5681,Afanasyev 2010]. What was goodfor the Internet when the bulk of the TCP connections carried SMTP, FTP, and Tel-net traffic is not necessarily good for todays HTTP-dominated Internet or for afuture Internet with services that are still undreamed of.The need for continued evolution of TCP can be illustrated by considering thehigh-speed TCP connections that are needed for grid- and cloud-computing applica-tions. For example, consider a TCP connection with 1,500-byte segments and a 100ms RTT, and suppose we want to send data through this connection at 10 Gbps.Following [RFC 3649], we note that using the TCP throughput formula above, inorder to achieve a 10 Gbps throughput, the average congestion window size wouldneed to be 83,333 segments. Thats a lot of segments, leading us to be rather con-cerned that one of these 83,333 in-flight segments might be lost. What would happenin the case of a loss? Or, put another way, what fraction of the transmitted segmentscould be lost that would allow the TCP congestion-control algorithm specified in Fig-ure 3.52 still to achieve the desired 10 Gbps rate? In the homework questions for thischapter, you are led through the derivation of a formula relating the throughput of aTCP connection as a function of the loss rate (L), the round-trip time (RTT), and themaximum segment size (MSS):Using this formula, we can see that in order to achieve a throughput of 10 Gbps,todays TCP congestion-control algorithm can only tolerate a segment loss probabil-ity of 2 1010 (or equivalently, one loss event for every 5,000,000,000 segments)a very low rate. This observation has led a number of researchers to investigate newversions of TCP that are specifically designed for such high-speed environments;see [Jin 2004; RFC 3649; Kelly 2003; Ha 2008] for discussions of these efforts.3.7.1 FairnessConsider K TCP connections, each with a different end-to-end path, but all passingthrough a bottleneck link with transmission rate R bps. (By bottleneck link, we meanaverage throughput of a connection =1.22 MSSRTT 2L280 CHAPTER 3 TRANSPORT LAYERthat for each connection, all the other links along the connections path are not con-gested and have abundant transmission capacity as compared with the transmissioncapacity of the bottleneck link.) Suppose each connection is transferring a large fileand there is no UDP traffic passing through the bottleneck link. A congestion-con-trol mechanism is said to be fair if the average transmission rate of each connectionis approximately R/K; that is, each connection gets an equal share of the link band-width.Is TCPs AIMD algorithm fair, particularly given that different TCP connec-tions may start at different times and thus may have different window sizes at agiven point in time? [Chiu 1989] provides an elegant and intuitive explanation ofwhy TCP congestion control converges to provide an equal share of a bottlenecklinks bandwidth among competing TCP connections.Lets consider the simple case of two TCP connections sharing a single linkwith transmission rate R, as shown in Figure 3.55. Assume that the two connectionshave the same MSS and RTT (so that if they have the same congestion window size,then they have the same throughput), that they have a large amount of data to send,and that no other TCP connections or UDP datagrams traverse this shared link. Also,ignore the slow-start phase of TCP and assume the TCP connections are operatingin CA mode (AIMD) at all times.Figure 3.56 plots the throughput realized by the two TCP connections. If TCP isto share the link bandwidth equally between the two connections, then the realizedthroughput should fall along the 45-degree arrow (equal bandwidth share) emanat-ing from the origin. Ideally, the sum of the two throughputs should equal R. (Cer-tainly, each connection receiving an equal, but zero, share of the link capacity is nota desirable situation!) So the goal should be to have the achieved throughputs fallsomewhere near the intersection of the equal bandwidth share line and the full band-width utilization line in Figure 3.56.Suppose that the TCP window sizes are such that at a given point in time, con-nections 1 and 2 realize throughputs indicated by point A in Figure 3.56. Becausethe amount of link bandwidth jointly consumed by the two connections is less thanTCP connection 2TCP connection 1Bottleneckrouter capacity RFigure 3.55 Two TCP connections sharing a single bottleneck link3.7 TCP CONGESTION CONTROL 281R, no loss will occur, and both connections will increase their window by 1 MSSper RTT as a result of TCPs congestion-avoidance algorithm. Thus, the jointthroughput of the two connections proceeds along a 45-degree line (equal increasefor both connections) starting from point A. Eventually, the link bandwidth jointlyconsumed by the two connections will be greater than R, and eventually packet losswill occur. Suppose that connections 1 and 2 experience packet loss when theyrealize throughputs indicated by point B. Connections 1 and 2 then decrease theirwindows by a factor of two. The resulting throughputs realized are thus at point C,halfway along a vector starting at B and ending at the origin. Because the jointbandwidth use is less than R at point C, the two connections again increase theirthroughputs along a 45-degree line starting from C. Eventually, loss will againoccur, for example, at point D, and the two connections again decrease their win-dow sizes by a factor of two, and so on. You should convince yourself that thebandwidth realized by the two connections eventually fluctuates along the equalbandwidth share line. You should also convince yourself that the two connectionswill converge to this behavior regardless of where they are in the two-dimensionalspace! Although a number of idealized assumptions lie behind this scenario, it stillprovides an intuitive feel for why TCP results in an equal sharing of bandwidthamong connections.In our idealized scenario, we assumed that only TCP connections traverse the bottleneck link, that the connections have the same RTT value, and that only aRREqualbandwidthshareConnection 1 throughputConnection 2 throughputDBCAFull bandwidthutilization lineFigure 3.56 Throughput realized by TCP connections 1 and 2282 CHAPTER 3 TRANSPORT LAYERsingle TCP connection is associated with a host-destination pair. In practice, theseconditions are typically not met, and client-server applications can thus obtain veryunequal portions of link bandwidth. In particular, it has been shown that when mul-tiple connections share a common bottleneck, those sessions with a smaller RTT areable to grab the available bandwidth at that link more quickly as it becomes free(that is, open their congestion windows faster) and thus will enjoy higher through-put than those connections with larger RTTs [Lakshman 1997].Fairness and UDPWe have just seen how TCP congestion control regulates an applications transmis-sion rate via the congestion window mechanism. Many multimedia applications,such as Internet phone and video conferencing, often do not run over TCP for thisvery reasonthey do not want their transmission rate throttled, even if the networkis very congested. Instead, these applications prefer to run over UDP, which doesnot have built-in congestion control. When running over UDP, applications canpump their audio and video into the network at a constant rate and occasionally losepackets, rather than reduce their rates to fair levels at times of congestion and notlose any packets. From the perspective of TCP, the multimedia applications runningover UDP are not being fairthey do not cooperate with the other connections noradjust their transmission rates appropriately. Because TCP congestion control willdecrease its transmission rate in the face of increasing congestion (loss), while UDPsources need not, it is possible for UDP sources to crowd out TCP traffic. An area ofresearch today is thus the development of congestion-control mechanisms for theInternet that prevent UDP traffic from bringing the Internets throughput to a grind-ing halt [Floyd 1999; Floyd 2000; Kohler 2006].Fairness and Parallel TCP ConnectionsBut even if we could force UDP traffic to behave fairly, the fairness problem wouldstill not be completely solved. This is because there is nothing to stop a TCP-basedapplication from using multiple parallel connections. For example, Web browsersoften use multiple parallel TCP connections to transfer the multiple objects withina Web page. (The exact number of multiple connections is configurable in mostbrowsers.) When an application uses multiple parallel connections, it gets a largerfraction of the bandwidth in a congested link. As an example, consider a link of rateR supporting nine ongoing client-server applications, with each of the applicationsusing one TCP connection. If a new application comes along and also uses oneTCP connection, then each application gets approximately the same transmissionrate of R/10. But if this new application instead uses 11 parallel TCP connections,then the new application gets an unfair allocation of more than R/2. Because Web traffic is so pervasive in the Internet, multiple parallel connections are notuncommon.3.8 SUMMARY 2833.8 SummaryWe began this chapter by studying the services that a transport-layer protocol canprovide to network applications. At one extreme, the transport-layer protocol can bevery simple and offer a no-frills service to applications, providing only a multiplex-ing/demultiplexing function for communicating processes. The Internets UDP pro-tocol is an example of such a no-frills transport-layer protocol. At the other extreme,a transport-layer protocol can provide a variety of guarantees to applications, suchas reliable delivery of data, delay guarantees, and bandwidth guarantees. Neverthe-less, the services that a transport protocol can provide are often constrained by theservice model of the underlying network-layer protocol. If the network-layer proto-col cannot provide delay or bandwidth guarantees to transport-layer segments, thenthe transport-layer protocol cannot provide delay or bandwidth guarantees for themessages sent between processes.We learned in Section 3.4 that a transport-layer protocol can provide reliabledata transfer even if the underlying network layer is unreliable. We saw that provid-ing reliable data transfer has many subtle points, but that the task can be accom-plished by carefully combining acknowledgments, timers, retransmissions, andsequence numbers.Although we covered reliable data transfer in this chapter, we should keep inmind that reliable data transfer can be provided by link-, network-, transport-, orapplication-layer protocols. Any of the upper four layers of the protocol stack canimplement acknowledgments, timers, retransmissions, and sequence numbers andprovide reliable data transfer to the layer above. In fact, over the years, engineersand computer scientists have independently designed and implemented link-, net-work-, transport-, and application-layer protocols that provide reliable data transfer(although many of these protocols have quietly disappeared).In Section 3.5, we took a close look at TCP, the Internets connection-orientedand reliable transport-layer protocol. We learned that TCP is complex, involvingconnection management, flow control, and round-trip time estimation, as well asreliable data transfer. In fact, TCP is actually more complex than our descriptionwe intentionally did not discuss a variety of TCP patches, fixes, and improvementsthat are widely implemented in various versions of TCP. All of this complexity,however, is hidden from the network application. If a client on one host wants tosend data reliably to a server on another host, it simply opens a TCP socket to theserver and pumps data into that socket. The client-server application is blissfullyunaware of TCPs complexity.In Section 3.6, we examined congestion control from a broad perspective, and inSection 3.7, we showed how TCP implements congestion control. We learned thatcongestion control is imperative for the well-being of the network. Without conges-tion control, a network can easily become gridlocked, with little or no data being trans-ported end-to-end. In Section 3.7 we learned that TCP implements an end-to-end284 CHAPTER 3 TRANSPORT LAYERcongestion-control mechanism that additively increases its transmission rate when theTCP connections path is judged to be congestion-free, and multiplicatively decreasesits transmission rate when loss occurs. This mechanism also strives to give each TCPconnection passing through a congested link an equal share of the link bandwidth. Wealso examined in some depth the impact of TCP connection establishment and slowstart on latency. We observed that in many important scenarios, connection establish-ment and slow start significantly contribute to end-to-end delay. We emphasize oncemore that while TCP congestion control has evolved over the years, it remains an areaof intensive research and will likely continue to evolve in the upcoming years.Our discussion of specific Internet transport protocols in this chapter hasfocused on UDP and TCPthe two work horses of the Internet transport layer.However, two decades of experience with these two protocols has identifiedcircumstances in which neither is ideally suited. Researchers have thus beenbusy developing additional transport-layer protocols, several of which are nowIETF proposed standards.The Datagram Congestion Control Protocol (DCCP) [RFC 4340] provides a low-overhead, message-oriented, UDP-like unreliable service, but with an application-selected form of congestion control that is compatible with TCP. If reliable orsemi-reliable data transfer is needed by an application, then this would be performedwithin the application itself, perhaps using the mechanisms we have studied in Section3.4. DCCP is envisioned for use in applications such as streaming media (see Chapter 7)that can exploit the tradeoff between timeliness and reliability of data delivery, but thatwant to be responsive to network congestion.The Stream Control Transmission Protocol (SCTP) [RFC 4960, RFC 3286] is areliable, message-oriented protocol that allows several different application-levelstreams to be multiplexed through a single SCTP connection (an approach known asmulti-streaming). From a reliability standpoint, the different streams within the con-nection are handled separately, so that packet loss in one stream does not affect thedelivery of data in other streams. SCTP also allows data to be transferred over two out-going paths when a host is connected to two or more networks, optional delivery of out-of-order data, and a number of other features. SCTPs flow- and congestion-controlalgorithms are essentially the same as in TCP.The TCP-Friendly Rate Control (TFRC) protocol [RFC 5348] is a congestion-control protocol rather than a full-fledged transport-layer protocol. It specifies acongestion-control mechanism that could be used in anther transport protocol such asDCCP (indeed one of the two application-selectable protocols available in DCCP isTFRC). The goal of TFRC is to smooth out the saw tooth behavior (see Figure 3.54)in TCP congestion control, while maintaining a long-term sending rate that is reason-ably close to that of TCP. With a smoother sending rate than TCP, TFRC is well-suitedfor multimedia applications such as IP telephony or streaming media where such asmooth rate is important. TFRC is an equation-based protocol that uses the measuredpacket loss rate as input to an equation [Padhye 2000] that estimates what TCPsthroughput would be if a TCP session experiences that loss rate. This rate is then takenas TFRCs target sending rate.HOMEWORK PROBLEMS AND QUESTIONS 285Only the future will tell whether DCCP, SCTP, or TFRC will see widespreaddeployment. While these protocols clearly provide enhanced capabilities over TCP andUDP, TCP and UDP have proven themselves good enough over the years. Whetherbetter wins out over good enough will depend on a complex mix of technical,social, and business considerations.In Chapter 1, we said that a computer network can be partitioned into thenetwork edge and the network core. The network edge covers everything thathappens in the end systems. Having now covered the application layer and thetransport layer, our discussion of the network edge is complete. It is time toexplore the network core! This journey begins in the next chapter, where wellstudy the network layer, and continues into Chapter 5, where well study the link layer.Homework Problems and QuestionsChapter 3 Review QuestionsSECTIONS 3.13.3R1. Suppose the network layer provides the following service. The network layerin the source host accepts a segment of maximum size 1,200 bytes and a des-tination host address from the transport layer. The network layer then guaran-tees to deliver the segment to the transport layer at the destination host.Suppose many network application processes can be running at thedestination host.a. Design the simplest possible transport-layer protocol that will get applica-tion data to the desired process at the destination host. Assume the operat-ing system in the destination host has assigned a 4-byte port number toeach running application process.b. Modify this protocol so that it provides a return address to the destina-tion process.c. In your protocols, does the transport layer have to do anything in thecore of the computer network?R2. Consider a planet where everyone belongs to a family of six, every familylives in its own house, each house has a unique address, and each person in agiven house has a unique name. Suppose this planet has a mail service thatdelivers letters from source house to destination house. The mail servicerequires that (1) the letter be in an envelope, and that (2) the address of thedestination house (and nothing more) be clearly written on the envelope. Sup-pose each family has a delegate family member who collects and distributesletters for the other family members. The letters do not necessarily provideany indication of the recipients of the letters.286 CHAPTER 3 TRANSPORT LAYERa. Using the solution to Problem R1 above as inspiration, describe a protocolthat the delegates can use to deliver letters from a sending family memberto a receiving family member.b. In your protocol, does the mail service ever have to open the envelope andexamine the letter in order to provide its service?R3. Consider a TCP connection between Host A and Host B. Suppose that theTCP segments traveling from Host A to Host B have source port number xand destination port number y. What are the source and destination port num-bers for the segments traveling from Host B to Host A?R4. Describe why an application developer might choose to run an applicationover UDP rather than TCP.R5. Why is it that voice and video traffic is often sent over TCP rather than UDPin todays Internet? (Hint: The answer we are looking for has nothing to dowith TCPs congestion-control mechanism.)R6. Is it possible for an application to enjoy reliable data transfer even when theapplication runs over UDP? If so, how?R7. Suppose a process in Host C has a UDP socket with port number 6789. Sup-pose both Host A and Host B each send a UDP segment to Host C with desti-nation port number 6789. Will both of these segments be directed to the samesocket at Host C? If so, how will the process at Host C know that these twosegments originated from two different hosts?R8. Suppose that a Web server runs in Host C on port 80. Suppose this Webserver uses persistent connections, and is currently receiving requests fromtwo different Hosts, A and B. Are all of the requests being sent through thesame socket at Host C? If they are being passed through different sockets, doboth of the sockets have port 80? Discuss and explain.SECTION 3.4R9. In our rdt protocols, why did we need to introduce sequence numbers?R10. In our rdt protocols, why did we need to introduce timers?R11. Suppose that the roundtrip delay between sender and receiver is constant andknown to the sender. Would a timer still be necessary in protocol rdt 3.0,assuming that packets can be lost? Explain.R12. Visit the Go-Back-N Java applet at the companion Web site.a. Have the source send five packets, and then pause the animation beforeany of the five packets reach the destination. Then kill the first packet andresume the animation. Describe what happens.b. Repeat the experiment, but now let the first packet reach the destinationand kill the first acknowledgment. Describe again what happens.c. Finally, try sending six packets. What happens?HOMEWORK PROBLEMS AND QUESTIONS 287R13. Repeat R12, but now with the Selective Repeat Java applet. How are Selec-tive Repeat and Go-Back-N different?SECTION 3.5R14. True or false?a. Host A is sending Host B a large file over a TCP connection. AssumeHost B has no data to send Host A. Host B will not send acknowledg-ments to Host A because Host B cannot piggyback the acknowledgmentson data.b. The size of the TCP rwnd never changes throughout the duration of theconnection.c. Suppose Host A is sending Host B a large file over a TCP connection. Thenumber of unacknowledged bytes that A sends cannot exceed the size ofthe receive buffer.d. Suppose Host A is sending a large file to Host B over a TCP connection. Ifthe sequence number for a segment of this connection is m, then thesequence number for the subsequent segment will necessarily be m + 1.e. The TCP segment has a field in its header for rwnd.f. Suppose that the last SampleRTT in a TCP connection is equal to 1 sec.The current value of TimeoutInterval for the connection will neces-sarily be 1 sec.g. Suppose Host A sends one segment with sequence number 38 and 4 bytesof data over a TCP connection to Host B. In this same segment theacknowledgment number is necessarily 42.R15. Suppose Host A sends two TCP segments back to back to Host B over a TCPconnection. The first segment has sequence number 90; the second hassequence number 110.a. How much data is in the first segment?b. Suppose that the first segment is lost but the second segment arrives at B.In the acknowledgment that Host B sends to Host A, what will be theacknowledgment number?R16. Consider the Telnet example discussed in Section 3.5. A few seconds after theuser types the letter C, the user types the letter R.After typing the letterR, how many segments are sent, and what is put in the sequence numberand acknowledgment fields of the segments?SECTION 3.7R17. Suppose two TCP connections are present over some bottleneck link of rate Rbps. Both connections have a huge file to send (in the same direction over the288 CHAPTER 3 TRANSPORT LAYERbottleneck link). The transmissions of the files start at the same time. Whattransmission rate would TCP like to give to each of the connections?R18. True or false? Consider congestion control in TCP. When the timer expires atthe sender, the value of ssthresh is set to one half of its previous value.R19. In the discussion of TCP splitting in the sidebar in Section 7.2, it wasclaimed that the response time with TCP splitting is approximatelyJustify this claim.ProblemsP1. Suppose Client A initiates a Telnet session with Server S. At about the sametime, Client B also initiates a Telnet session with Server S. Provide possiblesource and destination port numbers fora. The segments sent from A to S.b. The segments sent from B to S.c. The segments sent from S to A.d. The segments sent from S to B.e. If A and B are different hosts, is it possible that the source port number inthe segments from A to S is the same as that from B to S?f. How about if they are the same host?P2. Consider Figure 3.5. What are the source and destination port values in the seg-ments flowing from the server back to the clients processes? What are the IPaddresses in the network-layer datagrams carrying the transport-layer segments?P3. UDP and TCP use 1s complement for their checksums. Suppose you have thefollowing three 8-bit bytes: 01010011, 01100110, 01110100. What is the 1scomplement of the sum of these 8-bit bytes? (Note that although UDP andTCP use 16-bit words in computing the checksum, for this problem you arebeing asked to consider 8-bit sums.) Show all work. Why is it that UDP takesthe 1s complement of the sum; that is, why not just use the sum? With the 1scomplement scheme, how does the receiver detect errors? Is it possible that a1-bit error will go undetected? How about a 2-bit error?P4. a. Suppose you have the following 2 bytes: 01011100 and 01100101. What isthe 1s complement of the sum of these 2 bytes?b. Suppose you have the following 2 bytes: 11011010 and 01100101. What isthe 1s complement of the sum of these 2 bytes?c. For the bytes in part (a), give an example where one bit is flipped in eachof the 2 bytes and yet the 1s complement doesnt change.4 RTTFE + RTTBE + processing time.PROBLEMS 289P5. Suppose that the UDP receiver computes the Internet checksum for the receivedUDP segment and finds that it matches the value carried in the checksum field.Can the receiver be absolutely certain that no bit errors have occurred? Explain.P6. Consider our motivation for correcting protocol rdt2.1. Show that thereceiver, shown in Figure 3.57, when operating with the sender shown in Fig-ure 3.11, can lead the sender and receiver to enter into a deadlock state, whereeach is waiting for an event that will never occur.P7. In protocol rdt3.0, the ACK packets flowing from the receiver to thesender do not have sequence numbers (although they do have an ACK fieldthat contains the sequence number of the packet they are acknowledging).Why is it that our ACK packets do not require sequence numbers?P8. Draw the FSM for the receiver side of protocol rdt3.0.P9. Give a trace of the operation of protocol rdt3.0 when data packets andacknowledgment packets are garbled. Your trace should be similar to thatused in Figure 3.16.P10. Consider a channel that can lose packets but has a maximum delay that isknown. Modify protocol rdt2.1 to include sender timeout and retransmit.Informally argue why your protocol can communicate correctly over thischannel.Wait for0 frombelowrdt_rcv(rcvpkt) &&(corrupt(rcvpkt)||has_seq0(rcvpkt)))compute chksummake_pkt(sndpkt,NAK,chksum)udt_send(sndpkt)rdt_rcv(rcvpkt) &&(corrupt(rcvpkt)||has_seq1(rcvpkt)))compute chksummake_pkt(sndpkt,NAK,chksum)udt_send(sndpkt) rdt_rvc(rcvpkt) && notcorrupt(rcvpkt)&& has_seq1(rcvpkt)extract(rcvpkt,data)deliver_data(data)compute chksummake_pkt(sendpkt,ACK,chksum)udt_send(sndpkt)rdt_rvc(rcvpkt) && notcorrupt(rcvpkt)&& has_seq0(rcvpkt)extract(rcvpkt,data)deliver_data(data)compute chksummake_pkt(sendpkt,ACK,chksum)udt_send(sndpkt)Wait for1 frombelowFigure 3.57 An incorrect receiver for protocol rdt 2.1290 CHAPTER 3 TRANSPORT LAYERP11. Consider the rdt2.2 receiver in Figure 3.14, and the creation of a new packetin the self-transition (i.e., the transition from the state back to itself) in the Wait-for-0-from-below and the Wait-for-1-from-below states: sndpkt=make_pkt(ACK,0,checksum) and sndpkt=make_pkt(ACK,0,checksum). Would the protocol work correctly if this action were removedfrom the self-transition in the Wait-for-1-from-below state? Justify youranswer. What if this event were removed from the self-transition in the Wait-for-0-from-below state? [Hint: In this latter case, consider what would hap-pen if the first sender-to-receiver packet were corrupted.]P12. The sender side of rdt3.0 simply ignores (that is, takes no action on) all received packets that are either in error or have the wrong value in theacknum field of an acknowledgment packet. Suppose that in such circum-stances, rdt3.0 were simply to retransmit the current data packet. Wouldthe protocol still work? (Hint: Consider what would happen if there wereonly bit errors; there are no packet losses but premature timeouts can occur.Consider how many times the nth packet is sent, in the limit as napproaches infinity.)P13. Consider the rdt 3.0 protocol. Draw a diagram showing that if the net-work connection between the sender and receiver can reorder messages(that is, that two messages propagating in the medium between the senderand receiver can be reordered), then the alternating-bit protocol will notwork correctly (make sure you clearly identify the sense in which it willnot work correctly). Your diagram should have the sender on the left andthe receiver on the right, with the time axis running down the page, show-ing data (D) and acknowledgment (A) message exchange. Make sure youindicate the sequence number associated with any data or acknowledgmentsegment.P14. Consider a reliable data transfer protocol that uses only negative acknowledg-ments. Suppose the sender sends data only infrequently. Would a NAK-onlyprotocol be preferable to a protocol that uses ACKs? Why? Now suppose thesender has a lot of data to send and the end-to-end connection experiencesfew losses. In this second case, would a NAK-only protocol be preferable to aprotocol that uses ACKs? Why?P15. Consider the cross-country example shown in Figure 3.17. How big wouldthe window size have to be for the channel utilization to be greater than 98percent? Suppose that the size of a packet is 1,500 bytes, including bothheader fields and data.P16. Suppose an application uses rdt 3.0 as its transport layer protocol. As thestop-and-wait protocol has very low channel utilization (shown in the cross-country example), the designers of this application let the receiver keep send-ing back a number (more than two) of alternating ACK 0 and ACK 1 even ifPROBLEMS 291the corresponding data have not arrived at the receiver. Would this applica-tion design increase the channel utilization? Why? Are there any potentialproblems with this approach? Explain. P17. Consider two network entities, A and B, which are connected by a perfect bi-directional channel (i.e., any message sent will be received correctly; thechannel will not corrupt, lose, or re-order packets). A and B are to deliverdata messages to each other in an alternating manner: First, A must deliver amessage to B, then B must deliver a message to A, then A must deliver a mes-sage to B and so on. If an entity is in a state where it should not attempt todeliver a message to the other side, and there is an event likerdt_send(data) call from above that attempts to pass data down fortransmission to the other side, this call from above can simply be ignoredwith a call to rdt_unable_to_send(data), which informs the higherlayer that it is currently not able to send data. [Note: This simplifyingassumption is made so you dont have to worry about buffering data.]Draw a FSM specification for this protocol (one FSM for A, and one FSM forB!). Note that you do not have to worry about a reliability mechanism here;the main point of this question is to create a FSM specification that reflectsthe synchronized behavior of the two entities. You should use the followingevents and actions that have the same meaning as protocol rdt1.0 in Figure 3.9: rdt_send(data), packet = make_pkt(data),udt_send(packet), rdt_rcv(packet), extract(packet,data), deliver_data(data). Make sure your protocolreflects the strict alternation of sending between A and B. Also, make sure toindicate the initial states for A and B in your FSM descriptions.P18. In the generic SR protocol that we studied in Section 3.4.4, the sender trans-mits a message as soon as it is available (if it is in the window) without wait-ing for an acknowledgment. Suppose now that we want an SR protocol thatsends messages two at a time. That is, the sender will send a pair of messagesand will send the next pair of messages only when it knows that both mes-sages in the first pair have been received correctly.Suppose that the channel may lose messages but will not corrupt or reordermessages. Design an error-control protocol for the unidirectional reliabletransfer of messages. Give an FSM description of the sender and receiver.Describe the format of the packets sent between sender and receiver, and viceversa. If you use any procedure calls other than those in Section 3.4 (forexample, udt_send(), start_timer(), rdt_rcv(), and so on),clearly state their actions. Give an example (a timeline trace of sender andreceiver) showing how your protocol recovers from a lost packet.P19. Consider a scenario in which Host A wants to simultaneously send packets toHosts B and C. A is connected to B and C via a broadcast channela packet292 CHAPTER 3 TRANSPORT LAYERsent by A is carried by the channel to both B and C. Suppose that the broad-cast channel connecting A, B, and C can independently lose and corruptpackets (and so, for example, a packet sent from A might be correctlyreceived by B, but not by C). Design a stop-and-wait-like error-control proto-col for reliably transferring packets from A to B and C, such that A will notget new data from the upper layer until it knows that both B and C have cor-rectly received the current packet. Give FSM descriptions of A and C. (Hint:The FSM for B should be essentially the same as for C.) Also, give a descrip-tion of the packet format(s) used.P20. Consider a scenario in which Host A and Host B want to send messages toHost C. Hosts A and C are connected by a channel that can lose and corrupt(but not reorder) messages. Hosts B and C are connected by another chan-nel (independent of the channel connecting A and C) with the same proper-ties. The transport layer at Host C should alternate in delivering messagesfrom A and B to the layer above (that is, it should first deliver the data froma packet from A, then the data from a packet from B, and so on). Design astop-and-wait-like error-control protocol for reliably transferring packetsfrom A and B to C, with alternating delivery at C as described above. GiveFSM descriptions of A and C. (Hint: The FSM for B should be essentiallythe same as for A.) Also, give a description of the packet format(s) used.P21. Suppose we have two network entities, A and B. B has a supply of data mes-sages that will be sent to A according to the following conventions. When Agets a request from the layer above to get the next data (D) message from B,A must send a request (R) message to B on the A-to-B channel. Only when Breceives an R message can it send a data (D) message back to A on the B-to-A channel. A should deliver exactly one copy of each D message to the layerabove. R messages can be lost (but not corrupted) in the A-to-B channel; Dmessages, once sent, are always delivered correctly. The delay along bothchannels is unknown and variable.Design (give an FSM description of) a protocol that incorporates the appro-priate mechanisms to compensate for the loss-prone A-to-B channel andimplements message passing to the layer above at entity A, as discussedabove. Use only those mechanisms that are absolutely necessary.P22. Consider the GBN protocol with a sender window size of 4 and a sequencenumber range of 1,024. Suppose that at time t, the next in-order packet that thereceiver is expecting has a sequence number of k. Assume that the mediumdoes not reorder messages. Answer the following questions:a. What are the possible sets of sequence numbers inside the senders win-dow at time t? Justify your answer.b. What are all possible values of the ACK field in all possible messages cur-rently propagating back to the sender at time t? Justify your answer.PROBLEMS 293P23. Consider the GBN and SR protocols. Suppose the sequence number space isof size k. What is the largest allowable sender window that will avoid theoccurrence of problems such as that in Figure 3.27 for each of these protocols?P24. Answer true or false to the following questions and briefly justify youranswer:a. With the SR protocol, it is possible for the sender to receive an ACK for apacket that falls outside of its current window.b. With GBN, it is possible for the sender to receive an ACK for a packet thatfalls outside of its current window.c. The alternating-bit protocol is the same as the SR protocol with a senderand receiver window size of 1.d. The alternating-bit protocol is the same as the GBN protocol with a senderand receiver window size of 1.P25. We have said that an application may choose UDP for a transport protocolbecause UDP offers finer application control (than TCP) of what data is sentin a segment and when.a. Why does an application have more control of what data is sent in a segment?b. Why does an application have more control on when the segment is sent?P26. Consider transferring an enormous file of L bytes from Host A to Host B.Assume an MSS of 536 bytes.a. What is the maximum value of L such that TCP sequence numbers are notexhausted? Recall that the TCP sequence number field has 4 bytes.b. For the L you obtain in (a), find how long it takes to transmit the file.Assume that a total of 66 bytes of transport, network, and data-link headerare added to each segment before the resulting packet is sent out over a155 Mbps link. Ignore flow control and congestion control so A can pumpout the segments back to back and continuously.P27. Host A and B are communicating over a TCP connection, and Host B hasalready received from A all bytes up through byte 126. Suppose Host A thensends two segments to Host B back-to-back. The first and second segmentscontain 80 and 40 bytes of data, respectively. In the first segment, thesequence number is 127, the source port number is 302, and the destinationport number is 80. Host B sends an acknowledgment whenever it receives asegment from Host A.a. In the second segment sent from Host A to B, what are the sequence num-ber, source port number, and destination port number?b. If the first segment arrives before the second segment, in the acknowledg-ment of the first arriving segment, what is the acknowledgment number,the source port number, and the destination port number?294 CHAPTER 3 TRANSPORT LAYERc. If the second segment arrives before the first segment, in the acknowl-edgment of the first arriving segment, what is the acknowledgmentnumber?d. Suppose the two segments sent by A arrive in order at B. The first acknowl-edgment is lost and the second acknowledgment arrives after the first time-out interval. Draw a timing diagram, showing these segments and all othersegments and acknowledgments sent. (Assume there is no additional packetloss.) For each segment in your figure, provide the sequence number andthe number of bytes of data; for each acknowledgment that you add, pro-vide the acknowledgment number.P28. Host A and B are directly connected with a 100 Mbps link. There is one TCPconnection between the two hosts, and Host A is sending to Host B an enor-mous file over this connection. Host A can send its application data into its TCPsocket at a rate as high as 120 Mbps but Host B can read out of its TCP receivebuffer at a maximum rate of 50 Mbps. Describe the effect of TCP flow control.P29. SYN cookies were discussed in Section 3.5.6.a. Why is it necessary for the server to use a special initial sequence numberin the SYNACK?b. Suppose an attacker knows that a target host uses SYN cookies. Can theattacker create half-open or fully open connections by simply sending anACK packet to the target? Why or why not?c. Suppose an attacker collects a large amount of initial sequence numbers sentby the server. Can the attacker cause the server to create many fully openconnections by sending ACKs with those initial sequence numbers? Why?P30. Consider the network shown in Scenario 2 in Section 3.6.1. Suppose bothsending hosts A and B have some fixed timeout values. a. Argue that increasing the size of the finite buffer of the router might possi-bly decrease the throughput (out).b. Now suppose both hosts dynamically adjust their timeout values (likewhat TCP does) based on the buffering delay at the router. Would increas-ing the buffer size help to increase the throughput? Why?P31. Suppose that the five measured SampleRTT values (see Section 3.5.3) are106 ms, 120 ms, 140 ms, 90 ms, and 115 ms. Compute the EstimatedRTTafter each of these SampleRTT values is obtained, using a value of = 0.125and assuming that the value of EstimatedRTT was 100 ms just before thefirst of these five samples were obtained. Compute also the DevRTT aftereach sample is obtained, assuming a value of = 0.25 and assuming thevalue of DevRTT was 5 ms just before the first of these five samples wasobtained. Last, compute the TCP TimeoutInterval after each of thesesamples is obtained.PROBLEMS 295P32. Consider the TCP procedure for estimating RTT. Suppose that = 0.1. LetSampleRTT1 be the most recent sample RTT, let SampleRTT2 be the nextmost recent sample RTT, and so on.a. For a given TCP connection, suppose four acknowledgments have beenreturned with corresponding sample RTTs: SampleRTT4, SampleRTT3,SampleRTT2, and SampleRTT1. Express EstimatedRTT in terms ofthe four sample RTTs.b. Generalize your formula for n sample RTTs.c. For the formula in part (b) let n approach infinity. Comment on why thisaveraging procedure is called an exponential moving average.P33. In Section 3.5.3, we discussed TCPs estimation of RTT. Why do you thinkTCP avoids measuring the SampleRTT for retransmitted segments?P34. What is the relationship between the variable SendBase in Section 3.5.4and the variable LastByteRcvd in Section 3.5.5?P35. What is the relationship between the variable LastByteRcvd in Section3.5.5 and the variable y in Section 3.5.4?P36. In Section 3.5.4, we saw that TCP waits until it has received three duplicate ACKs before performing a fast retransmit. Why do you think theTCP designers chose not to perform a fast retransmit after the first duplicateACK for a segment is received?P37. Compare GBN, SR, and TCP (no delayed ACK). Assume that the timeoutvalues for all three protocols are sufficiently long such that 5 consecutive datasegments and their corresponding ACKs can be received (if not lost in thechannel) by the receiving host (Host B) and the sending host (Host A) respec-tively. Suppose Host A sends 5 data segments to Host B, and the 2nd segment(sent from A) is lost. In the end, all 5 data segments have been correctlyreceived by Host B. a. How many segments has Host A sent in total and how many ACKs hasHost B sent in total? What are their sequence numbers? Answer this ques-tion for all three protocols.b. If the timeout values for all three protocol are much longer than 5 RTT,then which protocol successfully delivers all five data segments in shortesttime interval? P38. In our description of TCP in Figure 3.53, the value of the threshold,ssthresh, is set as ssthresh=cwnd/2 in several places andssthresh value is referred to as being set to half the window size when aloss event occurred. Must the rate at which the sender is sending when theloss event occurred be approximately equal to cwnd segments per RTT?Explain your answer. If your answer is no, can you suggest a differentmanner in which ssthresh should be set?296 CHAPTER 3 TRANSPORT LAYERP39. Consider Figure 3.46(b). If in increases beyond R/2, can out increasebeyond R/3? Explain. Now consider Figure 3.46(c). If in increasesbeyond R/2, can out increase beyond R/4 under the assumption that apacket will be forwarded twice on average from the router to the receiver?Explain.P40. Consider Figure 3.58. Assuming TCP Reno is the protocol experiencing thebehavior shown above, answer the following questions. In all cases, youshould provide a short discussion justifying your answer.a. Identify the intervals of time when TCP slow start is operating.b. Identify the intervals of time when TCP congestion avoidance is operating.c. After the 16th transmission round, is segment loss detected by a tripleduplicate ACK or by a timeout?d. After the 22nd transmission round, is segment loss detected by a tripleduplicate ACK or by a timeout?e. What is the initial value of ssthresh at the first transmission round?f. What is the value of ssthresh at the 18th transmission round?g. What is the value of ssthresh at the 24th transmission round?h. During what transmission round is the 70th segment sent?i. Assuming a packet loss is detected after the 26th round by the receipt of atriple duplicate ACK, what will be the values of the congestion windowsize and of ssthresh?00 2 4 6 8 10 12Transmission round14 16 18 20 22 24 26510152025Congestion window size (segments)30354045Figure 3.58 TCP window size as a function of timeVideoNoteExamining the behavior of TCPPROBLEMS 297j. Suppose TCP Tahoe is used (instead of TCP Reno), and assume that tripleduplicate ACKs are received at the 16th round. What are the ssthreshand the congestion window size at the 19th round?k. Again suppose TCP Tahoe is used, and there is a timeout event at 22ndround. How many packets have been sent out from 17th round till 22ndround, inclusive?P41. Refer to Figure 3.56, which illustrates the convergence of TCPs AIMD algorithm. Suppose that instead of a multiplicative decrease, TCP decreasedthe window size by a constant amount. Would the resulting AIAD algorithmconverge to an equal share algorithm? Justify your answer using a diagramsimilar to Figure 3.56.P42. In Section 3.5.4, we discussed the doubling of the timeout interval after atimeout event. This mechanism is a form of congestion control. Why doesTCP need a window-based congestion-control mechanism (as studied inSection 3.7) in addition to this doubling-timeout-interval mechanism?P43. Host A is sending an enormous file to Host B over a TCP connection. Over this connection there is never any packet loss and the timers neverexpire. Denote the transmission rate of the link connecting Host A to theInternet by R bps. Suppose that the process in Host A is capable of sendingdata into its TCP socket at a rate S bps, where S = 10 R. Further suppose that the TCP receive buffer is large enough to hold the entire file, and thesend buffer can hold only one percent of the file. What would prevent theprocess in Host A from continuously passing data to its TCP socket at rate Sbps? TCP flow control? TCP congestion control? Or something else?Elaborate.P44. Consider sending a large file from a host to another over a TCP connectionthat has no loss.a. Suppose TCP uses AIMD for its congestion control without slow start.Assuming cwnd increases by 1 MSS every time a batch of ACKs is receivedand assuming approximately constant round-trip times, how long does it takefor cwnd increase from 6 MSS to 12 MSS (assuming no loss events)?b. What is the average throughout (in terms of MSS and RTT) for this con-nection up through time = 6 RTT?P45. Recall the macroscopic description of TCP throughput. In the period of timefrom when the connections rate varies from W/(2 RTT) to W/RTT, only onepacket is lost (at the very end of the period).a. Show that the loss rate (fraction of packets lost) is equal toL = loss rate =138W2 +34W298 CHAPTER 3 TRANSPORT LAYERb. Use the result above to show that if a connection has loss rate L, then itsaverage rate is approximately given byP46. Consider that only a single TCP (Reno) connection uses one 10Mbps linkwhich does not buffer any data. Suppose that this link is the only congestedlink between the sending and receiving hosts. Assume that the TCP senderhas a huge file to send to the receiver, and the receivers receive buffer ismuch larger than the congestion window. We also make the followingassumptions: each TCP segment size is 1,500 bytes; the two-way propagationdelay of this connection is 150 msec; and this TCP connection is always incongestion avoidance phase, that is, ignore slow start. a. What is the maximum window size (in segments) that this TCP connectioncan achieve?b. What is the average window size (in segments) and average throughput (inbps) of this TCP connection?c. How long would it take for this TCP connection to reach its maximumwindow again after recovering from a packet loss?P47. Consider the scenario described in the previous problem. Suppose that the10Mbps link can buffer a finite number of segments. Argue that in order forthe link to always be busy sending data, we would like to choose a buffer sizethat is at least the product of the link speed C and the two-way propagationdelay between the sender and the receiver.P48. Repeat Problem 43, but replacing the 10 Mbps link with a 10 Gbps link. Notethat in your answer to part c, you will realize that it takes a very long time forthe congestion window size to reach its maximum window size after recover-ing from a packet loss. Sketch a solution to solve this problem.P49. Let T (measured by RTT) denote the time interval that a TCP connectiontakes to increase its congestion window size from W/2 to W, where W is themaximum congestion window size. Argue that T is a function of TCPs aver-age throughput.P50. Consider a simplified TCPs AIMD algorithm where the congestion windowsize is measured in number of segments, not in bytes. In additive increase, thecongestion window size increases by one segment in each RTT. In multiplica-tive decrease, the congestion window size decreases by half (if the result isnot an integer, round down to the nearest integer). Suppose that two TCPconnections, C1 and C2, share a single congested link of speed 30 segmentsper second. Assume that both C1 and C2 are in the congestion avoidance1.22 MSSRTT 2LPROBLEMS 299phase. Connection C1s RTT is 50 msec and connection C2s RTT is 100 msec. Assume that when the data rate in the link exceeds the linksspeed, all TCP connections experience data segment loss. a. If both C1 and C2 at time t0 have a congestion window of 10 segments,what are their congestion window sizes after 1000 msec? b. In the long run, will these two connections get the same share of the band-width of the congested link? Explain. P51. Consider the network described in the previous problem. Now suppose thatthe two TCP connections, C1 and C2, have the same RTT of 100 msec. Sup-pose that at time t0, C1s congestion window size is 15 segments but C2scongestion window size is 10 segments. a. What are their congestion window sizes after 2200msec?b. In the long run, will these two connections get about the same share of thebandwidth of the congested link? c. We say that two connections are synchronized, if both connections reachtheir maximum window sizes at the same time and reach their minimumwindow sizes at the same time. In the long run, will these two connectionsget synchronized eventually? If so, what are their maximum window sizes?d. Will this synchronization help to improve the utilization of the sharedlink? Why? Sketch some idea to break this synchronization.P52. Consider a modification to TCPs congestion control algorithm. Instead ofadditive increase, we can use multiplicative increase. A TCP sender increasesits window size by a small positive constant a (0 < a < 1) whenever itreceives a valid ACK. Find the functional relationship between loss rate Land maximum congestion window W. Argue that for this modified TCP,regardless of TCPs average throughput, a TCP connection always spends thesame amount of time to increase its congestion window size from W/2 to W.P53. In our discussion of TCP futures in Section 3.7, we noted that to achieve athroughput of 10 Gbps, TCP could only tolerate a segment loss probability of2 10-10 (or equivalently, one loss event for every 5,000,000,000 segments).Show the derivation for the values of 2 10-10 (1 out of 5,000,000) for theRTT and MSS values given in Section 3.7. If TCP needed to support a 100Gbps connection, what would the tolerable loss be?P54. In our discussion of TCP congestion control in Section 3.7, we implicitlyassumed that the TCP sender always had data to send. Consider now the casethat the TCP sender sends a large amount of data and then goes idle (since ithas no more data to send) at t1. TCP remains idle for a relatively long period oftime and then wants to send more data at t2. What are the advantages and dis-advantages of having TCP use the cwnd and ssthresh values from t1 whenstarting to send data at t2? What alternative would you recommend? Why?300 CHAPTER 3 TRANSPORT LAYERP55. In this problem we investigate whether either UDP or TCP provides a degreeof end-point authentication.a. Consider a server that receives a request within a UDP packet andresponds to that request within a UDP packet (for example, as done by aDNS server). If a client with IP address X spoofs its address with addressY, where will the server send its response?b. Suppose a server receives a SYN with IP source address Y, and afterresponding with a SYNACK, receives an ACK with IP source address Ywith the correct acknowledgment number. Assuming the server chooses arandom initial sequence number and there is no man-in-the-middle, canthe server be certain that the client is indeed at Y (and not at some otheraddress X that is spoofing Y)?P56. In this problem, we consider the delay introduced by the TCP slow-startphase. Consider a client and a Web server directly connected by one link ofrate R. Suppose the client wants to retrieve an object whose size is exactlyequal to 15 S, where S is the maximum segment size (MSS). Denote theround-trip time between client and server as RTT (assumed to be constant).Ignoring protocol headers, determine the time to retrieve the object (includingTCP connection establishment) whena. 4 S/R > S/R + RTT > 2S/R
b. S/R + RTT > 4 S/R
c. S/R > RTT.
Programming Assignments
Implementing a Reliable Transport Protocol
In this laboratory programming assignment, you will be writing the sending and
receiving transport-level code for implementing a simple reliable data transfer pro-
tocol. There are two versions of this lab, the alternating-bit-protocol version and the
GBN version. This lab should be funyour implementation will differ very little
from what would be required in a real-world situation.
Since you probably dont have standalone machines (with an OS that you can
modify), your code will have to execute in a simulated hardware/software envi-
ronment. However, the programming interface provided to your routinesthe
code that would call your entities from above and from belowis very close to
what is done in an actual UNIX environment. (Indeed, the software interfaces
described in this programming assignment are much more realistic than the infi-
nite loop senders and receivers that many texts describe.) Stopping and starting
timers are also simulated, and timer interrupts will cause your timer handling rou-
tine to be activated.
The full lab assignment, as well as code you will need to compile with your
own code, are available at this books Web site: http://www.awl.com/kurose-ross.
Wireshark Lab: Exploring TCP
In this lab, youll use your Web browser to access a file from a Web server. As in
earlier Wireshark labs, youll use Wireshark to capture the packets arriving at your
computer. Unlike earlier labs, youll also be able to download a Wireshark-readable
packet trace from the Web server from which you downloaded the file. In this server
trace, youll find the packets that were generated by your own access of the Web
server. Youll analyze the client- and server-side traces to explore aspects of TCP. In
particular, youll evaluate the performance of the TCP connection between your
computer and the Web server. Youll trace TCPs window behavior, and infer packet
loss, retransmission, flow control and congestion control behavior, and estimated
roundtrip time.
As is the case with all Wireshark labs, the full description of this lab is avail-
able at this books Web site, http://www.awl.com/kurose-ross.
Wireshark Lab: Exploring UDP
In this short lab, youll do a packet capture and analysis of your favorite application
that uses UDP (for example, DNS or a multimedia application such as Skype). As we
learned in Section 3.3, UDP is a simple, no-frills transport protocol. In this lab, youll
investigate the header fields in the UDP segment as well as the checksum calculation.
As is the case with all Wireshark labs, the full description of this lab is available
at this books Web site, http://www.awl.com/kurose-ross.
WIRESHARK LAB: EXPLORING UDP 301
http://www.awl.com/kurose-ross
http://www.awl.com/kurose-ross
http://www.awl.com/kurose-ross
302
Please describe one or two of the most exciting projects you have worked on during your
career. What were the biggest challenges?
School teaches us lots of ways to find answers. In every interesting problem Ive worked
on, the challenge has been finding the right question. When Mike Karels and I started look-
ing at TCP congestion, we spent months staring at protocol and packet traces asking Why
is it failing?. One day in Mikes office, one of us said The reason I cant figure out why it
fails is because I dont understand how it ever worked to begin with. That turned out to be
the right question and it forced us to figure out the ack clocking that makes TCP work.
After that, the rest was easy.
More generally, where do you see the future of networking and the Internet?
For most people, the Web is the Internet. Networking geeks smile politely since we know
the Web is an application running over the Internet but what if theyre right? The Internet is
about enabling conversations between pairs of hosts. The Web is about distributed informa-
tion production and consumption. Information propagation is a very general view of com-
munication of which pairwise conversation is a tiny subset. We need to move into the
larger tent. Networking today deals with broadcast media (radios, PONs, etc.) by pretending
its a point-to-point wire. Thats massively inefficient. Terabits-per-second of data are being
exchanged all over the World via thumb drives or smart phones but we dont know how to
treat that as networking. ISPs are busily setting up caches and CDNs to scalably distribute
video and audio. Caching is a necessary part of the solution but theres no part of todays
networkingfrom Information, Queuing or Traffic Theory down to the Internet protocol
specsthat tells us how to engineer and deploy it. I think and hope that over the next few
years, networking will evolve to embrace the much larger vision of communication that
underlies the Web.
Van Jacobson
Van Jacobson is a Research Fellow at PARC. Prior to that, he was
co-founder and Chief Scientist of Packet Design. Before that, he was
Chief Scientist at Cisco. Before joining Cisco, he was head of the
Network Research Group at Lawrence Berkeley National Laboratory
and taught at UC Berkeley and Stanford. Van received the ACM
SIGCOMM Award in 2001 for outstanding lifetime contribution to
the field of communication networks and the IEEE Kobayashi Award
in 2002 for contributing to the understanding of network congestion
and developing congestion control mechanisms that enabled the suc-
cessful scaling of the Internet. He was elected to the U.S. National
Academy of Engineering in 2004.
AN INTERVIEW WITH
303
What people inspired you professionally?
When I was in grad school, Richard Feynman visited and gave a colloquium. He talked
about a piece of Quantum theory that Id been struggling with all semester and his explana-
tion was so simple and lucid that what had been incomprehensible gibberish to me became
obvious and inevitable. That ability to see and convey the simplicity that underlies our com-
plex world seems to me a rare and wonderful gift.
What are your recommendations for students who want careers in computer science and
networking?
Its a wonderful fieldcomputers and networking have probably had more impact on socie-
ty than any invention since the book. Networking is fundamentally about connecting stuff,
and studying it helps you make intellectual connections: Ant foraging & Bee dances demon-
strate protocol design better than RFCs, traffic jams or people leaving a packed stadium are
the essence of congestion, and students finding flights back to school in a post-Thanksgiving
blizzard are the core of dynamic routing. If youre interested in lots of stuff and want to
have an impact, its hard to imagine a better field.
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CHAPTER 4
The Network
Layer
305
We learned in the previous chapter that the transport layer provides various forms of
process-to-process communication by relying on the network layers host-to-host
communication service. We also learned that the transport layer does so without any
knowledge about how the network layer actually implements this service. So per-
haps youre now wondering, whats under the hood of the host-to-host communica-
tion service, what makes it tick?
In this chapter, well learn exactly how the network layer implements the host-
to-host communication service. Well see that unlike the transport and application
layers, there is a piece of the network layer in each and every host and router in the
network. Because of this, network-layer protocols are among the most challenging
(and therefore among the most interesting!) in the protocol stack.
The network layer is also one of the most complex layers in the protocol stack,
and so well have a lot of ground to cover here. Well begin our study with an
overview of the network layer and the services it can provide. Well then examine
two broad approaches towards structuring network-layer packet deliverythe data-
gram and the virtual-circuit modeland see the fundamental role that addressing
plays in delivering a packet to its destination host.
In this chapter, well make an important distinction between the forwarding
and routing functions of the network layer. Forwarding involves the transfer of a
packet from an incoming link to an outgoing link within a single router. Routing
involves all of a networks routers, whose collective interactions via routing proto-
cols determine the paths that packets take on their trips from source to destination
node. This will be an important distinction to keep in mind as you progress through
this chapter.
In order to deepen our understanding of packet forwarding, well look inside
a routerat its hardware architecture and organization. Well then look at packet
forwarding in the Internet, along with the celebrated Internet Protocol (IP). Well
investigate network-layer addressing and the IPv4 datagram format. Well then
explore network address translation (NAT), datagram fragmentation, the Internet
Control Message Protocol (ICMP), and IPv6.
Well then turn our attention to the network layers routing function. Well see
that the job of a routing algorithm is to determine good paths (equivalently, routes)
from senders to receivers. Well first study the theory of routing algorithms, concen-
trating on the two most prevalent classes of algorithms: link-state and distance-
vector algorithms. Since the complexity of routing algorithms grows considerably
as the number of network routers increases, hierarchical routing approaches will
also be of interest. Well then see how theory is put into practice when we cover the
Internets intra-autonomous system routing protocols (RIP, OSPF, and IS-IS) and its
inter-autonomous system routing protocol, BGP. Well close this chapter with a dis-
cussion of broadcast and multicast routing.
In summary, this chapter has three major parts. The first part, Sections 4.1 and
4.2, covers network-layer functions and services. The second part, Sections 4.3 and
4.4, covers forwarding. Finally, the third part, Sections 4.5 through 4.7, covers
routing.
4.1 Introduction
Figure 4.1 shows a simple network with two hosts, H1 and H2, and several routers
on the path between H1 and H2. Suppose that H1 is sending information to H2, and
consider the role of the network layer in these hosts and in the intervening routers.
The network layer in H1 takes segments from the transport layer in H1, encapsu-
lates each segment into a datagram (that is, a network-layer packet), and then sends
the datagrams to its nearby router, R1. At the receiving host, H2, the network layer
receives the datagrams from its nearby router R2, extracts the transport-layer seg-
ments, and delivers the segments up to the transport layer at H2. The primary role of
the routers is to forward datagrams from input links to output links. Note that the
routers in Figure 4.1 are shown with a truncated protocol stack, that is, with no
upper layers above the network layer, because (except for control purposes) routers
do not run application- and transport-layer protocols such as those we examined in
Chapters 2 and 3.
306 CHAPTER 4 THE NETWORK LAYER
4.1 INTRODUCTION 307
Mobile Network
Router R1
Router R2
National or
Global ISP
Local or
Regional ISP
Enterprise Network
Home Network
End system H1
Data link
Physical
Application
Transport
Network
End system H2
Data link
Physical
Application
Transport
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Network
Data link
Physical
Network
Figure 4.1 The network layer
4.1.1 Forwarding and Routing
The role of the network layer is thus deceptively simpleto move packets from a
sending host to a receiving host. To do so, two important network-layer functions
can be identified:
Forwarding. When a packet arrives at a routers input link, the router must move
the packet to the appropriate output link. For example, a packet arriving from
Host H1 to Router R1 must be forwarded to the next router on a path to H2. In
Section 4.3, well look inside a router and examine how a packet is actually for-
warded from an input link to an output link within a router.
Routing. The network layer must determine the route or path taken by packets as
they flow from a sender to a receiver. The algorithms that calculate these paths
are referred to as routing algorithms. A routing algorithm would determine, for
example, the path along which packets flow from H1 to H2.
The terms forwarding and routing are often used interchangeably by authors dis-
cussing the network layer. Well use these terms much more precisely in this book.
Forwarding refers to the router-local action of transferring a packet from an input link
interface to the appropriate output link interface. Routing refers to the network-wide
process that determines the end-to-end paths that packets take from source to destina-
tion. Using a driving analogy, consider the trip from Pennsylvania to Florida under-
taken by our traveler back in Section 1.3.1. During this trip, our driver passes through
many interchanges en route to Florida. We can think of forwarding as the process of
getting through a single interchange: A car enters the interchange from one road and
determines which road it should take to leave the interchange. We can think of routing
as the process of planning the trip from Pennsylvania to Florida: Before embarking on
the trip, the driver has consulted a map and chosen one of many paths possible, with
each path consisting of a series of road segments connected at interchanges.
Every router has a forwarding table. A router forwards a packet by examin-
ing the value of a field in the arriving packets header, and then using this header
value to index into the routers forwarding table. The value stored in the forward-
ing table entry for that header indicates the routers outgoing link interface to
which that packet is to be forwarded. Depending on the network-layer protocol,
the header value could be the destination address of the packet or an indication of
the connection to which the packet belongs. Figure 4.2 provides an example. In
Figure 4.2, a packet with a header field value of 0111 arrives to a router. The
router indexes into its forwarding table and determines that the output link
interface for this packet is interface 2. The router then internally forwards the
packet to interface 2. In Section 4.3, well look inside a router and examine the
forwarding function in much greater detail.
You might now be wondering how the forwarding tables in the routers are con-
figured. This is a crucial issue, one that exposes the important interplay between
308 CHAPTER 4 THE NETWORK LAYER
routing and forwarding. As shown in Figure 4.2, the routing algorithm determines
the values that are inserted into the routers forwarding tables. The routing algorithm
may be centralized (e.g., with an algorithm executing on a central site and down-
loading routing information to each of the routers) or decentralized (i.e., with a
piece of the distributed routing algorithm running in each router). In either case, a
router receives routing protocol messages, which are used to configure its forward-
ing table. The distinct and different purposes of the forwarding and routing func-
tions can be further illustrated by considering the hypothetical (and unrealistic, but
technically feasible) case of a network in which all forwarding tables are configured
directly by human network operators physically present at the routers. In this case,
no routing protocols would be required! Of course, the human operators would need
to interact with each other to ensure that the forwarding tables were configured in
such a way that packets reached their intended destinations. Its also likely that
human configuration would be more error-prone and much slower to respond to
changes in the network topology than a routing protocol. Were thus fortunate that
all networks have both a forwarding and a routing function!
4.1 INTRODUCTION 309
Value in arriving
packets header
1
2
3
Routing algorithm
Local forwarding table
header value
0100
0101
0111
1001
0111
3
2
2
1
output link
Figure 4.2 Routing algorithms determine values in forwarding tables
While were on the topic of terminology, its worth mentioning two other terms
that are often used interchangeably, but that we will use more carefully. Well reserve
the term packet switch to mean a general packet-switching device that transfers a
packet from input link interface to output link interface, according to the value in a field
in the header of the packet. Some packet switches, called link-layer switches (exam-
ined in Chapter 5), base their forwarding decision on values in the fields of the link-
layer frame; switches are thus referred to as link-layer (layer 2) devices. Other packet
switches, called routers, base their forwarding decision on the value in the network-
layer field. Routers arethus network-layer (layer 3) devices, but must also implement
layer 2 protocols as well, since layer 3 devices require the services of layer 2 to imple-
ment their (layer 3) functionality. (To fully appreciate this important distinction, you
might want to review Section 1.5.2, where we discuss network-layer datagrams and
link-layer frames and their relationship.) To confuse matters, marketing literature often
refers to layer 3 switches for routers with Ethernet interfaces, but these are really
layer 3 devices. Since our focus in this chapter is on the network layer, we use the term
router in place of packet switch. Well even use the term router when talking about
packet switches in virtual-circuit networks (soon to be discussed).
Connection Setup
We just said that the network layer has two important functions, forwarding and rout-
ing. But well soon see that in some computer networks there is actually a third impor-
tant network-layer function, namely, connection setup. Recall from our study of TCP
that a three-way handshake is required before data can flow from sender to receiver.
This allows the sender and receiver to set up the needed state information (for example,
sequence number and initial flow-control window size). In an analogous manner, some
network-layer architecturesfor example, ATM, frame relay, and MPLS (which we
will study in Section 5.8)require the routers along the chosen path from source to
destination to handshake with each other in order to set up state before network-layer
data packets within a given source-to-destination connection can begin to flow. In the
network layer, this process is referred to as connection setup. Well examine connec-
tion setup in Section 4.2.
4.1.2 Network Service Models
Before delving into the network layer, lets take the broader view and consider the dif-
ferent types of service that might be offered by the network layer. When the transport
layer at a sending host transmits a packet into the network (that is, passes it down to
the network layer at the sending host), can the transport layer rely on the network layer
to deliver the packet to the destination? When multiple packets are sent, will they be
delivered to the transport layer in the receiving host in the order in which they were
sent? Will the amount of time between the sending of two sequential packet transmis-
sions be the same as the amount of time between their reception? Will the network
310 CHAPTER 4 THE NETWORK LAYER
provide any feedback about congestion in the network? What is the abstract view
(properties) of the channel connecting the transport layer in the sending and receiving
hosts? The answers to these questions and others are determined by the service model
provided by the network layer. The network service model defines the characteristics
of end-to-end transport of packets between sending and receiving end systems.
Lets now consider some possible services that the network layer could provide.
In the sending host, when the transport layer passes a packet to the network layer,
specific services that could be provided by the network layer include:
Guaranteed delivery. This service guarantees that the packet will eventually
arrive at its destination.
Guaranteed delivery with bounded delay. This service not only guarantees deliv-
ery of the packet, but delivery within a specified host-to-host delay bound (for
example, within 100 msec).
Furthermore, the following services could be provided to a flow of packets between
a given source and destination:
In-order packet delivery. This service guarantees that packets arrive at the desti-
nation in the order that they were sent.
Guaranteed minimal bandwidth. This network-layer service emulates the behavior
of a transmission link of a specified bit rate (for example, 1 Mbps) between send-
ing and receiving hosts. As long as the sending host transmits bits (as part of pack-
ets) at a rate below the specified bit rate, then no packet is lost and each packet
arrives within a prespecified host-to-host delay (for example, within 40 msec).
Guaranteed maximum jitter. This service guarantees that the amount of time
between the transmission of two successive packets at the sender is equal to the
amount of time between their receipt at the destination (or that this spacing
changes by no more than some specified value).
Security services. Using a secret session key known only by a source and desti-
nation host, the network layer in the source host could encrypt the payloads of
all datagrams being sent to the destination host. The network layer in the
destination host would then be responsible for decrypting the payloads. With
such a service, confidentiality would be provided to all transport-layer segments
(TCP and UDP) between the source and destination hosts. In addition to confi-
dentiality, the network layer could provide data integrity and source authentica-
tion services.
This is only a partial list of services that a network layer could providethere are
countless variations possible.
The Internets network layer provides a single service, known as best-effort
service. From Table 4.1, it might appear that best-effort service is a euphemism for
4.1 INTRODUCTION 311
no service at all. With best-effort service, timing between packets is not guaranteed
to be preserved, packets are not guaranteed to be received in the order in which they
were sent, nor is the eventual delivery of transmitted packets guaranteed. Given this
definition, a network that delivered no packets to the destination would satisfy the
definition of best-effort delivery service. As well discuss shortly, however, there
are sound reasons for such a minimalist network-layer service model.
Other network architectures have defined and implemented service models that
go beyond the Internets best-effort service. For example, the ATM network archi-
tecture [MFA Forum 2012, Black 1995] provides for multiple service models, mean-
ing that different connections can be provided with different classes of service
within the same network. A discussion of how an ATM network provides such serv-
ices is well beyond the scope of this book; our aim here is only to note that alterna-
tives do exist to the Internets best-effort model. Two of the more important ATM
service models are constant bit rate and available bit rate service:
Constant bit rate (CBR) ATM network service. This was the first ATM service
model to be standardized, reflecting early interest by the telephone companies in
ATM and the suitability of CBR service for carrying real-time, constant bit rate
audio and video traffic. The goal of CBR service is conceptually simpleto pro-
vide a flow of packets (known as cells in ATM terminology) with a virtual pipe
whose properties are the same as if a dedicated fixed-bandwidth transmission
link existed between sending and receiving hosts. With CBR service, a flow of
ATM cells is carried across the network in such a way that a cells end-to-end
delay, the variability in a cells end-to-end delay (that is, the jitter), and the fraction
of cells that are lost or delivered late are all guaranteed to be less than specified
values. These values are agreed upon by the sending host and the ATM network
when the CBR connection is first established.
312 CHAPTER 4 THE NETWORK LAYER
Table 4.1 Internet, ATM CBR, and ATM ABR service models
Network
Architecture
Service
Model
Bandwidth
Guarantee
No-Loss
Guarantee Ordering Timing
Internet Best Effort None None Any order
possible
Not
maintained
ATM CBR Guaranteed
constant rate
Yes In order Maintained
ATM ABR Guaranteed
minimum
None In order Not
maintained
Congestion
Indication
None
Congestion
will not occur
Congestion
indication
provided
Available bit rate (ABR) ATM network service. With the Internet offering so-
called best-effort service, ATMs ABR might best be characterized as being a
slightly-better-than-best-effort service. As with the Internet service model,
cells may be lost under ABR service. Unlike in the Internet, however, cells
cannot be reordered (although they may be lost), and a minimum cell transmis-
sion rate (MCR) is guaranteed to a connection using ABR service. If the net-
work has enough free resources at a given time, a sender may also be able to
send cells successfully at a higher rate than the MCR. Additionally, as we saw
in Section 3.6, ATM ABR service can provide feedback to the sender (in terms
of a congestion notification bit, or an explicit rate at which to send) that con-
trols how the sender adjusts its rate between the MCR and an allowable peak
cell rate.
4.2 Virtual Circuit and Datagram Networks
Recall from Chapter 3 that a transport layer can offer applications connectionless
service or connection-oriented service between two processes. For example, the Inter-
nets transport layer provides each application a choice between two services: UDP, a
connectionless service; or TCP, a connection-oriented service. In a similar manner, a
network layer can provide connectionless service or connection service between two
hosts. Network-layer connection and connectionless services in many ways parallel
transport-layer connection-oriented and connectionless services. For example, a net-
work-layer connection service begins with handshaking between the source and desti-
nation hosts; and a network-layer connectionless service does not have any
handshaking preliminaries.
Although the network-layer connection and connectionless services have some
parallels with transport-layer connection-oriented and connectionless services, there
are crucial differences:
In the network layer, these services are host-to-host services provided by the net-
work layer for the transport layer. In the transport layer these services are process-
to-process services provided by the transport layer for the application layer.
In all major computer network architectures to date (Internet, ATM, frame relay,
and so on), the network layer provides either a host-to-host connectionless serv-
ice or a host-to-host connection service, but not both. Computer networks that
provide only a connection service at the network layer are called virtual-circuit
(VC) networks; computer networks that provide only a connectionless service
at the network layer are called datagram networks.
The implementations of connection-oriented service in the transport layer and
the connection service in the network layer are fundamentally different. We saw
in the previous chapter that the transport-layer connection-oriented service is
4.2 VIRTUAL CIRCUIT AND DATAGRAM NETWORKS 313
implemented at the edge of the network in the end systems; well see shortly that
the network-layer connection service is implemented in the routers in the net-
work core as well as in the end systems.
Virtual-circuit and datagram networks are two fundamental classes of computer net-
works. They use very different information in making their forwarding decisions.
Lets now take a closer look at their implementations.
4.2.1 Virtual-Circuit Networks
While the Internet is a datagram network, many alternative network architectures
including those of ATM and frame relayare virtual-circuit networks and, there-
fore, use connections at the network layer. These network-layer connections are
called virtual circuits (VCs). Lets now consider how a VC service can be imple-
mented in a computer network.
A VC consists of (1) a path (that is, a series of links and routers) between the
source and destination hosts, (2) VC numbers, one number for each link along the
path, and (3) entries in the forwarding table in each router along the path. A packet
belonging to a virtual circuit will carry a VC number in its header. Because a virtual
circuit may have a different VC number on each link, each intervening router must
replace the VC number of each traversing packet with a new VC number. The new
VC number is obtained from the forwarding table.
To illustrate the concept, consider the network shown in Figure 4.3. The numbers
next to the links of R1 in Figure 4.3 are the link interface numbers. Suppose now that
Host A requests that the network establish a VC between itself and Host B. Suppose
also that the network chooses the path A-R1-R2-B and assigns VC numbers 12, 22,
and 32 to the three links in this path for this virtual circuit. In this case, when a packet
in this VC leaves Host A, the value in the VC number field in the packet header is 12;
when it leaves R1, the value is 22; and when it leaves R2, the value is 32.
How does the router determine the replacement VC number for a packet tra-
versing the router? For a VC network, each routers forwarding table includes VC
314 CHAPTER 4 THE NETWORK LAYER
R1 R2A B
1 2
3
1 2
3
R3 R4
Figure 4.3 A simple virtual circuit network
4.2 VIRTUAL CIRCUIT AND DATAGRAM NETWORKS 315
number translation; for example, the forwarding table in R1 might look something
like this:
Whenever a new VC is established across a router, an entry is added to the forward-
ing table. Similarly, whenever a VC terminates, the appropriate entries in each table
along its path are removed.
You might be wondering why a packet doesnt just keep the same VC number
on each of the links along its route. The answer is twofold. First, replacing the num-
ber from link to link reduces the length of the VC field in the packet header. Second,
and more importantly, VC setup is considerably simplified by permitting a different
VC number at each link along the path of the VC. Specifically, with multiple VC
numbers, each link in the path can choose a VC number independently of the VC
numbers chosen at other links along the path. If a common VC number were required
for all links along the path, the routers would have to exchange and process a sub-
stantial number of messages to agree on a common VC number (e.g., one that is not
being used by any other existing VC at these routers) to be used for a connection.
In a VC network, the networks routers must maintain connection state infor-
mation for the ongoing connections. Specifically, each time a new connection is
established across a router, a new connection entry must be added to the routers for-
warding table; and each time a connection is released, an entry must be removed
from the table. Note that even if there is no VC-number translation, it is still neces-
sary to maintain connection state information that associates VC numbers with out-
put interface numbers. The issue of whether or not a router maintains connection
state information for each ongoing connection is a crucial oneone that well return
to repeatedly in this book.
There are three identifiable phases in a virtual circuit:
VC setup. During the setup phase, the sending transport layer contacts the net-
work layer, specifies the receivers address, and waits for the network to set up
the VC. The network layer determines the path between sender and receiver, that
is, the series of links and routers through which all packets of the VC will travel.
The network layer also determines the VC number for each link along the path.
Finally, the network layer adds an entry in the forwarding table in each router
Incoming Interface Incoming VC # Outgoing Interface Outgoing VC #
1 12 2 22
2 63 1 18
3 7 2 17
1 97 3 87
along the path. During VC setup, the network layer may also reserve resources
(for example, bandwidth) along the path of the VC.
Data transfer. As shown in Figure 4.4, once the VC has been established, pack-
ets can begin to flow along the VC.
VC teardown. This is initiated when the sender (or receiver) informs the network
layer of its desire to terminate the VC. The network layer will then typically
inform the end system on the other side of the network of the call termination
and update the forwarding tables in each of the packet routers on the path to indi-
cate that the VC no longer exists.
There is a subtle but important distinction between VC setup at the network
layer and connection setup at the transport layer (for example, the TCP three-way
handshake we studied in Chapter 3). Connection setup at the transport layer
involves only the two end systems. During transport-layer connection setup, the
two end systems alone determine the parameters (for example, initial sequence
number and flow-control window size) of their transport-layer connection.
Although the two end systems are aware of the transport-layer connection, the
routers within the network are completely oblivious to it. On the other hand, with
a VC network layer, routers along the path between the two end systems are
involved in VC setup, and each router is fully aware of all the VCs passing
through it.
The messages that the end systems send into the network to initiate or terminate a
VC, and the messages passed between the routers to set up the VC (that is, to modify
connection state in router tables) are known as signaling messages, and the protocols
316 CHAPTER 4 THE NETWORK LAYER
Transport
Data link
Physical
Application
Network
Transport
Data link
Physical
Application
Network
1. Initiate call 2. Incoming call
5. Data flow
begins
6. Receive
data
4. Call connected 3. Accept call
Figure 4.4 Virtual-circuit setup
used to exchange these messages are often referred to as signaling protocols. VC setup
is shown pictorially in Figure 4.4. Well not cover VC signaling protocols in this book;
see [Black 1997] for a general discussion of signaling in connection-oriented networks
and [ITU-T Q.2931 1995] for the specification of ATMs Q.2931 signaling protocol.
4.2.2 Datagram Networks
In a datagram network, each time an end system wants to send a packet, it stamps
the packet with the address of the destination end system and then pops the packet
into the network. As shown in Figure 4.5, there is no VC setup and routers do not
maintain any VC state information (because there are no VCs!).
As a packet is transmitted from source to destination, it passes through a series
of routers. Each of these routers uses the packets destination address to forward the
packet. Specifically, each router has a forwarding table that maps destination
addresses to link interfaces; when a packet arrives at the router, the router uses the
packets destination address to look up the appropriate output link interface in the
forwarding table. The router then intentionally forwards the packet to that output
link interface.
To get some further insight into the lookup operation, lets look at a specific
example. Suppose that all destination addresses are 32 bits (which just happens to
be the length of the destination address in an IP datagram). A brute-force implemen-
tation of the forwarding table would have one entry for every possible destination
address. Since there are more than 4 billion possible addresses, this option is totally
out of the question.
4.2 VIRTUAL CIRCUIT AND DATAGRAM NETWORKS 317
Transport
1. Send
data
2. Receive
dataData link
Physical
Application
Network
Transport
Data link
Physical
Application
Network
Figure 4.5 Datagram network
Now lets further suppose that our router has four links, numbered 0 through 3,
and that packets are to be forwarded to the link interfaces as follows:
Destination Address Range Link Interface
11001000 00010111 00010000 00000000
through 0
11001000 00010111 00010111 11111111
11001000 00010111 00011000 00000000
through 1
11001000 00010111 00011000 11111111
11001000 00010111 00011001 00000000
through 2
11001000 00010111 00011111 11111111
otherwise 3
Clearly, for this example, it is not necessary to have 4 billion entries in the routers
forwarding table. We could, for example, have the following forwarding table with
just four entries:
Prefix Match Link Interface
11001000 00010111 00010 0
11001000 00010111 00011000 1
11001000 00010111 00011 2
otherwise 3
With this style of forwarding table, the router matches a prefix of the packets desti-
nation address with the entries in the table; if theres a match, the router forwards
the packet to a link associated with the match. For example, suppose the packets
destination address is 11001000 00010111 00010110 10100001; because the 21-bit
prefix of this address matches the first entry in the table, the router forwards the
packet to link interface 0. If a prefix doesnt match any of the first three entries, then
the router forwards the packet to interface 3. Although this sounds simple enough,
theres an important subtlety here. You may have noticed that it is possible for a des-
tination address to match more than one entry. For example, the first 24 bits of the
address 11001000 00010111 00011000 10101010 match the second entry in the
table, and the first 21 bits of the address match the third entry in the table. When
there are multiple matches, the router uses the longest prefix matching rule; that
is, it finds the longest matching entry in the table and forwards the packet to the link
318 CHAPTER 4 THE NETWORK LAYER
interface associated with the longest prefix match. Well see exactly why this
longest prefix-matching rule is used when we study Internet addressing in more
detail in Section 4.4.
Although routers in datagram networks maintain no connection state informa-
tion, they nevertheless maintain forwarding state information in their forwarding
tables. However, the time scale at which this forwarding state information changes
is relatively slow. Indeed, in a datagram network the forwarding tables are modified
by the routing algorithms, which typically update a forwarding table every one-to-
five minutes or so. In a VC network, a forwarding table in a router is modified
whenever a new connection is set up through the router or whenever an existing
connection through the router is torn down. This could easily happen at a microsec-
ond timescale in a backbone, tier-1 router.
Because forwarding tables in datagram networks can be modified at any time, a
series of packets sent from one end system to another may follow different paths
through the network and may arrive out of order. [Paxson 1997] and [Jaiswal 2003]
present interesting measurement studies of packet reordering and other phenomena
in the public Internet.
4.2.3 Origins of VC and Datagram Networks
The evolution of datagram and VC networks reflects their origins. The notion of a
virtual circuit as a central organizing principle has its roots in the telephony world,
which uses real circuits. With call setup and per-call state being maintained at the
routers within the network, a VC network is arguably more complex than a data-
gram network (although see [Molinero-Fernandez 2002] for an interesting compari-
son of the complexity of circuit- versus packet-switched networks). This, too, is in
keeping with its telephony heritage. Telephone networks, by necessity, had their
complexity within the network, since they were connecting dumb end-system
devices such as rotary telephones. (For those too young to know, a rotary phone is
an analog telephone with no buttonsonly a dial.)
The Internet as a datagram network, on the other hand, grew out of the need to
connect computers together. Given more sophisticated end-system devices, the
Internet architects chose to make the network-layer service model as simple as pos-
sible. As we have already seen in Chapters 2 and 3, additional functionality (for
example, in-order delivery, reliable data transfer, congestion control, and DNS name
resolution) is then implemented at a higher layer, in the end systems. This inverts
the model of the telephone network, with some interesting consequences:
Since the resulting Internet network-layer service model makes minimal (no!)
service guarantees, it imposes minimal requirements on the network layer. This
makes it easier to interconnect networks that use very different link-layer tech-
nologies (for example, satellite, Ethernet, fiber, or radio) that have very different
transmission rates and loss characteristics. We will address the interconnection
of IP networks in detail in Section 4.4.
4.2 VIRTUAL CIRCUIT AND DATAGRAM NETWORKS 319
As we saw in Chapter 2, applications such as e-mail, the Web, and even some
network infrastructure services such as the DNS are implemented in hosts
(servers) at the network edge. The ability to add a new service simply by attach-
ing a host to the network and defining a new application-layer protocol (such as
HTTP) has allowed new Internet applications such as the Web to be deployed in
a remarkably short period of time.
4.3 Whats Inside a Router?
Now that weve overviewed the network layers services and functions, lets turn
our attention to its forwarding functionthe actual transfer of packets from a
routers incoming links to the appropriate outgoing links at that router. We
already took a brief look at a few aspects of forwarding in Section 4.2, namely,
addressing and longest prefix matching. We mention here in passing that the terms
forwarding and switching are often used interchangeably by computer-networking
researchers and practitioners; well use both terms interchangeably in this
textbook as well.
A high-level view of a generic router architecture is shown in Figure 4.6. Four
router components can be identified:
Input ports. An input port performs several key functions. It performs the
physical layer function of terminating an incoming physical link at a router;
this is shown in the leftmost box of the input port and the rightmost box of the
output port in Figure 4.6. An input port also performs link-layer functions
needed to interoperate with the link layer at the other side of the incoming
link; this is represented by the middle boxes in the input and output ports. Per-
haps most crucially, the lookup function is also performed at the input port;
this will occur in the rightmost box of the input port. It is here that the for-
warding table is consulted to determine the router output port to which an
arriving packet will be forwarded via the switching fabric. Control packets
(for example, packets carrying routing protocol information) are forwarded
from an input port to the routing processor. Note that the term port here
referring to the physical input and output router interfacesis distinctly
different from the software ports associated with network applications and
sockets discussed in Chapters 2 and 3.
Switching fabric. The switching fabric connects the routers input ports to its
output ports. This switching fabric is completely contained within the router
a network inside of a network router!
Output ports. An output port stores packets received from the switching fabric
and transmits these packets on the outgoing link by performing the necessary
link-layer and physical-layer functions. When a link is bidirectional (that is,
320 CHAPTER 4 THE NETWORK LAYER
carries traffic in both directions), an output port will typically be paired with the
input port for that link on the same line card (a printed circuit board containing
one or more input ports, which is connected to the switching fabric).
Routing processor. The routing processor executes the routing protocols (which
well study in Section 4.6), maintains routing tables and attached link state infor-
mation, and computes the forwarding table for the router. It also performs the
network management functions that well study in Chapter 9.
Recall that in Section 4.1.1 we distinguished between a routers forwarding and
routing functions. A routers input ports, output ports, and switching fabric
together implement the forwarding function and are almost always implemented
in hardware, as shown in Figure 4.6. These forwarding functions are sometimes
collectively referred to as the router forwarding plane. To appreciate why a
hardware implementation is needed, consider that with a 10 Gbps input link and a
64-byte IP datagram, the input port has only 51.2 ns to process the datagram
before another datagram may arrive. If N ports are combined on a line card (as is
often done in practice), the datagram-processing pipeline must operate N times
fasterfar too fast for software implementation. Forwarding plane hardware can
be implemented either using a router vendors own hardware designs, or con-
structed using purchased merchant-silicon chips (e.g., as sold by companies such
as Intel and Broadcom).
While the forwarding plane operates at the nanosecond time scale, a routers
control functionsexecuting the routing protocols, responding to attached links that
4.3 WHATS INSIDE A ROUTER? 321
Input port Output port
Input port Output port
Routing
processor
Routing, management
control plane (software)
Forwarding
data plane (hardware)
Switch
fabric
Figure 4.6 Router architecture
go up or down, and performing management functions such as those well study in
Chapter 9operate at the millisecond or second timescale. These router control
plane functions are usually implemented in software and execute on the routing
processor (typically a traditional CPU).
Before delving into the details of a routers control and data plane, lets return to
our analogy of Section 4.1.1, where packet forwarding was compared to cars entering
and leaving an interchange. Lets suppose that the interchange is a roundabout, and that
before a car enters the roundabout, a bit of processing is requiredthe car stops at an
entry station and indicates its final destination (not at the local roundabout, but the ulti-
mate destination of its journey). An attendant at the entry station looks up the final des-
tination, determines the roundabout exit that leads to that final destination, and tells the
driver which roundabout exit to take. The car enters the roundabout (which may be
filled with other cars entering from other input roads and heading to other roundabout
exits) and eventually leaves at the prescribed roundabout exit ramp, where it may
encounter other cars leaving the roundabout at that exit.
We can recognize the principal router components in Figure 4.6 in this anal-
ogythe entry road and entry station correspond to the input port (with a lookup
function to determine to local outgoing port); the roundabout corresponds to the
switch fabric; and the roundabout exit road corresponds to the output port. With
this analogy, its instructive to consider where bottlenecks might occur. What hap-
pens if cars arrive blazingly fast (for example, the roundabout is in Germany or
Italy!) but the station attendant is slow? How fast must the attendant work to ensure
theres no backup on an entry road? Even with a blazingly fast attendant, what hap-
pens if cars traverse the roundabout slowlycan backups still occur? And what
happens if most of the entering cars all want to leave the roundabout at the same
exit rampcan backups occur at the exit ramp or elsewhere? How should the
roundabout operate if we want to assign priorities to different cars, or block certain
cars from entering the roundabout in the first place? These are all analogous to crit-
ical questions faced by router and switch designers.
In the following subsections, well look at router functions in more detail. [Iyer
2008, Chao 2001; Chuang 2005; Turner 1988; McKeown 1997a; Partridge 1998]
provide a discussion of specific router architectures. For concreteness, the ensuing
discussion assumes a datagram network in which forwarding decisions are based
on the packets destination address (rather than a VC number in a virtual-circuit
network). However, the concepts and techniques are quite similar for a virtual-
circuit network.
4.3.1 Input Processing
A more detailed view of input processing is given in Figure 4.7. As discussed above,
the input ports line termination function and link-layer processing implement the
physical and link layers for that individual input link. The lookup performed in the
input port is central to the routers operationit is here that the router uses the for-
warding table to look up the output port to which an arriving packet will be
322 CHAPTER 4 THE NETWORK LAYER
forwarded via the switching fabric. The forwarding table is computed and updated
by the routing processor, with a shadow copy typically stored at each input port. The
forwarding table is copied from the routing processor to the line cards over a sepa-
rate bus (e.g., a PCI bus) indicated by the dashed line from the routing processor to
the input line cards in Figure 4.6. With a shadow copy, forwarding decisions can be
made locally, at each input port, without invoking the centralized routing processor
on a per-packet basis and thus avoiding a centralized processing bottleneck.
Given the existence of a forwarding table, lookup is conceptually simplewe just
search through the forwarding table looking for the longest prefix match, as described
4.3 WHATS INSIDE A ROUTER? 323
Line
termination
Data link
processing
(protocol,
decapsulation)
Lookup, fowarding,
queuing Switch
fabric
Figure 4.7 Input port processing
CISCO SYSTEMS: DOMINATING THE NETWORK CORE
As of this writing 2012, Cisco employs more than 65,000 people. How did this
gorilla of a networking company come to be? It all started in 1984 in the living room
of a Silicon Valley apartment.
Len Bosak and his wife Sandy Lerner were working at Stanford University when they
had the idea to build and sell Internet routers to research and academic institutions, the
primary adopters of the Internet at that time. Sandy Lerner came up with the name Cisco
(an abbreviation for San Francisco), and she also designed the companys bridge logo.
Corporate headquarters was their living room, and they financed the project with credit
cards and moonlighting consulting jobs. At the end of 1986, Ciscos revenues reached
$250,000 a month. At the end of 1987, Cisco succeeded in attracting venture capital
$2 million from Sequoia Capital in exchange for one-third of the company. Over the next
few years, Cisco continued to grow and grab more and more market share. At the same
time, relations between Bosak/Lerner and Cisco management became strained. Cisco
went public in 1990; in the same year Lerner and Bosak left the company.
Over the years, Cisco has expanded well beyond the router market, selling security,
wireless caching, Ethernet switch, datacenter infrastructure, video conferencing, and
voice-over IP products and services. However, Cisco is facing increased international
competition, including from Huawei, a rapidly growing Chinese network-gear compa-
ny. Other sources of competition for Cisco in the router and switched Ethernet space
include Alcatel-Lucent and Juniper.
CASE HISTORY
in Section 4.2.2. But at Gigabit transmission rates, this lookup must be performed in
nanoseconds (recall our earlier example of a 10 Gbps link and a 64-byte IP datagram).
Thus, not only must lookup be performed in hardware, but techniques beyond a simple
linear search through a large table are needed; surveys of fast lookup algorithms can be
found in [Gupta 2001, Ruiz-Sanchez 2001]. Special attention must also be paid to mem-
ory access times, resulting in designs with embedded on-chip DRAM and faster SRAM
(used as a DRAM cache) memories. Ternary Content Address Memories (TCAMs) are
also often used for lookup. With a TCAM, a 32-bit IP address is presented to the mem-
ory, which returns the content of the forwarding table entry for that address in essen-
tially constant time. The Cisco 8500 has a 64K CAM for each input port.
Once a packets output port has been determined via the lookup, the packet can
be sent into the switching fabric. In some designs, a packet may be temporarily
blocked from entering the switching fabric if packets from other input ports are cur-
rently using the fabric. A blocked packet will be queued at the input port and then
scheduled to cross the fabric at a later point in time. Well take a closer look at the
blocking, queuing, and scheduling of packets (at both input ports and output ports)
in Section 4.3.4. Although lookup is arguably the most important action in input
port processing, many other actions must be taken: (1) physical- and link-layer pro-
cessing must occur, as discussed above; (2) the packets version number, checksum
and time-to-live fieldall of which well study in Section 4.4.1must be checked
and the latter two fields rewritten;and (3) counters used for network management
(such as the number of IP datagrams received) must be updated.
Lets close our discussion of input port processing by noting that the input port
steps of looking up an IP address (match) then sending the packet into the switching
fabric (action) is a specific case of a more general match plus action abstraction
that is performed in many networked devices, not just routers. In link-layer switches
(covered in Chapter 5), link-layer destination addresses are looked up and several
actions may be taken in addition to sending the frame into the switching fabric towards
the output port. In firewalls (covered in Chapter 8)devices that filter out selected
incoming packetsan incoming packet whose header matches a given criteria (e.g., a
combination of source/destination IP addresses and transport-layer port numbers) may
be prevented from being forwarded (action). In a network address translator (NAT, cov-
ered in Section 4.4), an incoming packet whose transport-layer port number matches a
given value will have its port number rewritten before forwarding (action). Thus, the
match plus action abstraction is both powerful and prevalent in network devices.
4.3.2 Switching
The switching fabric is at the very heart of a router, as it is through this fabric that
the packets are actually switched (that is, forwarded) from an input port to an output
port. Switching can be accomplished in a number of ways, as shown in Figure 4.8:
Switching via memory. The simplest, earliest routers were traditional computers,
with switching between input and output ports being done under direct control of
324 CHAPTER 4 THE NETWORK LAYER
the CPU (routing processor). Input and output ports functioned as traditional I/O
devices in a traditional operating system. An input port with an arriving packet
first signaled the routing processor via an interrupt. The packet was then copied
from the input port into processor memory. The routing processor then extracted
the destination address from the header, looked up the appropriate output port in
the forwarding table, and copied the packet to the output ports buffers. In this
scenario, if the memory bandwidth is such that B packets per second can be writ-
ten into, or read from, memory, then the overall forwarding throughput (the total
rate at which packets are transferred from input ports to output ports) must be
less than B/2. Note also that two packets cannot be forwarded at the same time,
even if they have different destination ports, since only one memory read/write
over the shared system bus can be done at a time.
Many modern routers switch via memory. A major difference from early routers,
however, is that the lookup of the destination address and the storing of the packet
into the appropriate memory location are performed by processing on the input
line cards. In some ways, routers that switch via memory look very much like
shared-memory multiprocessors, with the processing on a line card switching
(writing) packets into the memory of the appropriate output port. Ciscos Catalyst
8500 series switches [Cisco 8500 2012] forward packets via a shared memory.
4.3 WHATS INSIDE A ROUTER? 325
Memory
A
B
C
X
Y
Z
Memory
Key:
Input port Output port
A
X Y Z
B
C
Crossbar
A
B
C
X
Y
Z
Bus
Figure 4.8 Three switching techniques
Switching via a bus. In this approach, an input port transfers a packet directly to the
output port over a shared bus, without intervention by the routing processor. This is
typically done by having the input port pre-pend a switch-internal label (header) to
the packet indicating the local output port to which this packet is being transferred
and transmitting the packet onto the bus. The packet is received by all output ports,
but only the port that matches the label will keep the packet. The label is then
removed at the output port, as this label is only used within the switch to cross the
bus. If multiple packets arrive to the router at the same time, each at a different input
port, all but one must wait since only one packet can cross the bus at a time. Because
every packet must cross the single bus, the switching speed of the router is limited
to the bus speed; in our roundabout analogy, this is as if the roundabout could only
contain one car at a time. Nonetheless, switching via a bus is often sufficient for
routers that operate in small local area and enterprise networks. The Cisco 5600
[Cisco Switches 2012] switches packets over a 32 Gbps backplane bus.
Switching via an interconnection network. One way to overcome the bandwidth
limitation of a single, shared bus is to use a more sophisticated interconnection net-
work, such as those that have been used in the past to interconnect processors in a
multiprocessor computer architecture. A crossbar switch is an interconnection net-
work consisting of 2N buses that connect N input ports to N output ports, as shown
in Figure 4.8. Each vertical bus intersects each horizontal bus at a crosspoint, which
can be opened or closed at any time by the switch fabric controller (whose logic is
part of the switching fabric itself). When a packet arrives from port A and needs to
be forwarded to port Y, the switch controller closes the crosspoint at the intersection
of busses A and Y, and port A then sends the packet onto its bus, which is picked up
(only) by bus Y. Note that a packet from port B can be forwarded to port X at the
same time, since the A-to-Y and B-to-X packets use different input and output
busses. Thus, unlike the previous two switching approaches, crossbar networks are
capable of forwarding multiple packets in parallel. However, if two packets from
two different input ports are destined to the same output port, then one will have to
wait at the input, since only one packet can be sent over any given bus at a time.
More sophisticated interconnection networks use multiple stages of switching
elements to allow packets from different input ports to proceed towards the same
output port at the same time through the switching fabric. See [Tobagi 1990] for
a survey of switch architectures. Cisco 12000 family switches [Cisco 12000
2012] use an interconnection network.
4.3.3 Output Processing
Output port processing, shown in Figure 4.9, takes packets that have been stored in
the output ports memory and transmits them over the output link. This includes
selecting and de-queueing packets for transmission, and performing the needed link-
layer and physical-layer transmission functions.
326 CHAPTER 4 THE NETWORK LAYER
4.3.4 Where Does Queueing Occur?
If we consider input and output port functionality and the configurations shown in
Figure 4.8, its clear that packet queues may form at both the input ports and the out-
put ports, just as we identified cases where cars may wait at the inputs and outputs of
the traffic intersection in our roundabout analogy. The location and extent of queueing
(either at the input port queues or the output port queues) will depend on the traffic
load, the relative speed of the switching fabric, and the line speed. Lets now consider
these queues in a bit more detail, since as these queues grow large, the routers mem-
ory can eventually be exhausted and packet loss will occur when no memory is avail-
able to store arriving packets. Recall that in our earlier discussions, we said that
packets were lost within the network or dropped at a router. It is here, at these
queues within a router, where such packets are actually dropped and lost.
Suppose that the input and output line speeds (transmission rates) all have an
identical transmission rate of Rline packets per second, and that there are N input
ports and N output ports. To further simplify the discussion, lets assume that all
packets have the same fixed length, and the packets arrive to input ports in a syn-
chronous manner. That is, the time to send a packet on any link is equal to the time
to receive a packet on any link, and during such an interval of time, either zero or
one packet can arrive on an input link. Define the switching fabric transfer rate
Rswitch as the rate at which packets can be moved from input port to output port. If
Rswitch is N times faster than Rline, then only negligible queuing will occur at the
input ports. This is because even in the worst case, where all N input lines are
receiving packets, and all packets are to be forwarded to the same output port, each
batch of N packets (one packet per input port) can be cleared through the switch fab-
ric before the next batch arrives.
But what can happen at the output ports? Lets suppose that Rswitch is still N
times faster than Rline. Once again, packets arriving at each of the N input ports
are destined to the same output port. In this case, in the time it takes to send a single
packet onto the outgoing link, N new packets will arrive at this output port. Since
the output port can transmit only a single packet in a unit of time (the packet trans-
mission time), the N arriving packets will have to queue (wait) for transmission over
the outgoing link. Then N more packets can possibly arrive in the time it takes to
4.3 WHATS INSIDE A ROUTER? 327
Line
termination
Data link
processing
(protocol,
encapsulation)
Queuing (buffer
management)Switch
fabric
Figure 4.9 Output port processing
transmit just one of the N packets that had just previously been queued. And so on.
Eventually, the number of queued packets can grow large enough to exhaust avail-
able memory at the output port, in which case packets are dropped.
Output port queuing is illustrated in Figure 4.10. At time t, a packet has arrived at
each of the incoming input ports, each destined for the uppermost outgoing port.
Assuming identical line speeds and a switch operating at three times the line speed,
one time unit later (that is, in the time needed to receive or send a packet), all three
original packets have been transferred to the outgoing port and are queued awaiting
transmission. In the next time unit, one of these three packets will have been transmit-
ted over the outgoing link. In our example, two new packets have arrived at the incom-
ing side of the switch; one of these packets is destined for this uppermost output port.
Given that router buffers are needed to absorb the fluctuations in traffic load, the
natural question to ask is how much buffering is required. For many years, the rule of
thumb [RFC 3439] for buffer sizing was that the amount of buffering (B) should be
equal to an average round-trip time (RTT, say 250 msec) times the link capacity (C).
This result is based on an analysis of the queueing dynamics of a relatively small num-
ber of TCP flows [Villamizar 1994]. Thus, a 10 Gbps link with an RTT of 250 msec
would need an amount of buffering equal to B = RTT C = 2.5 Gbits of buffers. Recent
328 CHAPTER 4 THE NETWORK LAYER
Switch
fabric
Output port contention at time t
One packet time later
Switch
fabric
Figure 4.10 Output port queuing
theoretical and experimental efforts [Appenzeller 2004], however, suggest that when
there are a large number of TCP flows (N) passing through a link, the amount of buffer-
ing needed is B = RTT C/N
. With a large number of flows typically passing through
large backbone router links (see, e.g., [Fraleigh 2003]), the value of N can be large, with
the decrease in needed buffer size becoming quite significant. [Appenzellar 2004; Wis-
chik 2005; Beheshti 2008] provide very readable discussions of the buffer sizing prob-
lem from a theoretical, implementation, and operational standpoint.
A consequence of output port queuing is that a packet scheduler at the output
port must choose one packet among those queued for transmission. This selection
might be done on a simple basis, such as first-come-first-served (FCFS) scheduling,
or a more sophisticated scheduling discipline such as weighted fair queuing (WFQ),
which shares the outgoing link fairly among the different end-to-end connections
that have packets queued for transmission. Packet scheduling plays a crucial role in
providing quality-of-service guarantees. Well thus cover packet scheduling exten-
sively in Chapter 7. A discussion of output port packet scheduling disciplines is
[Cisco Queue 2012].
Similarly, if there is not enough memory to buffer an incoming packet, a decision
must be made to either drop the arriving packet (a policy known as drop-tail) or
remove one or more already-queued packets to make room for the newly arrived
packet. In some cases, it may be advantageous to drop (or mark the header of) a packet
before the buffer is full in order to provide a congestion signal to the sender. A number
of packet-dropping and -marking policies (which collectively have become known as
active queue management (AQM) algorithms) have been proposed and analyzed
[Labrador 1999, Hollot 2002]. One of the most widely studied and implemented AQM
algorithms is the Random Early Detection (RED) algorithm. Under RED, a
weighted average is maintained for the length of the output queue. If the average
queue length is less than a minimum threshold, minth, when a packet arrives, the
packet is admitted to the queue. Conversely, if the queue is full or the average queue
length is greater than a maximum threshold, maxth, when a packet arrives, the packet
is marked or dropped. Finally, if the packet arrives to find an average queue length in
the interval [minth, maxth], the packet is marked or dropped with a probability that is
typically some function of the average queue length, minth, and maxth. A number of
probabilistic marking/dropping functions have been proposed, and various versions of
RED have been analytically modeled, simulated, and/or implemented. [Christiansen
2001] and [Floyd 2012] provide overviews and pointers to additional reading.
4.3 WHATS INSIDE A ROUTER? 329
If the switch fabric is not fast enough (relative to the input line speeds) to transfer
all arriving packets through the fabric without delay, then packet queuing can also
occur at the input ports, as packets must join input port queues to wait their turn to be
transferred through the switching fabric to the output port. To illustrate an important
consequence of this queuing, consider a crossbar switching fabric and suppose that
(1) all link speeds are identical, (2) that one packet can be transferred from any one
input port to a given output port in the same amount of time it takes for a packet to be
received on an input link, and (3) packets are moved from a given input queue to their
desired output queue in an FCFS manner. Multiple packets can be transferred in paral-
lel, as long as their output ports are different. However, if two packets at the front of
two input queues are destined for the same output queue, then one of the packets will
be blocked and must wait at the input queuethe switching fabric can transfer only
one packet to a given output port at a time.
Figure 4.11 shows an example in which two packets (darkly shaded) at the front
of their input queues are destined for the same upper-right output port. Suppose that
the switch fabric chooses to transfer the packet from the front of the upper-left
queue. In this case, the darkly shaded packet in the lower-left queue must wait. But
not only must this darkly shaded packet wait, so too must the lightly shaded packet
that is queued behind that packet in the lower-left queue, even though there is no
contention for the middle-right output port (the destination for the lightly shaded
packet). This phenomenon is known as head-of-the-line (HOL) blocking in an
330 CHAPTER 4 THE NETWORK LAYER
Switch
fabric
Output port contention at time t
one dark packet can be transferred
Light blue packet experiences HOL blocking
Switch
fabric
Key:
destined for upper output
port
destined for middle output
port
destined for lower output
port
Figure 4.11 HOL blocking at an input queued switch
input-queued switcha queued packet in an input queue must wait for transfer
through the fabric (even though its output port is free) because it is blocked by
another packet at the head of the line. [Karol 1987] shows that due to HOL block-
ing, the input queue will grow to unbounded length (informally, this is equivalent to
saying that significant packet loss will occur) under certain assumptions as soon as
the packet arrival rate on the input links reaches only 58 percent of their capacity. A
number of solutions to HOL blocking are discussed in [McKeown 1997b].
4.3.5 The Routing Control Plane
In our discussion thus far and in Figure 4.6, weve implicitly assumed that the rout-
ing control plane fully resides and executes in a routing processor within the router.
The network-wide routing control plane is thus decentralizedwith different pieces
(e.g., of a routing algorithm) executing at different routers and interacting by send-
ing control messages to each other. Indeed, todays Internet routers and the routing
algorithms well study in Section 4.6 operate in exactly this manner. Additionally,
router and switch vendors bundle their hardware data plane and software control
plane together into closed (but inter-operable) platforms in a vertically integrated
product.
Recently, a number of researchers [Caesar 2005a, Casado 2009, McKeown
2008] have begun exploring new router control plane architectures in which part of
the control plane is implemented in the routers (e.g., local measurement/reporting of
link state, forwarding table installation and maintenance) along with the data plane,
and part of the control plane can be implemented externally to the router (e.g., in a
centralized server, which could perform route calculation). A well-defined API dic-
tates how these two parts interact and communicate with each other. These
researchers argue that separating the software control plane from the hardware data
plane (with a minimal router-resident control plane) can simplify routing by replac-
ing distributed routing calculation with centralized routing calculation, and enable
network innovation by allowing different customized control planes to operate over
fast hardware data planes.
4.4 The Internet Protocol (IP): Forwarding and
Addressing in the Internet
Our discussion of network-layer addressing and forwarding thus far has been
without reference to any specific computer network. In this section, well turn our
attention to how addressing and forwarding are done in the Internet. Well see that
Internet addressing and forwarding are important components of the Internet
Protocol (IP). There are two versions of IP in use today. Well first examine the
widely deployed IP protocol version 4, which is usually referred to simply as IPv4
4.4 THE INTERNET PROTOCOL (IP) 331
[RFC 791]. Well examine IP version 6 [RFC 2460; RFC 4291], which has been
proposed to replace IPv4, at the end of this section.
But before beginning our foray into IP, lets take a step back and consider the
components that make up the Internets network layer. As shown in Figure 4.12,
the Internets network layer has three major components. The first component is
the IP protocol, the topic of this section. The second major component is the rout-
ing component, which determines the path a datagram follows from source to des-
tination. We mentioned earlier that routing protocols compute the forwarding
tables that are used to forward packets through the network. Well study the
Internets routing protocols in Section 4.6. The final component of the network
layer is a facility to report errors in datagrams and respond to requests for certain
network-layer information. Well cover the Internets network-layer error- and
information-reporting protocol, the Internet Control Message Protocol (ICMP), in
Section 4.4.3.
4.4.1 Datagram Format
Recall that a network-layer packet is referred to as a datagram. We begin our study
of IP with an overview of the syntax and semantics of the IPv4 datagram. You
might be thinking that nothing could be drier than the syntax and semantics of a
packets bits. Nevertheless, the datagram plays a central role in the Internetevery
networking student and professional needs to see it, absorb it, and master it. The
332 CHAPTER 4 THE NETWORK LAYER
Routing protocols
path selection
RIP, OSPF, BGP
IP protocol
addressing conventions
datagram format
packet handling
conventions
ICMP protocol
error reporting
router signaling
Forwarding
table
Transport layer: TCP, UDP
Link layer
Physical layer
Network layer
Figure 4.12 A look inside the Internets network layer
IPv4 datagram format is shown in Figure 4.13. The key fields in the IPv4 datagram
are the following:
Version number. These 4 bits specify the IP protocol version of the datagram.
By looking at the version number, the router can determine how to interpret
the remainder of the IP datagram. Different versions of IP use different data-
gram formats. The datagram format for the current version of IP, IPv4, is
shown in Figure 4.13. The datagram format for the new version of IP (IPv6) is
discussed at the end of this section.
Header length. Because an IPv4 datagram can contain a variable number of
options (which are included in the IPv4 datagram header), these 4 bits are needed
to determine where in the IP datagram the data actually begins. Most IP data-
grams do not contain options, so the typical IP datagram has a 20-byte header.
Type of service. The type of service (TOS) bits were included in the IPv4 header
to allow different types of IP datagrams (for example, datagrams particularly
requiring low delay, high throughput, or reliability) to be distinguished from each
other. For example, it might be useful to distinguish real-time datagrams (such as
those used by an IP telephony application) from non-real-time traffic (for exam-
ple, FTP). The specific level of service to be provided is a policy issue deter-
mined by the routers administrator. Well explore the topic of differentiated
service in Chapter 7.
4.4 THE INTERNET PROTOCOL (IP) 333
Version Type of service
Header
length
Upper-layer
protocol
16-bit Identifier
Time-to-live
13-bit Fragmentation offsetFlags
Datagram length (bytes)
Header checksum
32 bits
32-bit Source IP address
32-bit Destination IP address
Options (if any)
Data
Figure 4.13 IPv4 datagram format
Datagram length. This is the total length of the IP datagram (header plus data),
measured in bytes. Since this field is 16 bits long, the theoretical maximum size
of the IP datagram is 65,535 bytes. However, datagrams are rarely larger than
1,500 bytes.
Identifier, flags, fragmentation offset. These three fields have to do with so-called
IP fragmentation, a topic we will consider in depth shortly. Interestingly, the new
version of IP, IPv6, does not allow for fragmentation at routers.
Time-to-live. The time-to-live (TTL) field is included to ensure that datagrams
do not circulate forever (due to, for example, a long-lived routing loop) in the
network. This field is decremented by one each time the datagram is processed
by a router. If the TTL field reaches 0, the datagram must be dropped.
Protocol. This field is used only when an IP datagram reaches its final destina-
tion. The value of this field indicates the specific transport-layer protocol to
which the data portion of this IP datagram should be passed. For example, a
value of 6 indicates that the data portion is passed to TCP, while a value of 17
indicates that the data is passed to UDP. For a list of all possible values, see
[IANA Protocol Numbers 2012]. Note that the protocol number in the IP data-
gram has a role that is analogous to the role of the port number field in the transport-
layer segment. The protocol number is the glue that binds the network and transport
layers together, whereas the port number is the glue that binds the transport and
application layers together. Well see in Chapter 5 that the link-layer frame also
has a special field that binds the link layer to the network layer.
Header checksum. The header checksum aids a router in detecting bit errors in a
received IP datagram. The header checksum is computed by treating each 2 bytes
in the header as a number and summing these numbers using 1s complement
arithmetic. As discussed in Section 3.3, the 1s complement of this sum, known
as the Internet checksum, is stored in the checksum field. A router computes the
header checksum for each received IP datagram and detects an error condition if
the checksum carried in the datagram header does not equal the computed check-
sum. Routers typically discard datagrams for which an error has been detected.
Note that the checksum must be recomputed and stored again at each router, as
the TTL field, and possibly the options field as well, may change. An interesting
discussion of fast algorithms for computing the Internet checksum is [RFC
1071]. A question often asked at this point is, why does TCP/IP perform error
checking at both the transport and network layers? There are several reasons for
this repetition. First, note that only the IP header is checksummed at the IP layer,
while the TCP/UDP checksum is computed over the entire TCP/UDP segment.
Second, TCP/UDP and IP do not necessarily both have to belong to the same pro-
tocol stack. TCP can, in principle, run over a different protocol (for example,
ATM) and IP can carry data that will not be passed to TCP/UDP.
Source and destination IP addresses. When a source creates a datagram, it inserts
its IP address into the source IP address field and inserts the address of the
334 CHAPTER 4 THE NETWORK LAYER
ultimate destination into the destination IP address field. Often the source host
determines the destination address via a DNS lookup, as discussed in Chapter 2.
Well discuss IP addressing in detail in Section 4.4.2.
Options. The options fields allow an IP header to be extended. Header options
were meant to be used rarelyhence the decision to save overhead by not
including the information in options fields in every datagram header. However,
the mere existence of options does complicate matterssince datagram headers
can be of variable length, one cannot determine a priori where the data field will
start. Also, since some datagrams may require options processing and others may
not, the amount of time needed to process an IP datagram at a router can vary
greatly. These considerations become particularly important for IP processing in
high-performance routers and hosts. For these reasons and others, IP options
were dropped in the IPv6 header, as discussed in Section 4.4.4.
Data (payload). Finally, we come to the last and most important fieldthe rai-
son dtre for the datagram in the first place! In most circumstances, the data
field of the IP datagram contains the transport-layer segment (TCP or UDP) to
be delivered to the destination. However, the data field can carry other types of
data, such as ICMP messages (discussed in Section 4.4.3).
Note that an IP datagram has a total of 20 bytes of header (assuming no options). If
the datagram carries a TCP segment, then each (nonfragmented) datagram carries a
total of 40 bytes of header (20 bytes of IP header plus 20 bytes of TCP header) along
with the application-layer message.
IP Datagram Fragmentation
Well see in Chapter 5 that not all link-layer protocols can carry network-layer pack-
ets of the same size. Some protocols can carry big datagrams, whereas other proto-
cols can carry only little packets. For example, Ethernet frames can carry up to 1,500
bytes of data, whereas frames for some wide-area links can carry no more than 576
bytes. The maximum amount of data that a link-layer frame can carry is called the
maximum transmission unit (MTU). Because each IP datagram is encapsulated
within the link-layer frame for transport from one router to the next router, the MTU
of the link-layer protocol places a hard limit on the length of an IP datagram. Having
a hard limit on the size of an IP datagram is not much of a problem. What is a prob-
lem is that each of the links along the route between sender and destination can use
different link-layer protocols, and each of these protocols can have different MTUs.
To understand the forwarding issue better, imagine that you are a router that
interconnects several links, each running different link-layer protocols with differ-
ent MTUs. Suppose you receive an IP datagram from one link. You check your for-
warding table to determine the outgoing link, and this outgoing link has an MTU
that is smaller than the length of the IP datagram. Time to panichow are you going
to squeeze this oversized IP datagram into the payload field of the link-layer frame?
4.4 THE INTERNET PROTOCOL (IP) 335
The solution is to fragment the data in the IP datagram into two or more smaller IP
datagrams, encapsulate each of these smaller IP datagrams in a separate link-layer
frame; and send these frames over the outgoing link. Each of these smaller data-
grams is referred to as a fragment.
Fragments need to be reassembled before they reach the transport layer at the des-
tination. Indeed, both TCP and UDP are expecting to receive complete, unfragmented
segments from the network layer. The designers of IPv4 felt that reassembling data-
grams in the routers would introduce significant complication into the protocol and
put a damper on router performance. (If you were a router, would you want to be
reassembling fragments on top of everything else you had to do?) Sticking to the prin-
ciple of keeping the network core simple, the designers of IPv4 decided to put the job
of datagram reassembly in the end systems rather than in network routers.
When a destination host receives a series of datagrams from the same source, it
needs to determine whether any of these datagrams are fragments of some original,
larger datagram. If some datagrams are fragments, it must further determine when it
has received the last fragment and how the fragments it has received should be
pieced back together to form the original datagram. To allow the destination host to
perform these reassembly tasks, the designers of IP (version 4) put identification,
flag, and fragmentation offset fields in the IP datagram header. When a datagram is
created, the sending host stamps the datagram with an identification number as well
as source and destination addresses. Typically, the sending host increments the iden-
tification number for each datagram it sends. When a router needs to fragment a
datagram, each resulting datagram (that is, fragment) is stamped with the source
address, destination address, and identification number of the original datagram.
When the destination receives a series of datagrams from the same sending host, it
can examine the identification numbers of the datagrams to determine which of the
datagrams are actually fragments of the same larger datagram. Because IP is an
unreliable service, one or more of the fragments may never arrive at the destination.
For this reason, in order for the destination host to be absolutely sure it has received
the last fragment of the original datagram, the last fragment has a flag bit set to 0,
whereas all the other fragments have this flag bit set to 1. Also, in order for the des-
tination host to determine whether a fragment is missing (and also to be able to
reassemble the fragments in their proper order), the offset field is used to specify
where the fragment fits within the original IP datagram.
Figure 4.14 illustrates an example. A datagram of 4,000 bytes (20 bytes of IP
header plus 3,980 bytes of IP payload) arrives at a router and must be forwarded to
a link with an MTU of 1,500 bytes. This implies that the 3,980 data bytes in the
original datagram must be allocated to three separate fragments (each of which is
also an IP datagram). Suppose that the original datagram is stamped with an identi-
fication number of 777. The characteristics of the three fragments are shown in
Table 4.2. The values in Table 4.2 reflect the requirement that the amount of origi-
nal payload data in all but the last fragment be a multiple of 8 bytes, and that the off-
set value be specified in units of 8-byte chunks.
336 CHAPTER 4 THE NETWORK LAYER
4.4 THE INTERNET PROTOCOL (IP) 337
Fragmentation:
In: one large datagram (4,000 bytes)
Out: 3 smaller datagrams
Reassembly:
In: 3 smaller datagrams
Out: one large datagram (4,000 bytes)
Link MTU: 1,500 bytes
Figure 4.14 IP fragmentation and reassembly
Table 4.2 IP fragments
Fragment Bytes ID Offset Flag
1st fragment 1,480 bytes in
the data field of
the IP datagram
identification 777 offset 0 (meaning the data
should be inserted beginning
at byte 0)
2nd fragment 1,480 bytes
of data
identification 777 offset 185 (meaning the data
should be inserted beginning at byte
1,480. Note that 185 8 1,480)
3rd fragment 1,020 bytes
( 3,9801,4801,480)
of data
identification 777 offset 370 (meaning the data
should be inserted beginning at byte
2,960. Note that 370 8 2,960)
flag 1 (meaning
there is more)
flag 1 (meaning
there is more)
flag 0 (meaning this
is the last fragment)
At the destination, the payload of the datagram is passed to the transport layer
only after the IP layer has fully reconstructed the original IP datagram. If one or
more of the fragments does not arrive at the destination, the incomplete datagram is
discarded and not passed to the transport layer. But, as we learned in the previous
chapter, if TCP is being used at the transport layer, then TCP will recover from this
loss by having the source retransmit the data in the original datagram.
We have just learned that IP fragmentation plays an important role in gluing
together the many disparate link-layer technologies. But fragmentation also has its
costs. First, it complicates routers and end systems, which need to be designed to
accommodate datagram fragmentation and reassembly. Second, fragmentation can
be used to create lethal DoS attacks, whereby the attacker sends a series of bizarre
and unexpected fragments. A classic example is the Jolt2 attack, where the attacker
sends a stream of small fragments to the target host, none of which has an offset of
zero. The target can collapse as it attempts to rebuild datagrams out of the degener-
ate packets. Another class of exploits sends overlapping IP fragments, that is, frag-
ments whose offset values are set so that the fragments do not align properly.
Vulnerable operating systems, not knowing what to do with overlapping fragments,
can crash [Skoudis 2006]. As well see at the end of this section, a new version of
the IP protocol, IPv6, does away with fragmentation altogether, thereby streamlin-
ing IP packet processing and making IP less vulnerable to attack.
At this books Web site, we provide a Java applet that generates fragments. You
provide the incoming datagram size, the MTU, and the incoming datagram identifi-
cation. The applet automatically generates the fragments for you. See http://
www.awl.com/kurose-ross.
4.4.2 IPv4 Addressing
We now turn our attention to IPv4 addressing. Although you may be thinking that
addressing must be a straightforward topic, hopefully by the end of this chapter
youll be convinced that Internet addressing is not only a juicy, subtle, and interest-
ing topic but also one that is of central importance to the Internet. Excellent treat-
ments of IPv4 addressing are [3Com Addressing 2012] and the first chapter in
[Stewart 1999].
Before discussing IP addressing, however, well need to say a few words about
how hosts and routers are connected into the network. A host typically has only a
single link into the network; when IP in the host wants to send a datagram, it does
so over this link. The boundary between the host and the physical link is called an
interface. Now consider a router and its interfaces. Because a routers job is to
receive a datagram on one link and forward the datagram on some other link, a
router necessarily has two or more links to which it is connected. The boundary
between the router and any one of its links is also called an interface. A router thus
has multiple interfaces, one for each of its links. Because every host and router is
capable of sending and receiving IP datagrams, IP requires each host and router
interface to have its own IP address. Thus, an IP address is technically associated
with an interface, rather than with the host or router containing that interface.
Each IP address is 32 bits long (equivalently, 4 bytes), and there are thus a total
of 232 possible IP addresses. By approximating 210 by 103, it is easy to see that there
338 CHAPTER 4 THE NETWORK LAYER
http://www.awl.com/kurose-ross
http://www.awl.com/kurose-ross
are about 4 billion possible IP addresses. These addresses are typically written in
so-called dotted-decimal notation, in which each byte of the address is written in
its decimal form and is separated by a period (dot) from other bytes in the address.
For example, consider the IP address 193.32.216.9. The 193 is the decimal equiv-
alent of the first 8 bits of the address; the 32 is the decimal equivalent of the second
8 bits of the address, and so on. Thus, the address 193.32.216.9 in binary notation is
11000001 00100000 11011000 00001001
Each interface on every host and router in the global Internet must have an IP
address that is globally unique (except for interfaces behind NATs, as discussed at
the end of this section). These addresses cannot be chosen in a willy-nilly manner,
however. A portion of an interfaces IP address will be determined by the subnet to
which it is connected.
Figure 4.15 provides an example of IP addressing and interfaces. In this figure,
one router (with three interfaces) is used to interconnect seven hosts. Take a close look
at the IP addresses assigned to the host and router interfaces, as there are several things
to notice. The three hosts in the upper-left portion of Figure 4.15, and the router inter-
face to which they are connected, all have an IP address of the form 223.1.1.xxx. That
is, they all have the same leftmost 24 bits in their IP address. The four interfaces are
also interconnected to each other by a network that contains no routers. This network
4.4 THE INTERNET PROTOCOL (IP) 339
223.1.1.1
223.1.2.1
223.1.2.2
223.1.1.2
223.1.1.4 223.1.2.9
223.1.3.27
223.1.1.3
223.1.3.1 223.1.3.2
Figure 4.15 Interface addresses and subnets
could be interconnected by an Ethernet LAN, in which case the interfaces would be
interconnected by an Ethernet switch (as well discuss in Chapter 5), or by a wireless
access point (as well discuss in Chapter 6). Well represent this routerless network
connecting these hosts as a cloud for now, and dive into the internals of such networks
in Chapters 5 and 6.
In IP terms, this network interconnecting three host interfaces and one router
interface forms a subnet [RFC 950]. (A subnet is also called an IP network or
simply a network in the Internet literature.) IP addressing assigns an address to this
subnet: 223.1.1.0/24, where the /24 notation, sometimes known as a subnet mask,
indicates that the leftmost 24 bits of the 32-bit quantity define the subnet
address. The subnet 223.1.1.0/24 thus consists of the three host interfaces
(223.1.1.1, 223.1.1.2, and 223.1.1.3) and one router interface (223.1.1.4). Any addi-
tional hosts attached to the 223.1.1.0/24 subnet would be required to have an
address of the form 223.1.1.xxx. There are two additional subnets shown in Figure
4.15: the 223.1.2.0/24 network and the 223.1.3.0/24 subnet. Figure 4.16 illustrates
the three IP subnets present in Figure 4.15.
The IP definition of a subnet is not restricted to Ethernet segments that connect
multiple hosts to a router interface. To get some insight here, consider Figure 4.17,
which shows three routers that are interconnected with each other by point-to-point
links. Each router has three interfaces, one for each point-to-point link and one for
the broadcast link that directly connects the router to a pair of hosts. What subnets
are present here? Three subnets, 223.1.1.0/24, 223.1.2.0/24, and 223.1.3.0/24, are
similar to the subnets we encountered in Figure 4.15. But note that there are three
340 CHAPTER 4 THE NETWORK LAYER
223.1.1.0/23
223.1.2.0/23
223.1.3.0/23
Figure 4.16 Subnet addresses
additional subnets in this example as well: one subnet, 223.1.9.0/24, for the interfaces
that connect routers R1 and R2; another subnet, 223.1.8.0/24, for the interfaces that
connect routers R2 and R3; and a third subnet, 223.1.7.0/24, for the interfaces that
connect routers R3 and R1. For a general interconnected system of routers and hosts,
we can use the following recipe to define the subnets in the system:
To determine the subnets, detach each interface from its host or router, creating
islands of isolated networks, with interfaces terminating the end points of the
isolated networks. Each of these isolated networks is called a subnet.
If we apply this procedure to the interconnected system in Figure 4.17, we get six
islands or subnets.
From the discussion above, its clear that an organization (such as a company
or academic institution) with multiple Ethernet segments and point-to-point links
will have multiple subnets, with all of the devices on a given subnet having the same
subnet address. In principle, the different subnets could have quite different subnet
addresses. In practice, however, their subnet addresses often have much in common.
To understand why, lets next turn our attention to how addressing is handled in the
global Internet.
4.4 THE INTERNET PROTOCOL (IP) 341
223.1.8.1 223.1.8.0
223.1.9.1 223.1.7.1
223.1.2.6
223.1.2.1 223.1.2.2 223.1.3.1 223.1.3.2
223.1.1.3
223.1.7.0223.1.9.2
223.1.3.27
223.1.1.1 223.1.1.4
R1
R2 R3
Figure 4.17 Three routers interconnecting six subnets
The Internets address assignment strategy is known as Classless Interdomain
Routing (CIDRpronounced cider) [RFC 4632]. CIDR generalizes the notion of
subnet addressing. As with subnet addressing, the 32-bit IP address is divided into
two parts and again has the dotted-decimal form a.b.c.d/x, where x indicates the
number of bits in the first part of the address.
The x most significant bits of an address of the form a.b.c.d/x constitute the
network portion of the IP address, and are often referred to as the prefix (or net-
work prefix) of the address. An organization is typically assigned a block of con-
tiguous addresses, that is, a range of addresses with a common prefix (see the
Principles in Practice sidebar). In this case, the IP addresses of devices within the
organization will share the common prefix. When we cover the Internets BGP
342 CHAPTER 4 THE NETWORK LAYER
This example of an ISP that connects eight organizations to the Internet nicely illustrates
how carefully allocated CIDRized addresses facilitate routing. Suppose, as shown in Figure
4.18, that the ISP (which well call Fly-By-Night-ISP) advertises to the outside world that it
should be sent any datagrams whose first 20 address bits match 200.23.16.0/20. The
rest of the world need not know that within the address block 200.23.16.0/20 there are
in fact eight other organizations, each with its own subnets. This ability to use a single pre-
fix to advertise multiple networks is often referred to as address aggregation (also
route aggregation or route summarization).
Address aggregation works extremely well when addresses are allocated in blocks to
ISPs and then from ISPs to client organizations. But what happens when addresses are
not allocated in such a hierarchical manner? What would happen, for example, if Fly-By-
Night-ISP acquires ISPs-R-Us and then has Organization 1 connect to the Internet through
its subsidiary ISPs-R-Us? As shown in Figure 4.18, the subsidiary ISPs-R-Us owns the
address block 199.31.0.0/16, but Organization 1s IP addresses are unfortunately out-
side of this address block. What should be done here? Certainly, Organization 1 could
renumber all of its routers and hosts to have addresses within the ISPs-R-Us address
block. But this is a costly solution, and Organization 1 might well be reassigned to
another subsidiary in the future. The solution typically adopted is for Organization 1
to keep its IP addresses in 200.23.18.0/23. In this case, as shown in Figure 4.19,
Fly-By-Night-ISP continues to advertise the address block 200.23.16.0/20 and ISPs-R-Us
continues to advertise 199.31.0.0/16. However, ISPs-R-Us now also advertises the block
of addresses for Organization 1, 200.23.18.0/23. When other routers in the larger
Internet see the address blocks 200.23.16.0/20 (from Fly-By-Night-ISP) and
200.23.18.0/23 (from ISPs-R-Us) and want to route to an address in the block
200.23.18.0/23, they will use longest prefix matching (see Section 4.2.2), and route
toward ISPs-R-Us, as it advertises the longest (most specific) address prefix that matches
the destination address.
PRINCIPLES IN PRACTICE
4.4 THE INTERNET PROTOCOL (IP) 343
Organization 0
200.23.16.0/23
Organization 1
Fly-By-Night-ISP
Send me anything
with addresses
beginning
200.23.16.0/20
ISPs-R-Us
200.23.18.0/23
Organization 2
200.23.20.0/23
Organization 7
200.23.30.0/23
Internet
Send me anything
with addresses
beginning
199.31.0.0/16
Figure 4.18 Hierarchical addressing and route aggregation
Organization 0
200.23.16.0/23
Organization 2
Fly-By-Night-ISP
Send me anything
with addresses
beginning
200.23.16.0/20
ISPs-R-Us
200.23.20.0/23
Organization 7
200.23.30.0/23
Organization 1
200.23.18.0/23
Internet
Send me anything
with addresses
beginning
199.31.0.0/16 or
200.23.18.0/23
Figure 4.19 ISPs-R-Us has a more specific route to Organization 1
routing protocol in Section 4.6, well see that only these x leading prefix bits are
considered by routers outside the organizations network. That is, when a router
outside the organization forwards a datagram whose destination address is inside
the organization, only the leading x bits of the address need be considered. This
considerably reduces the size of the forwarding table in these routers, since a sin-
gle entry of the form a.b.c.d/x will be sufficient to forward packets to any destina-
tion within the organization.
The remaining 32-x bits of an address can be thought of as distinguishing
among the devices within the organization, all of which have the same network pre-
fix. These are the bits that will be considered when forwarding packets at routers
within the organization. These lower-order bits may (or may not) have an additional
subnetting structure, such as that discussed above. For example, suppose the first 21
bits of the CIDRized address a.b.c.d/21 specify the organizations network prefix
and are common to the IP addresses of all devices in that organization. The remain-
ing 11 bits then identify the specific hosts in the organization. The organizations
internal structure might be such that these 11 rightmost bits are used for subnetting
within the organization, as discussed above. For example, a.b.c.d/24 might refer to a
specific subnet within the organization.
Before CIDR was adopted, the network portions of an IP address were con-
strained to be 8, 16, or 24 bits in length, an addressing scheme known as classful
addressing, since subnets with 8-, 16-, and 24-bit subnet addresses were known as
class A, B, and C networks, respectively. The requirement that the subnet portion of
an IP address be exactly 1, 2, or 3 bytes long turned out to be problematic for sup-
porting the rapidly growing number of organizations with small and medium-sized
subnets. A class C (/24) subnet could accommodate only up to 28 2 = 254 hosts
(two of the 28 = 256 addresses are reserved for special use)too small for many
organizations. However, a class B (/16) subnet, which supports up to 65,634 hosts,
was too large. Under classful addressing, an organization with, say, 2,000 hosts was
typically allocated a class B (/16) subnet address. This led to a rapid depletion of the
class B address space and poor utilization of the assigned address space. For exam-
ple, the organization that used a class B address for its 2,000 hosts was allocated
enough of the address space for up to 65,534 interfacesleaving more than 63,000
addresses that could not be used by other organizations.
We would be remiss if we did not mention yet another type of IP address, the IP
broadcast address 255.255.255.255. When a host sends a datagram with destination
address 255.255.255.255, the message is delivered to all hosts on the same subnet.
Routers optionally forward the message into neighboring subnets as well (although
they usually dont).
Having now studied IP addressing in detail, we need to know how hosts and
subnets get their addresses in the first place. Lets begin by looking at how an
organization gets a block of addresses for its devices, and then look at how a device
(such as a host) is assigned an address from within the organizations block of
addresses.
344 CHAPTER 4 THE NETWORK LAYER
Obtaining a Block of Addresses
In order to obtain a block of IP addresses for use within an organizations subnet, a
network administrator might first contact its ISP, which would provide addresses
from a larger block of addresses that had already been allocated to the ISP. For
example, the ISP may itself have been allocated the address block 200.23.16.0/20.
The ISP, in turn, could divide its address block into eight equal-sized contiguous
address blocks and give one of these address blocks out to each of up to eight organ-
izations that are supported by this ISP, as shown below. (We have underlined the
subnet part of these addresses for your convenience.)
ISPs block 200.23.16.0/20 11001000000101110001000000000000
Organization 0 200.23.16.0/23 11001000000101110001000000000000
Organization 1 200.23.18.0/23 11001000000101110001001000000000
Organization 2 200.23.20.0/23 11001000000101110001010000000000
. . . . . . . . .
Organization 7 200.23.30.0/23 11001000000101110001111000000000
While obtaining a set of addresses from an ISP is one way to get a block of
addresses, it is not the only way. Clearly, there must also be a way for the ISP itself
to get a block of addresses. Is there a global authority that has ultimate responsibility
for managing the IP address space and allocating address blocks to ISPs and other
organizations? Indeed there is! IP addresses are managed under the authority of the
Internet Corporation for Assigned Names and Numbers (ICANN) [ICANN 2012],
based on guidelines set forth in [RFC 2050]. The role of the nonprofit ICANN organ-
ization [NTIA 1998] is not only to allocate IP addresses, but also to manage the DNS
root servers. It also has the very contentious job of assigning domain names and
resolving domain name disputes. The ICANN allocates addresses to regional Inter-
net registries (for example, ARIN, RIPE, APNIC, and LACNIC, which together
form the Address Supporting Organization of ICANN [ASO-ICANN 2012]), and
handle the allocation/management of addresses within their regions.
Obtaining a Host Address: the Dynamic Host Configuration Protocol
Once an organization has obtained a block of addresses, it can assign individual IP
addresses to the host and router interfaces in its organization. A system administra-
tor will typically manually configure the IP addresses into the router (often
remotely, with a network management tool). Host addresses can also be configured
manually, but more often this task is now done using the Dynamic Host Configu-
ration Protocol (DHCP) [RFC 2131]. DHCP allows a host to obtain (be allocated)
an IP address automatically. A network administrator can configure DHCP so that a
4.4 THE INTERNET PROTOCOL (IP) 345
given host receives the same IP address each time it connects to the network, or a
host may be assigned a temporary IP address that will be different each time the
host connects to the network. In addition to host IP address assignment, DHCP also
allows a host to learn additional information, such as its subnet mask, the address of
its first-hop router (often called the default gateway), and the address of its local
DNS server.
Because of DHCPs ability to automate the network-related aspects of connect-
ing a host into a network, it is often referred to as a plug-and-play protocol. This
capability makes it very attractive to the network administrator who would other-
wise have to perform these tasks manually! DHCP is also enjoying widespread use
in residential Internet access networks and in wireless LANs, where hosts join and
leave the network frequently. Consider, for example, the student who carries a lap-
top from a dormitory room to a library to a classroom. It is likely that in each loca-
tion, the student will be connecting into a new subnet and hence will need a new IP
address at each location. DHCP is ideally suited to this situation, as there are many
users coming and going, and addresses are needed for only a limited amount of time.
DHCP is similarly useful in residential ISP access networks. Consider, for example,
a residential ISP that has 2,000 customers, but no more than 400 customers are ever
online at the same time. In this case, rather than needing a block of 2,048 addresses,
a DHCP server that assigns addresses dynamically needs only a block of 512
addresses (for example, a block of the form a.b.c.d/23). As the hosts join and leave,
the DHCP server needs to update its list of available IP addresses. Each time a host
joins, the DHCP server allocates an arbitrary address from its current pool of avail-
able addresses; each time a host leaves, its address is returned to the pool.
DHCP is a client-server protocol. A client is typically a newly arriving host
wanting to obtain network configuration information, including an IP address for
itself. In the simplest case, each subnet (in the addressing sense of Figure 4.17) will
have a DHCP server. If no server is present on the subnet, a DHCP relay agent (typ-
ically a router) that knows the address of a DHCP server for that network is needed.
Figure 4.20 shows a DHCP server attached to subnet 223.1.2/24, with the router
serving as the relay agent for arriving clients attached to subnets 223.1.1/24 and
223.1.3/24. In our discussion below, well assume that a DHCP server is available
on the subnet.
For a newly arriving host, the DHCP protocol is a four-step process, as shown
in Figure 4.21 for the network setting shown in Figure 4.20. In this figure, yiaddr
(as in your Internet address) indicates the address being allocated to the newly
arriving client. The four steps are:
DHCP server discovery. The first task of a newly arriving host is to find a DHCP
server with which to interact. This is done using a DHCP discover message,
which a client sends within a UDP packet to port 67. The UDP packet is encap-
sulated in an IP datagram. But to whom should this datagram be sent? The host
doesnt even know the IP address of the network to which it is attaching, much
346 CHAPTER 4 THE NETWORK LAYER
less the address of a DHCP server for this network. Given this, the DHCP client
creates an IP datagram containing its DHCP discover message along with the
broadcast destination IP address of 255.255.255.255 and a this host source IP
address of 0.0.0.0. The DHCP client passes the IP datagram to the link layer,
which then broadcasts this frame to all nodes attached to the subnet (we will
cover the details of link-layer broadcasting in Section 5.4).
DHCP server offer(s). A DHCP server receiving a DHCP discover message
responds to the client with a DHCP offer message that is broadcast to all nodes
on the subnet, again using the IP broadcast address of 255.255.255.255. (You
might want to think about why this server reply must also be broadcast). Since
several DHCP servers can be present on the subnet, the client may find itself in
the enviable position of being able to choose from among several offers. Each
server offer message contains the transaction ID of the received discover mes-
sage, the proposed IP address for the client, the network mask, and an IP address
lease timethe amount of time for which the IP address will be valid. It is com-
mon for the server to set the lease time to several hours or days [Droms 2002].
4.4 THE INTERNET PROTOCOL (IP) 347
223.1.1.1
223.1.1.2
223.1.1.4 223.1.2.9
223.1.3.27
223.1.1.3
223.1.3.1 223.1.3.2
223.1.2.1
223.1.2.5
223.1.2.2
Arriving
DHCP
client
DHCP
server
Figure 4.20 DHCP client-server scenario
DHCP request. The newly arriving client will choose from among one or more
server offers and respond to its selected offer with a DHCP request message,
echoing back the configuration parameters.
DHCP ACK. The server responds to the DHCP request message with a DHCP
ACK message, confirming the requested parameters.
Once the client receives the DHCP ACK, the interaction is complete and the
client can use the DHCP-allocated IP address for the lease duration. Since a client
348 CHAPTER 4 THE NETWORK LAYER
DHCP server:
223.1.2.5
Arriving client
DHCP discover
Time Time
src: 0.0.0.0, 68
dest: 255.255.255.255,67
DHCPDISCOVER
yiaddr: 0.0.0.0
transaction ID: 654
src: 223.1.2.5, 67
dest: 255.255.255.255,68
DHCPOFFER
yiaddrr: 223.1.2.4
transaction ID: 654
DHCP server ID: 223.1.2.5
Lifetime: 3600 secs
DHCP offer
src: 223.1.2.5, 67
dest: 255.255.255.255,68
DHCPACK
yiaddrr: 223.1.2.4
transaction ID: 655
DHCP server ID: 223.1.2.5
Lifetime: 3600 secs
DHCP ACK
src: 0.0.0.0, 68
dest: 255.255.255.255, 67
DHCPREQUEST
yiaddrr: 223.1.2.4
transaction ID: 655
DHCP server ID: 223.1.2.5
Lifetime: 3600 secs
DHCP request
Figure 4.21 DHCP client-server interaction
may want to use its address beyond the leases expiration, DHCP also provides a
mechanism that allows a client to renew its lease on an IP address.
The value of DHCPs plug-and-play capability is clear, considering the fact that
the alternative is to manually configure a hosts IP address. Consider the student
who moves from classroom to library to dorm room with a laptop, joins a new sub-
net, and thus obtains a new IP address at each location. It is unimaginable that a sys-
tem administrator would have to reconfigure laptops at each location, and few
students (except those taking a computer networking class!) would have the expert-
ise to configure their laptops manually. From a mobility aspect, however, DHCP
does have shortcomings. Since a new IP address is obtained from DHCP each time
a node connects to a new subnet, a TCP connection to a remote application cannot
be maintained as a mobile node moves between subnets. In Chapter 6, we will
examine mobile IPa recent extension to the IP infrastructure that allows a mobile
node to use a single permanent address as it moves between subnets. Additional
details about DHCP can be found in [Droms 2002] and [dhc 2012]. An open source
reference implementation of DHCP is available from the Internet Systems Consor-
tium [ISC 2012].
Network Address Translation (NAT)
Given our discussion about Internet addresses and the IPv4 datagram format, were
now well aware that every IP-capable device needs an IP address. With the prolifer-
ation of small office, home office (SOHO) subnets, this would seem to imply that
whenever a SOHO wants to install a LAN to connect multiple machines, a range of
addresses would need to be allocated by the ISP to cover all of the SOHOs
machines. If the subnet grew bigger (for example, the kids at home have not only
their own computers, but have smartphones and networked Game Boys as well), a
larger block of addresses would have to be allocated. But what if the ISP had already
allocated the contiguous portions of the SOHO networks current address range?
And what typical homeowner wants (or should need) to know how to manage IP
addresses in the first place? Fortunately, there is a simpler approach to address allo-
cation that has found increasingly widespread use in such scenarios: network
address translation (NAT) [RFC 2663; RFC 3022; Zhang 2007].
Figure 4.22 shows the operation of a NAT-enabled router. The NAT-enabled
router, residing in the home, has an interface that is part of the home network on the
right of Figure 4.22. Addressing within the home network is exactly as we have seen
aboveall four interfaces in the home network have the same subnet address of
10.0.0/24. The address space 10.0.0.0/8 is one of three portions of the IP address
space that is reserved in [RFC 1918] for a private network or a realm with private
addresses, such as the home network in Figure 4.22. A realm with private addresses
refers to a network whose addresses only have meaning to devices within that
network. To see why this is important, consider the fact that there are hundreds of
4.4 THE INTERNET PROTOCOL (IP) 349
thousands of home networks, many using the same address space, 10.0.0.0/24.
Devices within a given home network can send packets to each other using
10.0.0.0/24 addressing. However, packets forwarded beyond the home network into
the larger global Internet clearly cannot use these addresses (as either a source or a
destination address) because there are hundreds of thousands of networks using this
block of addresses. That is, the 10.0.0.0/24 addresses can only have meaning within
the given home network. But if private addresses only have meaning within a given
network, how is addressing handled when packets are sent to or received from the
global Internet, where addresses are necessarily unique? The answer lies in under-
standing NAT.
The NAT-enabled router does not look like a router to the outside world. Instead
the NAT router behaves to the outside world as a single device with a single IP
address. In Figure 4.22, all traffic leaving the home router for the larger Internet has
a source IP address of 138.76.29.7, and all traffic entering the home router must
have a destination address of 138.76.29.7. In essence, the NAT-enabled router is hid-
ing the details of the home network from the outside world. (As an aside, you might
wonder where the home network computers get their addresses and where the router
gets its single IP address. Often, the answer is the sameDHCP! The router gets its
address from the ISPs DHCP server, and the router runs a DHCP server to provide
addresses to computers within the NAT-DHCP-router-controlled home networks
address space.)
350 CHAPTER 4 THE NETWORK LAYER
3
2
10.0.0.1
138.76.29.7
10.0.0.4 10.0.0.2
10.0.0.3
NAT translation table
WAN side
138.76.29.7, 5001
LAN side
10.0.0.1, 3345
. . . . . .
S = 138.76.29.7, 5001
D = 128.119.40.186, 80
1
4
S = 128.119.40.186, 80
D = 138.76.29.7, 5001
S = 128.119.40.186, 80
D = 10.0.0.1, 3345
S = 10.0.0.1, 3345
D = 128.119.40.186, 80
Figure 4.22 Network address translation
If all datagrams arriving at the NAT router from the WAN have the same desti-
nation IP address (specifically, that of the WAN-side interface of the NAT router),
then how does the router know the internal host to which it should forward a given
datagram? The trick is to use a NAT translation table at the NAT router, and to
include port numbers as well as IP addresses in the table entries.
Consider the example in Figure 4.22. Suppose a user sitting in a home network
behind host 10.0.0.1 requests a Web page on some Web server (port 80) with IP
address 128.119.40.186. The host 10.0.0.1 assigns the (arbitrary) source port num-
ber 3345 and sends the datagram into the LAN. The NAT router receives the data-
gram, generates a new source port number 5001 for the datagram, replaces the
source IP address with its WAN-side IP address 138.76.29.7, and replaces the origi-
nal source port number 3345 with the new source port number 5001. When generat-
ing a new source port number, the NAT router can select any source port number
that is not currently in the NAT translation table. (Note that because a port number
field is 16 bits long, the NAT protocol can support over 60,000 simultaneous con-
nections with a single WAN-side IP address for the router!) NAT in the router also
adds an entry to its NAT translation table. The Web server, blissfully unaware that
the arriving datagram containing the HTTP request has been manipulated by the
NAT router, responds with a datagram whose destination address is the IP address
of the NAT router, and whose destination port number is 5001. When this datagram
arrives at the NAT router, the router indexes the NAT translation table using the des-
tination IP address and destination port number to obtain the appropriate IP address
(10.0.0.1) and destination port number (3345) for the browser in the home network.
The router then rewrites the datagrams destination address and destination port
number, and forwards the datagram into the home network.
NAT has enjoyed widespread deployment in recent years. But we should
mention that many purists in the IETF community loudly object to NAT. First,
they argue, port numbers are meant to be used for addressing processes, not for
addressing hosts. (This violation can indeed cause problems for servers running
on the home network, since, as we have seen in Chapter 2, server processes wait
for incoming requests at well-known port numbers.) Second, they argue, routers
are supposed to process packets only up to layer 3. Third, they argue, the NAT
protocol violates the so-called end-to-end argument; that is, hosts should be talk-
ing directly with each other, without interfering nodes modifying IP addresses and
port numbers. And fourth, they argue, we should use IPv6 (see Section 4.4.4) to
solve the shortage of IP addresses, rather than recklessly patching up the problem
with a stopgap solution like NAT. But like it or not, NAT has become an important
component of the Internet.
Yet another major problem with NAT is that it interferes with P2P applications,
including P2P file-sharing applications and P2P Voice-over-IP applications. Recall
from Chapter 2 that in a P2P application, any participating Peer A should be able to
initiate a TCP connection to any other participating Peer B. The essence of the
problem is that if Peer B is behind a NAT, it cannot act as a server and accept TCP
4.4 THE INTERNET PROTOCOL (IP) 351
connections. As well see in the homework problems, this NAT problem can be cir-
cumvented if Peer A is not behind a NAT. In this case, Peer A can first contact Peer
B through an intermediate Peer C, which is not behind a NAT and to which B has
established an ongoing TCP connection. Peer A can then ask Peer B, via Peer C, to
initiate a TCP connection directly back to Peer A. Once the direct P2P TCP connec-
tion is established between Peers A and B, the two peers can exchange messages or
files. This hack, called connection reversal, is actually used by many P2P applica-
tions for NAT traversal. If both Peer A and Peer B are behind their own NATs, the
situation is a bit trickier but can be handled using application relays, as we saw with
Skype relays in Chapter 2.
UPnP
NAT traversal is increasingly provided by Universal Plug and Play (UPnP), which is
a protocol that allows a host to discover and configure a nearby NAT [UPnP Forum
2012]. UPnP requires that both the host and the NAT be UPnP compatible. With
UPnP, an application running in a host can request a NAT mapping between its
(private IP address, private port number) and the (public IP address, public port
number) for some requested public port number. If the NAT accepts the request and
creates the mapping, then nodes from the outside can initiate TCP connections to
(public IP address, public port number). Furthermore, UPnP lets the application
know the value of (public IP address, public port number), so that the application
can advertise it to the outside world.
As an example, suppose your host, behind a UPnP-enabled NAT, has private
address 10.0.0.1 and is running BitTorrent on port 3345. Also suppose that the
public IP address of the NAT is 138.76.29.7. Your BitTorrent application naturally
wants to be able to accept connections from other hosts, so that it can trade chunks
with them. To this end, the BitTorrent application in your host asks the NAT to cre-
ate a hole that maps (10.0.0.1, 3345) to (138.76.29.7, 5001). (The public port
number 5001 is chosen by the application.) The BitTorrent application in your host
could also advertise to its tracker that it is available at (138.76.29.7, 5001). In this
manner, an external host running BitTorrent can contact the tracker and learn that
your BitTorrent application is running at (138.76.29.7, 5001). The external host
can send a TCP SYN packet to (138.76.29.7, 5001). When the NAT receives the
SYN packet, it will change the destination IP address and port number in the
packet to (10.0.0.1, 3345) and forward the packet through the NAT.
In summary, UPnP allows external hosts to initiate communication sessions
to NATed hosts, using either TCP or UDP. NATs have long been a nemesis
for P2P applications; UPnP, providing an effective and robust NAT traversal
solution, may be their savior. Our discussion of NAT and UPnP here has been
necessarily brief. For more detailed discussions of NAT see [Huston 2004, Cisco
NAT 2012].
352 CHAPTER 4 THE NETWORK LAYER
4.4.3 Internet Control Message Protocol (ICMP)
Recall that the network layer of the Internet has three main components: the IP pro-
tocol, discussed in the previous section; the Internet routing protocols (including
RIP, OSPF, and BGP), which are covered in Section 4.6; and ICMP, which is the
subject of this section.
ICMP, specified in [RFC 792], is used by hosts and routers to communicate net-
work-layer information to each other. The most typical use of ICMP is for error
reporting. For example, when running a Telnet, FTP, or HTTP session, you may
have encountered an error message such as Destination network unreachable. This
message had its origins in ICMP. At some point, an IP router was unable to find a
path to the host specified in your Telnet, FTP, or HTTP application. That router cre-
ated and sent a type-3 ICMP message to your host indicating the error.
ICMP is often considered part of IP but architecturally it lies just above IP, as
ICMP messages are carried inside IP datagrams. That is, ICMP messages are carried
as IP payload, just as TCP or UDP segments are carried as IP payload. Similarly,
when a host receives an IP datagram with ICMP specified as the upper-layer proto-
col, it demultiplexes the datagrams contents to ICMP, just as it would demultiplex a
datagrams content to TCP or UDP.
ICMP messages have a type and a code field, and contain the header and the
first 8 bytes of the IP datagram that caused the ICMP message to be generated in the
first place (so that the sender can determine the datagram that caused the error).
Selected ICMP message types are shown in Figure 4.23. Note that ICMP messages
are used not only for signaling error conditions.
The well-known ping program sends an ICMP type 8 code 0 message to the
specified host. The destination host, seeing the echo request, sends back a type 0
code 0 ICMP echo reply. Most TCP/IP implementations support the ping server
directly in the operating system; that is, the server is not a process. Chapter 11 of
[Stevens 1990] provides the source code for the ping client program. Note that the
client program needs to be able to instruct the operating system to generate an ICMP
message of type 8 code 0.
Another interesting ICMP message is the source quench message. This message
is seldom used in practice. Its original purpose was to perform congestion control
to allow a congested router to send an ICMP source quench message to a host to
force that host to reduce its transmission rate. We have seen in Chapter 3 that TCP
has its own congestion-control mechanism that operates at the transport layer, with-
out the use of network-layer feedback such as the ICMP source quench message.
In Chapter 1 we introduced the Traceroute program, which allows us to trace a
route from a host to any other host in the world. Interestingly, Traceroute is imple-
mented with ICMP messages. To determine the names and addresses of the routers
between source and destination, Traceroute in the source sends a series of ordinary
IP datagrams to the destination. Each of these datagrams carries a UDP segment
with an unlikely UDP port number. The first of these datagrams has a TTL of 1, the
4.4 THE INTERNET PROTOCOL (IP) 353
second of 2, the third of 3, and so on. The source also starts timers for each of the
datagrams. When the nth datagram arrives at the nth router, the nth router observes
that the TTL of the datagram has just expired. According to the rules of the IP proto-
col, the router discards the datagram and sends an ICMP warning message to the
source (type 11 code 0). This warning message includes the name of the router and
its IP address. When this ICMP message arrives back at the source, the source
obtains the round-trip time from the timer and the name and IP address of the nth
router from the ICMP message.
How does a Traceroute source know when to stop sending UDP segments?
Recall that the source increments the TTL field for each datagram it sends. Thus,
one of the datagrams will eventually make it all the way to the destination host.
Because this datagram contains a UDP segment with an unlikely port number, the
destination host sends a port unreachable ICMP message (type 3 code 3) back to the
source. When the source host receives this particular ICMP message, it knows it
does not need to send additional probe packets. (The standard Traceroute program
actually sends sets of three packets with the same TTL; thus the Traceroute output
provides three results for each TTL.)
354 CHAPTER 4 THE NETWORK LAYER
ICMP Type Code Description
0 0 echo reply (to ping)
3 0 destination network unreachable
3 1 destination host unreachable
3 2 destination protocol unreachable
3 3 destination port unreachable
3 6 destination network unknown
3 7 destination host unknown
4 0 source quench (congestion control)
8 0 echo request
9 0 router advertisement
10 0 router discovery
11 0 TTL expired
12 0 IP header bad
Figure 4.23 ICMP message types
In this manner, the source host learns the number and the identities of routers
that lie between it and the destination host and the round-trip time between the two
hosts. Note that the Traceroute client program must be able to instruct the operating
system to generate UDP datagrams with specific TTL values and must also be able to
be notified by its operating system when ICMP messages arrive. Now that you under-
stand how Traceroute works, you may want to go back and play with it some more.
4.4 THE INTERNET PROTOCOL (IP) 355
INSPECTING DATAGRAMS: FIREWALLS AND INTRUSION DETECTION
SYSTEMS
Suppose you are assigned the task of administering a home, departmental, university, or
corporate network. Attackers, knowing the IP address range of your network, can easily
send IP datagrams to addresses in your range. These datagrams can do all kinds of
devious things, including mapping your network with ping sweeps and port scans,
crashing vulnerable hosts with malformed packets, flooding servers with a deluge of
ICMP packets, and infecting hosts by including malware in the packets. As the network
administrator, what are you going to do about all those bad guys out there, each capa-
ble of sending malicious packets into your network? Two popular defense mechanisms
to malicious packet attacks are firewalls and intrusion detection systems (IDSs).
As a network administrator, you may first try installing a firewall between your
network and the Internet. (Most access routers today have firewall capability.)
Firewalls inspect the datagram and segment header fields, denying suspicious data-
grams entry into the internal network. For example, a firewall may be configured to
block all ICMP echo request packets, thereby preventing an attacker from doing a
traditional ping sweep across your IP address range. Firewalls can also block pack-
ets based on source and destination IP addresses and port numbers. Additionally,
firewalls can be configured to track TCP connections, granting entry only to data-
grams that belong to approved connections.
Additional protection can be provided with an IDS. An IDS, typically situated at the
network boundary, performs deep packet inspection, examining not only header
fields but also the payloads in the datagram (including application-layer data). An IDS
has a database of packet signatures that are known to be part of attacks. This data-
base is automatically updated as new attacks are discovered. As packets pass through
the IDS, the IDS attempts to match header fields and payloads to the signatures in its
signature database. If such a match is found, an alert is created. An intrusion preven-
tion system (IPS) is similar to an IDS, except that it actually blocks packets in addition to
creating alerts. In Chapter 8, well explore firewalls and IDSs in more detail.
Can firewalls and IDSs fully shield your network from all attacks? The answer is
clearly no, as attackers continually find new attacks for which signatures are not yet
available. But firewalls and traditional signature-based IDSs are useful in protecting
your network from known attacks.
FOCUS ON SECURITY
4.4.4 IPv6
In the early 1990s, the Internet Engineering Task Force began an effort to develop a
successor to the IPv4 protocol. A prime motivation for this effort was the realization
that the 32-bit IP address space was beginning to be used up, with new subnets and
IP nodes being attached to the Internet (and being allocated unique IP addresses) at
a breathtaking rate. To respond to this need for a large IP address space, a new IP
protocol, IPv6, was developed. The designers of IPv6 also took this opportunity to
tweak and augment other aspects of IPv4, based on the accumulated operational
experience with IPv4.
The point in time when IPv4 addresses would be completely allocated (and
hence no new networks could attach to the Internet) was the subject of considerable
debate. The estimates of the two leaders of the IETFs Address Lifetime Expecta-
tions working group were that addresses would become exhausted in 2008 and 2018,
respectively [Solensky 1996]. In February 2011, IANA allocated out the last remain-
ing pool of unassigned IPv4 addresses to a regional registry. While these registries
still have available IPv4 addresses within their pool, once these addresses are
exhausted, there are no more available address blocks that can be allocated from a
central pool [Huston 2011a]. Although the mid-1990s estimates of IPv4 address
depletion suggested that a considerable amount of time might be left until the IPv4
address space was exhausted, it was realized that considerable time would be needed
to deploy a new technology on such an extensive scale, and so the Next Generation
IP (IPng) effort [Bradner 1996; RFC 1752] was begun. The result of this effort was
the specification of IP version 6 (IPv6) [RFC 2460] which well discuss below. (An
often-asked question is what happened to IPv5? It was initially envisioned that the
ST-2 protocol would become IPv5, but ST-2 was later dropped.) Excellent sources of
information about IPv6 are [Huitema 1998, IPv6 2012].
IPv6 Datagram Format
The format of the IPv6 datagram is shown in Figure 4.24. The most important
changes introduced in IPv6 are evident in the datagram format:
Expanded addressing capabilities. IPv6 increases the size of the IP address
from 32 to 128 bits. This ensures that the world wont run out of IP addresses.
Now, every grain of sand on the planet can be IP-addressable. In addition to
unicast and multicast addresses, IPv6 has introduced a new type of address,
called an anycast address, which allows a datagram to be delivered to any
one of a group of hosts. (This feature could be used, for example, to send an
HTTP GET to the nearest of a number of mirror sites that contain a given
document.)
A streamlined 40-byte header. As discussed below, a number of IPv4 fields have
been dropped or made optional. The resulting 40-byte fixed-length header allows
356 CHAPTER 4 THE NETWORK LAYER
for faster processing of the IP datagram. A new encoding of options allows for
more flexible options processing.
Flow labeling and priority. IPv6 has an elusive definition of a flow. RFC 1752
and RFC 2460 state that this allows labeling of packets belonging to particular
flows for which the sender requests special handling, such as a nondefault quality
of service or real-time service. For example, audio and video transmission might
likely be treated as a flow. On the other hand, the more traditional applications,
such as file transfer and e-mail, might not be treated as flows. It is possible that the
traffic carried by a high-priority user (for example, someone paying for better serv-
ice for their traffic) might also be treated as a flow. What is clear, however, is that
the designers of IPv6 foresee the eventual need to be able to differentiate among
the flows, even if the exact meaning of a flow has not yet been determined. The
IPv6 header also has an 8-bit traffic class field. This field, like the TOS field in
IPv4, can be used to give priority to certain datagrams within a flow, or it can be
used to give priority to datagrams from certain applications (for example, ICMP)
over datagrams from other applications (for example, network news).
As noted above, a comparison of Figure 4.24 with Figure 4.13 reveals the sim-
pler, more streamlined structure of the IPv6 datagram. The following fields are
defined in IPv6:
Version. This 4-bit field identifies the IP version number. Not surprisingly, IPv6
carries a value of 6 in this field. Note that putting a 4 in this field does not create
a valid IPv4 datagram. (If it did, life would be a lot simplersee the discussion
below regarding the transition from IPv4 to IPv6.)
4.4 THE INTERNET PROTOCOL (IP) 357
Version Traffic class
Payload length Next hdr Hop limit
Flow label
32 bits
Source address
(128 bits)
Destination address
(128 bits)
Data
Figure 4.24 IPv6 datagram format
Traffic class. This 8-bit field is similar in spirit to the TOS field we saw in IPv4.
Flow label. As discussed above, this 20-bit field is used to identify a flow of
datagrams.
Payload length. This 16-bit value is treated as an unsigned integer giving the
number of bytes in the IPv6 datagram following the fixed-length, 40-byte data-
gram header.
Next header. This field identifies the protocol to which the contents (data field)
of this datagram will be delivered (for example, to TCP or UDP). The field uses
the same values as the protocol field in the IPv4 header.
Hop limit. The contents of this field are decremented by one by each router that
forwards the datagram. If the hop limit count reaches zero, the datagram is
discarded.
Source and destination addresses. The various formats of the IPv6 128-bit
address are described in RFC 4291.
Data. This is the payload portion of the IPv6 datagram. When the datagram
reaches its destination, the payload will be removed from the IP datagram and
passed on to the protocol specified in the next header field.
The discussion above identified the purpose of the fields that are included in the
IPv6 datagram. Comparing the IPv6 datagram format in Figure 4.24 with the IPv4
datagram format that we saw in Figure 4.13, we notice that several fields appearing
in the IPv4 datagram are no longer present in the IPv6 datagram:
Fragmentation/Reassembly. IPv6 does not allow for fragmentation and reassem-
bly at intermediate routers; these operations can be performed only by the source
and destination. If an IPv6 datagram received by a router is too large to be for-
warded over the outgoing link, the router simply drops the datagram and sends a
Packet Too Big ICMP error message (see below) back to the sender. The
sender can then resend the data, using a smaller IP datagram size. Fragmentation
and reassembly is a time-consuming operation; removing this functionality from
the routers and placing it squarely in the end systems considerably speeds up IP
forwarding within the network.
Header checksum. Because the transport-layer (for example, TCP and UDP) and
link-layer (for example, Ethernet) protocols in the Internet layers perform check-
summing, the designers of IP probably felt that this functionality was sufficiently
redundant in the network layer that it could be removed. Once again, fast pro-
cessing of IP packets was a central concern. Recall from our discussion of IPv4
in Section 4.4.1 that since the IPv4 header contains a TTL field (similar to the
hop limit field in IPv6), the IPv4 header checksum needed to be recomputed at
every router. As with fragmentation and reassembly, this too was a costly opera-
tion in IPv4.
358 CHAPTER 4 THE NETWORK LAYER
Options. An options field is no longer a part of the standard IP header. How-
ever, it has not gone away. Instead, the options field is one of the possible next
headers pointed to from within the IPv6 header. That is, just as TCP or UDP
protocol headers can be the next header within an IP packet, so too can an
options field. The removal of the options field results in a fixed-length, 40-
byte IP header.
Recall from our discussion in Section 4.4.3 that the ICMP protocol is used by IP
nodes to report error conditions and provide limited information (for example, the
echo reply to a ping message) to an end system. A new version of ICMP has been
defined for IPv6 in RFC 4443. In addition to reorganizing the existing ICMP type
and code definitions, ICMPv6 also added new types and codes required by the new
IPv6 functionality. These include the Packet Too Big type, and an unrecognized
IPv6 options error code. In addition, ICMPv6 subsumes the functionality of the
Internet Group Management Protocol (IGMP) that well study in Section 4.7. IGMP,
which is used to manage a hosts joining and leaving of multicast groups, was previ-
ously a separate protocol from ICMP in IPv4.
Transitioning from IPv4 to IPv6
Now that we have seen the technical details of IPv6, let us consider a very practical
matter: How will the public Internet, which is based on IPv4, be transitioned to
IPv6? The problem is that while new IPv6-capable systems can be made backward-
compatible, that is, can send, route, and receive IPv4 datagrams, already deployed
IPv4-capable systems are not capable of handling IPv6 datagrams. Several options
are possible [Huston 2011b].
One option would be to declare a flag daya given time and date when all
Internet machines would be turned off and upgraded from IPv4 to IPv6. The last
major technology transition (from using NCP to using TCP for reliable transport
service) occurred almost 25 years ago. Even back then [RFC 801], when the Inter-
net was tiny and still being administered by a small number of wizards, it was
realized that such a flag day was not possible. A flag day involving hundreds of mil-
lions of machines and millions of network administrators and users is even more
unthinkable today. RFC 4213 describes two approaches (which can be used either
alone or together) for gradually integrating IPv6 hosts and routers into an IPv4
world (with the long-term goal, of course, of having all IPv4 nodes eventually tran-
sition to IPv6).
Probably the most straightforward way to introduce IPv6-capable nodes is a
dual-stack approach, where IPv6 nodes also have a complete IPv4 implementation.
Such a node, referred to as an IPv6/IPv4 node in RFC 4213, has the ability to send
and receive both IPv4 and IPv6 datagrams. When interoperating with an IPv4 node,
an IPv6/IPv4 node can use IPv4 datagrams; when interoperating with an IPv6 node,
it can speak IPv6. IPv6/IPv4 nodes must have both IPv6 and IPv4 addresses. They
4.4 THE INTERNET PROTOCOL (IP) 359
must furthermore be able to determine whether another node is IPv6-capable or
IPv4-only. This problem can be solved using the DNS (see Chapter 2), which can
return an IPv6 address if the node name being resolved is IPv6-capable, or other-
wise return an IPv4 address. Of course, if the node issuing the DNS request is only
IPv4-capable, the DNS returns only an IPv4 address.
In the dual-stack approach, if either the sender or the receiver is only IPv4-
capable, an IPv4 datagram must be used. As a result, it is possible that two IPv6-
capable nodes can end up, in essence, sending IPv4 datagrams to each other. This is
illustrated in Figure 4.25. Suppose Node A is IPv6-capable and wants to send an IP
datagram to Node F, which is also IPv6-capable. Nodes A and B can exchange an
IPv6 datagram. However, Node B must create an IPv4 datagram to send to C. Cer-
tainly, the data field of the IPv6 datagram can be copied into the data field of the
IPv4 datagram and appropriate address mapping can be done. However, in perform-
ing the conversion from IPv6 to IPv4, there will be IPv6-specific fields in the IPv6
datagram (for example, the flow identifier field) that have no counterpart in IPv4.
The information in these fields will be lost. Thus, even though E and F can exchange
IPv6 datagrams, the arriving IPv4 datagrams at E from D do not contain all of the
fields that were in the original IPv6 datagram sent from A.
An alternative to the dual-stack approach, also discussed in RFC 4213, is
known as tunneling. Tunneling can solve the problem noted above, allowing, for
example, E to receive the IPv6 datagram originated by A. The basic idea behind
tunneling is the following. Suppose two IPv6 nodes (for example, B and E in Fig-
ure 4.25) want to interoperate using IPv6 datagrams but are connected to each
other by intervening IPv4 routers. We refer to the intervening set of IPv4 routers
between two IPv6 routers as a tunnel, as illustrated in Figure 4.26. With tunnel-
ing, the IPv6 node on the sending side of the tunnel (for example, B) takes the
entire IPv6 datagram and puts it in the data (payload) field of an IPv4 datagram.
360 CHAPTER 4 THE NETWORK LAYER
A B C D E F
IPv6
A to B: IPv6 B to C: IPv4 D to E: IPv4 E to F: IPv6
IPv6 IPv4 IPv4 IPv6 IPv6
Flow: X
Source: A
Dest: F
data
Source: A
Dest: F
data
Source: A
Dest: F
data
Flow: ??
Source: A
Dest: F
data
Figure 4.25 A dual-stack approach
This IPv4 datagram is then addressed to the IPv6 node on the receiving side of
the tunnel (for example, E) and sent to the first node in the tunnel (for example,
C). The intervening IPv4 routers in the tunnel route this IPv4 datagram among
themselves, just as they would any other datagram, blissfully unaware that the
IPv4 datagram itself contains a complete IPv6 datagram. The IPv6 node on the
receiving side of the tunnel eventually receives the IPv4 datagram (it is the desti-
nation of the IPv4 datagram!), determines that the IPv4 datagram contains an
IPv6 datagram, extracts the IPv6 datagram, and then routes the IPv6 datagram
exactly as it would if it had received the IPv6 datagram from a directly connected
IPv6 neighbor.
We end this section by noting that while the adoption of IPv6 was initially
slow to take off [Lawton 2001], momentum has been building recently. See [Hus-
ton 2008b] for discussion of IPv6 deployment as of 2008; see [NIST IPv6 2012]
for a snapshort of US IPv6 deployment. The proliferation of devices such as IP-
enabled phones and other portable devices provides an additional push for more
4.4 THE INTERNET PROTOCOL (IP) 361
A B C D E F
IPv6
A to B: IPv6
Physical view
B to C: IPv4
(encapsulating IPv6)
D to E: IPv4
(encapsulating IPv6)
E to F: IPv6
IPv6 IPv4 IPv4 IPv6 IPv6
Flow: X
Source: A
Dest: F
data
Source: B
Dest: E
Source: B
Dest: E
A B E F
IPv6
Logical view
IPv6
Tunnel
IPv6 IPv6
Flow: X
Source: A
Dest: F
data
Flow: X
Source: A
Dest: F
data
Flow: X
Source: A
Dest: F
data
Figure 4.26 Tunneling
widespread deployment of IPv6. Europes Third Generation Partnership Program
[3GPP 2012] has specified IPv6 as the standard addressing scheme for mobile
multimedia.
One important lesson that we can learn from the IPv6 experience is that it is enor-
mously difficult to change network-layer protocols. Since the early 1990s, numerous
new network-layer protocols have been trumpeted as the next major revolution for the
Internet, but most of these protocols have had limited penetration to date. These proto-
cols include IPv6, multicast protocols (Section 4.7), and resource reservation proto-
cols (Chapter 7). Indeed, introducing new protocols into the network layer is like
replacing the foundation of a houseit is difficult to do without tearing the whole
house down or at least temporarily relocating the houses residents. On the other hand,
the Internet has witnessed rapid deployment of new protocols at the application layer.
The classic examples, of course, are the Web, instant messaging, and P2P file sharing.
Other examples include audio and video streaming and distributed games. Introducing
new application-layer protocols is like adding a new layer of paint to a houseit is
relatively easy to do, and if you choose an attractive color, others in the neighborhood
will copy you. In summary, in the future we can expect to see changes in the Internets
network layer, but these changes will likely occur on a time scale that is much slower
than the changes that will occur at the application layer.
4.4.5 A Brief Foray into IP Security
Section 4.4.3 covered IPv4 in some detail, including the services it provides and
how those services are implemented. While reading through that section, you may
have noticed that there was no mention of any security services. Indeed, IPv4 was
designed in an era (the 1970s) when the Internet was primarily used among mutu-
ally-trusted networking researchers. Creating a computer network that integrated a
multitude of link-layer technologies was already challenging enough, without hav-
ing to worry about security.
But with security being a major concern today, Internet researchers have moved
on to design new network-layer protocols that provide a variety of security services.
One of these protocols is IPsec, one of the more popular secure network-layer proto-
cols and also widely deployed in Virtual Private Networks (VPNs). Although IPsec and
its cryptographic underpinnings are covered in some detail in Chapter 8, we provide a
brief, high-level introduction into IPsec services in this section.
IPsec has been designed to be backward compatible with IPv4 and IPv6. In par-
ticular, in order to reap the benefits of IPsec, we dont need to replace the protocol
stacks in all the routers and hosts in the Internet. For example, using the transport
mode (one of two IPsec modes), if two hosts want to securely communicate, IPsec
needs to be available only in those two hosts. All other routers and hosts can con-
tinue to run vanilla IPv4.
For concreteness, well focus on IPsecs transport mode here. In this mode, two
hosts first establish an IPsec session between themselves. (Thus IPsec is connection-
oriented!) With the session in place, all TCP and UDP segments sent between the
362 CHAPTER 4 THE NETWORK LAYER
two hosts enjoy the security services provided by IPsec. On the sending side, the
transport layer passes a segment to IPsec. IPsec then encrypts the segment, appends
additional security fields to the segment, and encapsulates the resulting payload in
an ordinary IP datagram. (Its actually a little more complicated than this, as well
see in Chapter 8.) The sending host then sends the datagram into the Internet, which
transports it to the destination host. There, IPsec decrypts the segment and passes
the unencrypted segment to the transport layer.
The services provided by an IPsec session include:
Cryptographic agreement. Mechanisms that allow the two communicating hosts
to agree on cryptographic algorithms and keys.
Encryption of IP datagram payloads. When the sending host receives a segment
from the transport layer, IPsec encrypts the payload. The payload can only be
decrypted by IPsec in the receiving host.
Data integrity. IPsec allows the receiving host to verify that the datagrams
header fields and encrypted payload were not modified while the datagram was
en route from source to destination.
Origin authentication. When a host receives an IPsec datagram from a trusted
source (with a trusted keysee Chapter 8), the host is assured that the source IP
address in the datagram is the actual source of the datagram.
When two hosts have an IPsec session established between them, all TCP and
UDP segments sent between them will be encrypted and authenticated. IPsec there-
fore provides blanket coverage, securing all communication between the two hosts
for all network applications.
A company can use IPsec to communicate securely in the nonsecure public Inter-
net. For illustrative purposes, well just look at a simple example here. Consider a
company that has a large number of traveling salespeople, each possessing a company
laptop computer. Suppose the salespeople need to frequently consult sensitive com-
pany information (for example, pricing and product information) that is stored on a
server in the companys headquarters. Further suppose that the salespeople also need
to send sensitive documents to each other. How can this be done with IPsec? As you
might guess, we install IPsec in the server and in all of the salespeoples laptops. With
IPsec installed in these hosts, whenever a salesperson needs to communicate with the
server or with another salesperson, the communication session will be secure.
4.5 Routing Algorithms
So far in this chapter, weve mostly explored the network layers forwarding func-
tion. We learned that when a packet arrives to a router, the router indexes a forward-
ing table and determines the link interface to which the packet is to be directed. We
also learned that routing algorithms, operating in network routers, exchange and
4.5 ROUTING ALGORITHMS 363
compute the information that is used to configure these forwarding tables. The inter-
play between routing algorithms and forwarding tables was shown in Figure 4.2.
Having explored forwarding in some depth we now turn our attention to the other
major topic of this chapter, namely, the network layers critical routing function.
Whether the network layer provides a datagram service (in which case different pack-
ets between a given source-destination pair may take different routes) or a VC serv-
ice (in which case all packets between a given source and destination will take the
same path), the network layer must nonetheless determine the path that packets take
from senders to receivers. Well see that the job of routing is to determine good paths
(equivalently, routes), from senders to receivers, through the network of routers.
Typically a host is attached directly to one router, the default router for the
host (also called the first-hop router for the host). Whenever a host sends a packet,
the packet is transferred to its default router. We refer to the default router of the
source host as the source router and the default router of the destination host as the
destination router. The problem of routing a packet from source host to destination
host clearly boils down to the problem of routing the packet from source router to
destination router, which is the focus of this section.
The purpose of a routing algorithm is then simple: given a set of routers, with
links connecting the routers, a routing algorithm finds a good path from source
router to destination router. Typically, a good path is one that has the least cost.
Well see, however, that in practice, real-world concerns such as policy issues (for
example, a rule such as router x, belonging to organization Y, should not forward
any packets originating from the network owned by organization Z) also come into
play to complicate the conceptually simple and elegant algorithms whose theory
underlies the practice of routing in todays networks.
A graph is used to formulate routing problems. Recall that a graph G = (N,E)
is a set N of nodes and a collection E of edges, where each edge is a pair of nodes
from N. In the context of network-layer routing, the nodes in the graph represent
routersthe points at which packet-forwarding decisions are madeand the edges
connecting these nodes represent the physical links between these routers. Such a
graph abstraction of a computer network is shown in Figure 4.27. To view some
graphs representing real network maps, see [Dodge 2012, Cheswick 2000]; for a
discussion of how well different graph-based models model the Internet, see
[Zegura 1997, Faloutsos 1999, Li 2004].
As shown in Figure 4.27, an edge also has a value representing its cost. Typi-
cally, an edges cost may reflect the physical length of the corresponding link (for
example, a transoceanic link might have a higher cost than a short-haul terrestrial
link), the link speed, or the monetary cost associated with a link. For our purposes,
well simply take the edge costs as a given and wont worry about how they are
determined. For any edge (x,y) in E, we denote c(x,y) as the cost of the edge between
nodes x and y. If the pair (x,y) does not belong to E, we set c(x,y) = . Also, through-
out we consider only undirected graphs (i.e., graphs whose edges do not have a
direction), so that edge (x,y) is the same as edge (y,x) and that c(x,y) = c(y,x). Also, a
node y is said to be a neighbor of node x if (x,y) belongs to E.
364 CHAPTER 4 THE NETWORK LAYER
Given that costs are assigned to the various edges in the graph abstraction, a natu-
ral goal of a routing algorithm is to identify the least costly paths between sources and
destinations. To make this problem more precise, recall that a path in a graph G =
(N,E) is a sequence of nodes (x1, x2,, xp) such that each of the pairs (x1,x2),
(x2,x3),,(xp-1,xp) are edges in E. The cost of a path (x1,x2,, xp) is simply the sum of
all the edge costs along the path, that is, c(x1,x2) + c(x2,x3) + + c(xp-1,xp). Given any
two nodes x and y, there are typically many paths between the two nodes, with each
path having a cost. One or more of these paths is a least-cost path. The least-cost
problem is therefore clear: Find a path between the source and destination that has
least cost. In Figure 4.27, for example, the least-cost path between source node u and
destination node w is (u, x, y, w) with a path cost of 3. Note that if all edges in the
graph have the same cost, the least-cost path is also the shortest path (that is, the
path with the smallest number of links between the source and the destination).
As a simple exercise, try finding the least-cost path from node u to z in Figure
4.27 and reflect for a moment on how you calculated that path. If you are like most
people, you found the path from u to z by examining Figure 4.27, tracing a few routes
from u to z, and somehow convincing yourself that the path you had chosen had the
least cost among all possible paths. (Did you check all of the 17 possible paths
between u and z? Probably not!) Such a calculation is an example of a centralized
routing algorithmthe routing algorithm was run in one location, your brain, with
complete information about the network. Broadly, one way in which we can classify
routing algorithms is according to whether they are global or decentralized.
A global routing algorithm computes the least-cost path between a source and
destination using complete, global knowledge about the network. That is, the
algorithm takes the connectivity between all nodes and all link costs as inputs.
This then requires that the algorithm somehow obtain this information before
actually performing the calculation. The calculation itself can be run at one site
4.5 ROUTING ALGORITHMS 365
x y
v
3
5
2 5
2
3
1
1 2
1
u z
w
Figure 4.27 Abstract graph model of a computer network
(a centralized global routing algorithm) or replicated at multiple sites. The key
distinguishing feature here, however, is that a global algorithm has complete
information about connectivity and link costs. In practice, algorithms with global
state information are often referred to as link-state (LS) algorithms, since the
algorithm must be aware of the cost of each link in the network. Well study LS
algorithms in Section 4.5.1.
In a decentralized routing algorithm, the calculation of the least-cost path is
carried out in an iterative, distributed manner. No node has complete information
about the costs of all network links. Instead, each node begins with only the
knowledge of the costs of its own directly attached links. Then, through an itera-
tive process of calculation and exchange of information with its neighboring
nodes (that is, nodes that are at the other end of links to which it itself is
attached), a node gradually calculates the least-cost path to a destination or set of
destinations. The decentralized routing algorithm well study below in Section
4.5.2 is called a distance-vector (DV) algorithm, because each node maintains a
vector of estimates of the costs (distances) to all other nodes in the network.
A second broad way to classify routing algorithms is according to whether they
are static or dynamic. In static routing algorithms, routes change very slowly over
time, often as a result of human intervention (for example, a human manually edit-
ing a routers forwarding table). Dynamic routing algorithms change the routing
paths as the network traffic loads or topology change. A dynamic algorithm can be
run either periodically or in direct response to topology or link cost changes. While
dynamic algorithms are more responsive to network changes, they are also more
susceptible to problems such as routing loops and oscillation in routes.
A third way to classify routing algorithms is according to whether they are load-
sensitive or load-insensitive. In a load-sensitive algorithm, link costs vary dynami-
cally to reflect the current level of congestion in the underlying link. If a high cost is
associated with a link that is currently congested, a routing algorithm will tend to
choose routes around such a congested link. While early ARPAnet routing algo-
rithms were load-sensitive [McQuillan 1980], a number of difficulties were encoun-
tered [Huitema 1998]. Todays Internet routing algorithms (such as RIP, OSPF, and
BGP) are load-insensitive, as a links cost does not explicitly reflect its current (or
recent past) level of congestion.
4.5.1 The Link-State (LS) Routing Algorithm
Recall that in a link-state algorithm, the network topology and all link costs are
known, that is, available as input to the LS algorithm. In practice this is accom-
plished by having each node broadcast link-state packets to all other nodes in the
network, with each link-state packet containing the identities and costs of its
attached links. In practice (for example, with the Internets OSPF routing protocol,
discussed in Section 4.6.1) this is often accomplished by a link-state broadcast
366 CHAPTER 4 THE NETWORK LAYER
algorithm [Perlman 1999]. Well cover broadcast algorithms in Section 4.7. The
result of the nodes broadcast is that all nodes have an identical and complete view
of the network. Each node can then run the LS algorithm and compute the same set
of least-cost paths as every other node.
The link-state routing algorithm we present below is known as Dijkstras algo-
rithm, named after its inventor. A closely related algorithm is Prims algorithm; see
[Cormen 2001] for a general discussion of graph algorithms. Dijkstras algorithm
computes the least-cost path from one node (the source, which we will refer to as u)
to all other nodes in the network. Dijkstras algorithm is iterative and has the prop-
erty that after the kth iteration of the algorithm, the least-cost paths are known to k
destination nodes, and among the least-cost paths to all destination nodes, these k
paths will have the k smallest costs. Let us define the following notation:
D(v): cost of the least-cost path from the source node to destination v as of this
iteration of the algorithm.
p(v): previous node (neighbor of v) along the current least-cost path from the
source to v.
N : subset of nodes; v is in N if the least-cost path from the source to v is defin-
itively known.
The global routing algorithm consists of an initialization step followed by a
loop. The number of times the loop is executed is equal to the number of nodes in
the network. Upon termination, the algorithm will have calculated the shortest paths
from the source node u to every other node in the network.
Link-State (LS) Algorithm for Source Node u
1 Initialization:
2 N = {u}
3 for all nodes v
4 if v is a neighbor of u
5 then D(v) = c(u,v)
6 else D(v) =
7
8 Loop
9 find w not in N such that D(w) is a minimum
10 add w to N
11 update D(v) for each neighbor v of w and not in N:
12 D(v) = min( D(v), D(w) + c(w,v) )
13 /* new cost to v is either old cost to v or known
14 least path cost to w plus cost from w to v */
15 until N= N
4.5 ROUTING ALGORITHMS 367
As an example, lets consider the network in Figure 4.27 and compute the
least-cost paths from u to all possible destinations. A tabular summary of the
algorithms computation is shown in Table 4.3, where each line in the table gives
the values of the algorithms variables at the end of the iteration. Lets consider
the few first steps in detail.
In the initialization step, the currently known least-cost paths from u to its
directly attached neighbors, v, x, and w, are initialized to 2, 1, and 5, respectively.
Note in particular that the cost to w is set to 5 (even though we will soon see that
a lesser-cost path does indeed exist) since this is the cost of the direct (one hop)
link from u to w. The costs to y and z are set to infinity because they are not
directly connected to u.
In the first iteration, we look among those nodes not yet added to the set N and
find that node with the least cost as of the end of the previous iteration. That node
is x, with a cost of 1, and thus x is added to the set N. Line 12 of the LS algo-
rithm is then performed to update D(v) for all nodes v, yielding the results shown
in the second line (Step 1) in Table 4.3. The cost of the path to v is unchanged.
The cost of the path to w (which was 5 at the end of the initialization) through
node x is found to have a cost of 4. Hence this lower-cost path is selected and ws
predecessor along the shortest path from u is set to x. Similarly, the cost to y
(through x) is computed to be 2, and the table is updated accordingly.
In the second iteration, nodes v and y are found to have the least-cost paths (2),
and we break the tie arbitrarily and add y to the set N so that N now contains u,
x, and y. The cost to the remaining nodes not yet in N, that is, nodes v, w, and z,
are updated via line 12 of the LS algorithm, yielding the results shown in the
third row in the Table 4.3.
And so on. . . .
When the LS algorithm terminates, we have, for each node, its predecessor
along the least-cost path from the source node. For each predecessor, we also
368 CHAPTER 4 THE NETWORK LAYER
step N D(v),p(v) D(w),p(w) D(x),p(x) D(y),p(y) D(z),p(z)
0 u 2,u 5,u 1,u
1 ux 2,u 4,x 2,x
2 uxy 2,u 3,y 4,y
3 uxyv 3,y 4,y
4 uxyvw 4,y
5 uxyvwz
Table 4.3 Running the link-state algorithm on the network in Figure 4.27
VideoNote
Dijkstras algorithm:
discussion and example
have its predecessor, and so in this manner we can construct the entire path from
the source to all destinations. The forwarding table in a node, say node u, can
then be constructed from this information by storing, for each destination, the
next-hop node on the least-cost path from u to the destination. Figure 4.28
shows the resulting least-cost paths and forwarding table in u for the network in
Figure 4.27.
What is the computational complexity of this algorithm? That is, given n
nodes (not counting the source), how much computation must be done in the
worst case to find the least-cost paths from the source to all destinations? In the
first iteration, we need to search through all n nodes to determine the node, w, not
in N that has the minimum cost. In the second iteration, we need to check n 1
nodes to determine the minimum cost; in the third iteration n 2 nodes, and so
on. Overall, the total number of nodes we need to search through over all the iter-
ations is n(n + 1)/2, and thus we say that the preceding implementation of the LS
algorithm has worst-case complexity of order n squared: O(n2). (A more sophisti-
cated implementation of this algorithm, using a data structure known as a heap,
can find the minimum in line 9 in logarithmic rather than linear time, thus reduc-
ing the complexity.)
Before completing our discussion of the LS algorithm, let us consider a pathol-
ogy that can arise. Figure 4.29 shows a simple network topology where link costs
are equal to the load carried on the link, for example, reflecting the delay that would
be experienced. In this example, link costs are not symmetric; that is, c(u,v) equals
c(v,u) only if the load carried on both directions on the link (u,v) is the same. In this
example, node z originates a unit of traffic destined for w, node x also originates a
unit of traffic destined for w, and node y injects an amount of traffic equal to e, also
destined for w. The initial routing is shown in Figure 4.29(a) with the link costs cor-
responding to the amount of traffic carried.
When the LS algorithm is next run, node y determines (based on the link costs
shown in Figure 4.29(a)) that the clockwise path to w has a cost of 1, while the
counterclockwise path to w (which it had been using) has a cost of 1 + e. Hence ys
4.5 ROUTING ALGORITHMS 369
Destination Link
v
w
x
y
z
(u, v)
(u, x)
(u, x)
(u, x)
(u, x)X Y
V
U Z
W
Figure 4.28 Least cost path and forwarding table for nodule u
least-cost path to w is now clockwise. Similarly, x determines that its new least-cost
path to w is also clockwise, resulting in costs shown in Figure 4.29(b). When the
LS algorithm is run next, nodes x, y, and z all detect a zero-cost path to w in the
counterclockwise direction, and all route their traffic to the counterclockwise
routes. The next time the LS algorithm is run, x, y, and z all then route their traffic
to the clockwise routes.
What can be done to prevent such oscillations (which can occur in any algo-
rithm, not just an LS algorithm, that uses a congestion or delay-based link met-
ric)? One solution would be to mandate that link costs not depend on the amount
of traffic carriedan unacceptable solution since one goal of routing is to avoid
370 CHAPTER 4 THE NETWORK LAYER
w
y
z x
1
0 0
0 e
1 + e
1
a.Initial routing
1
e
w
y
z x
2 + e
1 + e 1
0 0
0
b. x, y detect better path
to w, clockwise
w
y
z x
0
0 0
1 1 + e
2+ e
c. x, y, z detect better path
to w, counterclockwise
w
y
z x
2 + e
1 + e 1
0 0
0
d. x, y, z, detect better path
to w, clockwise
1 1
e
1 1
e
1 1
e
Figure 4.29 Oscillations with congestion-sensitive routing
highly congested (for example, high-delay) links. Another solution is to ensure
that not all routers run the LS algorithm at the same time. This seems a more
reasonable solution, since we would hope that even if routers ran the LS algorithm
with the same periodicity, the execution instance of the algorithm would not be
the same at each node. Interestingly, researchers have found that routers in the
Internet can self-synchronize among themselves [Floyd Synchronization 1994].
That is, even though they initially execute the algorithm with the same period
but at different instants of time, the algorithm execution instance can eventually
become, and remain, synchronized at the routers. One way to avoid such self-
synchronization is for each router to randomize the time it sends out a link
advertisement.
Having studied the LS algorithm, lets consider the other major routing algo-
rithm that is used in practice todaythe distance-vector routing algorithm.
4.5.2 The Distance-Vector (DV) Routing Algorithm
Whereas the LS algorithm is an algorithm using global information, the distance-
vector (DV) algorithm is iterative, asynchronous, and distributed. It is distributed
in that each node receives some information from one or more of its directly
attached neighbors, performs a calculation, and then distributes the results of its
calculation back to its neighbors. It is iterative in that this process continues
on until no more information is exchanged between neighbors. (Interestingly, the
algorithm is also self-terminatingthere is no signal that the computation should
stop; it just stops.) The algorithm is asynchronous in that it does not require all of
the nodes to operate in lockstep with each other. Well see that an asynchronous,
iterative, self-terminating, distributed algorithm is much more interesting and fun
than a centralized algorithm!
Before we present the DV algorithm, it will prove beneficial to discuss an
important relationship that exists among the costs of the least-cost paths. Let dx(y)
be the cost of the least-cost path from node x to node y. Then the least costs are
related by the celebrated Bellman-Ford equation, namely,
dx(y) = minv{c(x,v) + dv(y)}, (4.1)
where the minv in the equation is taken over all of xs neighbors. The Bellman-Ford
equation is rather intuitive. Indeed, after traveling from x to v, if we then take the
least-cost path from v to y, the path cost will be c(x,v) + dv(y). Since we must begin
by traveling to some neighbor v, the least cost from x to y is the minimum of c(x,v)
+ dv(y) taken over all neighbors v.
But for those who might be skeptical about the validity of the equation, lets
check it for source node u and destination node z in Figure 4.27. The source node u
4.5 ROUTING ALGORITHMS 371
has three neighbors: nodes v, x, and w. By walking along various paths in the graph,
it is easy to see that dv(z) = 5, dx(z) = 3, and dw(z) = 3. Plugging these values into
Equation 4.1, along with the costs c(u,v) = 2, c(u,x) = 1, and c(u,w) = 5, gives du(z) =
min{2 + 5, 5 + 3, 1 + 3} = 4, which is obviously true and which is exactly what the
Dijskstra algorithm gave us for the same network. This quick verification should
help relieve any skepticism you may have.
The Bellman-Ford equation is not just an intellectual curiosity. It actually has
significant practical importance. In particular, the solution to the Bellman-Ford
equation provides the entries in node xs forwarding table. To see this, let v* be any
neighboring node that achieves the minimum in Equation 4.1. Then, if node x wants
to send a packet to node y along a least-cost path, it should first forward the packet
to node v*. Thus, node xs forwarding table would specify node v* as the next-hop
router for the ultimate destination y. Another important practical contribution of the
Bellman-Ford equation is that it suggests the form of the neighbor-to-neighbor com-
munication that will take place in the DV algorithm.
The basic idea is as follows. Each node x begins with Dx(y), an estimate of the
cost of the least-cost path from itself to node y, for all nodes in N. Let Dx = [Dx(y): y
in N] be node xs distance vector, which is the vector of cost estimates from x to all
other nodes, y, in N. With the DV algorithm, each node x maintains the following
routing information:
For each neighbor v, the cost c(x,v) from x to directly attached neighbor, v
Node xs distance vector, that is, Dx = [Dx(y): y in N], containing xs estimate of
its cost to all destinations, y, in N
The distance vectors of each of its neighbors, that is, Dv = [Dv(y): y in N] for each
neighbor v of x
In the distributed, asynchronous algorithm, from time to time, each node sends
a copy of its distance vector to each of its neighbors. When a node x receives a
new distance vector from any of its neighbors v, it saves vs distance vector, and
then uses the Bellman-Ford equation to update its own distance vector as fol-
lows:
Dx(y) minv{c(x,v) + Dv(y)} for each node y in N
If node xs distance vector has changed as a result of this update step, node x will
then send its updated distance vector to each of its neighbors, which can in turn
update their own distance vectors. Miraculously enough, as long as all the nodes
continue to exchange their distance vectors in an asynchronous fashion, each cost
estimate Dx(y) converges to dx(y), the actual cost of the least-cost path from node x
to node y [Bertsekas 1991]!
372 CHAPTER 4 THE NETWORK LAYER
Distance-Vector (DV) Algorithm
At each node, x:
4.5 ROUTING ALGORITHMS 373
1 Initialization:
2 for all destinations y in N:
3 Dx(y) = c(x,y) /* if y is not a neighbor then c(x,y) = */
4 for each neighbor w
5 Dw(y) = ? for all destinations y in N
6 for each neighbor w
7 send distance vector Dx = [Dx(y): y in N] to w
8
9 loop
10 wait (until I see a link cost change to some neighbor w or
11 until I receive a distance vector from some neighbor w)
12
13 for each y in N:
14 Dx(y) = minv{c(x,v) + Dv(y)}
15
16 if Dx(y) changed for any destination y
17 send distance vector Dx = [Dx(y): y in N] to all neighbors
18
19 forever
In the DV algorithm, a node x updates its distance-vector estimate when it
either sees a cost change in one of its directly attached links or receives a distance-
vector update from some neighbor. But to update its own forwarding table for a
given destination y, what node x really needs to know is not the shortest-path
distance to y but instead the neighboring node v*(y) that is the next-hop router along
the shortest path to y. As you might expect, the next-hop router v*(y) is the neighbor
v that achieves the minimum in Line 14 of the DV algorithm. (If there are multiple
neighbors v that achieve the minimum, then v*(y) can be any of the minimizing
neighbors.) Thus, in Lines 1314, for each destination y, node x also determines
v*(y) and updates its forwarding table for destination y.
Recall that the LS algorithm is a global algorithm in the sense that it requires
each node to first obtain a complete map of the network before running the Dijkstra
algorithm. The DV algorithm is decentralized and does not use such global infor-
mation. Indeed, the only information a node will have is the costs of the links to its
directly attached neighbors and information it receives from these neighbors. Each
node waits for an update from any neighbor (Lines 1011), calculates its new dis-
tance vector when receiving an update (Line 14), and distributes its new distance
vector to its neighbors (Lines 1617). DV-like algorithms are used in many routing
protocols in practice, including the Internets RIP and BGP, ISO IDRP, Novell IPX,
and the original ARPAnet.
Figure 4.30 illustrates the operation of the DV algorithm for the simple three-
node network shown at the top of the figure. The operation of the algorithm is illus-
trated in a synchronous manner, where all nodes simultaneously receive distance
vectors from their neighbors, compute their new distance vectors, and inform their
neighbors if their distance vectors have changed. After studying this example, you
374 CHAPTER 4 THE NETWORK LAYER
Node y table
Node x table
0 2 7
x y z
Time
7
2 1
y
x z
Node z table
fr
o
m
cost to
x
y
z
0 2 3
x y z
2 0 1
7 1 0f
ro
m
cost to
x
y
z
0 2 3
x y z
2 0 1
3 1 0f
ro
m
cost to
x
y
z
2 0 1
x y z
f
ro
m
cost to
x
y
z
0 2 7
x y z
2 0 1
7 1 0f
ro
m
cost to
x
y
z
0 2 3
x y z
2 0 1
3 1 0f
ro
m
cost to
x
y
z
7 1 0
x y z
fr
o
m
cost to
x
y
z
0 2 7
x y z
2 0 1
3 1 0f
ro
m
cost to
x
y
z
0 2 3
x y z
2 0 1
3 1 0f
ro
m
cost to
x
y
z
Figure 4.30 Distance-vector (DV) algorithm
should convince yourself that the algorithm operates correctly in an asynchronous
manner as well, with node computations and update generation/reception occurring
at any time.
The leftmost column of the figure displays three initial routing tables for each
of the three nodes. For example, the table in the upper-left corner is node xs initial
routing table. Within a specific routing table, each row is a distance vectorspecifi-
cally, each nodes routing table includes its own distance vector and that of each of
its neighbors. Thus, the first row in node xs initial routing table is Dx = [Dx(x),
Dx(y), Dx(z)] = [0, 2, 7]. The second and third rows in this table are the most recently
received distance vectors from nodes y and z, respectively. Because at initialization
node x has not received anything from node y or z, the entries in the second and third
rows are initialized to infinity.
After initialization, each node sends its distance vector to each of its two neigh-
bors. This is illustrated in Figure 4.30 by the arrows from the first column of tables
to the second column of tables. For example, node x sends its distance vector Dx =
[0, 2, 7] to both nodes y and z. After receiving the updates, each node recomputes its
own distance vector. For example, node x computes
Dx(x) = 0
Dx(y) = min{c(x,y) + Dy(y), c(x,z) + Dz(y)} = min{2 + 0, 7 + 1} = 2
Dx(z) = min{c(x,y) + Dy(z), c(x,z) + Dz(z)} = min{2 + 1, 7 + 0} = 3
The second column therefore displays, for each node, the nodes new distance vec-
tor along with distance vectors just received from its neighbors. Note, for example,
that node xs estimate for the least cost to node z, Dx(z), has changed from 7 to 3.
Also note that for node x, neighboring node y achieves the minimum in line 14 of
the DV algorithm; thus at this stage of the algorithm, we have at node x that v*(y) =
y and v*(z) = y.
After the nodes recompute their distance vectors, they again send their updated
distance vectors to their neighbors (if there has been a change). This is illustrated in
Figure 4.30 by the arrows from the second column of tables to the third column of
tables. Note that only nodes x and z send updates: node ys distance vector didnt
change so node y doesnt send an update. After receiving the updates, the nodes then
recompute their distance vectors and update their routing tables, which are shown in
the third column.
The process of receiving updated distance vectors from neighbors, recomputing
routing table entries, and informing neighbors of changed costs of the least-cost path
to a destination continues until no update messages are sent. At this point, since no
update messages are sent, no further routing table calculations will occur and the
algorithm will enter a quiescent state; that is, all nodes will be performing the wait
in Lines 1011 of the DV algorithm. The algorithm remains in the quiescent state
until a link cost changes, as discussed next.
4.5 ROUTING ALGORITHMS 375
Distance-Vector Algorithm: Link-Cost Changes and Link Failure
When a node running the DV algorithm detects a change in the link cost from itself to
a neighbor (Lines 1011), it updates its distance vector (Lines 1314) and, if theres a
change in the cost of the least-cost path, informs its neighbors (Lines 1617) of its new
distance vector. Figure 4.31(a) illustrates a scenario where the link cost from y to x
changes from 4 to 1. We focus here only on y and zs distance table entries to destina-
tion x. The DV algorithm causes the following sequence of events to occur:
At time t0, y detects the link-cost change (the cost has changed from 4 to 1),
updates its distance vector, and informs its neighbors of this change since its dis-
tance vector has changed.
At time t1, z receives the update from y and updates its table. It computes a new
least cost to x (it has decreased from a cost of 5 to a cost of 2) and sends its new
distance vector to its neighbors.
At time t2, y receives zs update and updates its distance table. ys least costs do
not change and hence y does not send any message to z. The algorithm comes to
a quiescent state.
Thus, only two iterations are required for the DV algorithm to reach a quiescent
state. The good news about the decreased cost between x and y has propagated
quickly through the network.
Lets now consider what can happen when a link cost increases. Suppose that
the link cost between x and y increases from 4 to 60, as shown in Figure 4.31(b).
1. Before the link cost changes, Dy(x) = 4, Dy(z) = 1, Dz(y) = 1, and Dz(x) = 5. At
time t0, y detects the link-cost change (the cost has changed from 4 to 60). y
computes its new minimum-cost path to x to have a cost of
Dy(x)= min{c(y,x) + Dx(x), c(y,z) + Dz(x)} = min{60 + 0, 1 + 5} = 6
376 CHAPTER 4 THE NETWORK LAYER
50
4
1 60
1
y
x
a. b.
z 50
4 1
y
x z
Figure 4.31 Changes in link cost
Of course, with our global view of the network, we can see that this new cost
via z is wrong. But the only information node y has is that its direct cost to x is
60 and that z has last told y that z could get to x with a cost of 5. So in order to
get to x, y would now route through z, fully expecting that z will be able to get
to x with a cost of 5. As of t1 we have a routing loopin order to get to x, y
routes through z, and z routes through y. A routing loop is like a black holea
packet destined for x arriving at y or z as of t1 will bounce back and forth
between these two nodes forever (or until the forwarding tables are changed).
2. Since node y has computed a new minimum cost to x, it informs z of its new
distance vector at time t1.
3. Sometime after t1, z receives ys new distance vector, which indicates that ys
minimum cost to x is 6. z knows it can get to y with a cost of 1 and hence
computes a new least cost to x of Dz(x) = min{50 + 0,1 + 6} = 7. Since zs
least cost to x has increased, it then informs y of its new distance vector at t2.
4. In a similar manner, after receiving zs new distance vector, y determines
Dy(x) = 8 and sends z its distance vector. z then determines Dz(x) = 9 and
sends y its distance vector, and so on.
How long will the process continue? You should convince yourself that the loop
will persist for 44 iterations (message exchanges between y and z)until z even-
tually computes the cost of its path via y to be greater than 50. At this point, z will
(finally!) determine that its least-cost path to x is via its direct connection to x. y
will then route to x via z. The result of the bad news about the increase in link
cost has indeed traveled slowly! What would have happened if the link cost c(y,
x) had changed from 4 to 10,000 and the cost c(z, x) had been 9,999? Because of
such scenarios, the problem we have seen is sometimes referred to as the count-
to-infinity problem.
Distance-Vector Algorithm: Adding Poisoned Reverse
The specific looping scenario just described can be avoided using a technique
known as poisoned reverse. The idea is simpleif z routes through y to get to
destination x, then z will advertise to y that its distance to x is infinity, that is, z will
advertise to y that Dz(x) = (even though z knows Dz(x) = 5 in truth). z will con-
tinue telling this little white lie to y as long as it routes to x via y. Since y believes
that z has no path to x, y will never attempt to route to x via z, as long as z continues
to route to x via y (and lies about doing so).
Lets now see how poisoned reverse solves the particular looping problem we
encountered before in Figure 4.31(b). As a result of the poisoned reverse, ys dis-
tance table indicates Dz(x) = . When the cost of the (x, y) link changes from 4 to 60
at time t0, y updates its table and continues to route directly to x, albeit at a higher
cost of 60, and informs z of its new cost to x, that is, Dy(x) = 60. After receiving the
4.5 ROUTING ALGORITHMS 377
update at t1, z immediately shifts its route to x to be via the direct (z, x) link at a cost
of 50. Since this is a new least-cost path to x, and since the path no longer passes
through y, z now informs y that Dz(x) = 50 at t2. After receiving the update from z, y
updates its distance table with Dy(x) = 51. Also, since z is now on ys least-cost path
to x, y poisons the reverse path from z to x by informing z at time t3 that Dy(x) =
(even though y knows that Dy(x) = 51 in truth).
Does poisoned reverse solve the general count-to-infinity problem? It does not.
You should convince yourself that loops involving three or more nodes (rather than
simply two immediately neighboring nodes) will not be detected by the poisoned
reverse technique.
A Comparison of LS and DV Routing Algorithms
The DV and LS algorithms take complementary approaches towards computing
routing. In the DV algorithm, each node talks to only its directly connected neigh-
bors, but it provides its neighbors with least-cost estimates from itself to all the
nodes (that it knows about) in the network. In the LS algorithm, each node talks with
all other nodes (via broadcast), but it tells them only the costs of its directly con-
nected links. Lets conclude our study of LS and DV algorithms with a quick com-
parison of some of their attributes. Recall that N is the set of nodes (routers) and E
is the set of edges (links).
Message complexity. We have seen that LS requires each node to know the
cost of each link in the network. This requires O(|N| |E|) messages to be sent.
Also, whenever a link cost changes, the new link cost must be sent to all
nodes. The DV algorithm requires message exchanges between directly con-
nected neighbors at each iteration. We have seen that the time needed for the
algorithm to converge can depend on many factors. When link costs change,
the DV algorithm will propagate the results of the changed link cost only if
the new link cost results in a changed least-cost path for one of the nodes
attached to that link.
Speed of convergence. We have seen that our implementation of LS is an O(|N|2)
algorithm requiring O(|N| |E|)) messages. The DV algorithm can converge slowly
and can have routing loops while the algorithm is converging. DV also suffers
from the count-to-infinity problem.
Robustness. What can happen if a router fails, misbehaves, or is sabotaged?
Under LS, a router could broadcast an incorrect cost for one of its attached
links (but no others). A node could also corrupt or drop any packets it received
as part of an LS broadcast. But an LS node is computing only its own forward-
ing tables; other nodes are performing similar calculations for themselves. This
means route calculations are somewhat separated under LS, providing a degree
of robustness. Under DV, a node can advertise incorrect least-cost paths to any
or all destinations. (Indeed, in 1997, a malfunctioning router in a small ISP
378 CHAPTER 4 THE NETWORK LAYER
provided national backbone routers with erroneous routing information. This
caused other routers to flood the malfunctioning router with traffic and caused
large portions of the Internet to become disconnected for up to several hours
[Neumann 1997].) More generally, we note that, at each iteration, a nodes cal-
culation in DV is passed on to its neighbor and then indirectly to its neighbors
neighbor on the next iteration. In this sense, an incorrect node calculation can
be diffused through the entire network under DV.
In the end, neither algorithm is an obvious winner over the other; indeed, both algo-
rithms are used in the Internet.
Other Routing Algorithms
The LS and DV algorithms we have studied are not only widely used in practice,
they are essentially the only routing algorithms used in practice today in the Inter-
net. Nonetheless, many routing algorithms have been proposed by researchers
over the past 30 years, ranging from the extremely simple to the very sophisticated
and complex. A broad class of routing algorithms is based on viewing packet traf-
fic as flows between sources and destinations in a network. In this approach, the
routing problem can be formulated mathematically as a constrained optimization
problem known as a network flow problem [Bertsekas 1991]. Yet another set of
routing algorithms we mention here are those derived from the telephony world.
These circuit-switched routing algorithms are of interest to packet-switched
data networking in cases where per-link resources (for example, buffers, or a frac-
tion of the link bandwidth) are to be reserved for each connection that is routed
over the link. While the formulation of the routing problem might appear quite
different from the least-cost routing formulation we have seen in this chapter,
there are a number of similarities, at least as far as the path-finding algorithm
(routing algorithm) is concerned. See [Ash 1998; Ross 1995; Girard 1990] for a
detailed discussion of this research area.
4.5.3 Hierarchical Routing
In our study of LS and DV algorithms, weve viewed the network simply as a col-
lection of interconnected routers. One router was indistinguishable from another in
the sense that all routers executed the same routing algorithm to compute routing
paths through the entire network. In practice, this model and its view of a homoge-
nous set of routers all executing the same routing algorithm is a bit simplistic for at
least two important reasons:
Scale. As the number of routers becomes large, the overhead involved in
computing, storing, and communicating routing information (for example,
4.5 ROUTING ALGORITHMS 379
LS updates or least-cost path changes) becomes prohibitive. Todays public
Internet consists of hundreds of millions of hosts. Storing routing information at
each of these hosts would clearly require enormous amounts of memory. The
overhead required to broadcast LS updates among all of the routers in the public
Internet would leave no bandwidth left for sending data packets! A distance-vec-
tor algorithm that iterated among such a large number of routers would surely
never converge. Clearly, something must be done to reduce the complexity of
route computation in networks as large as the public Internet.
Administrative autonomy. Although researchers tend to ignore issues such as a
companys desire to run its routers as it pleases (for example, to run whatever
routing algorithm it chooses) or to hide aspects of its networks internal organi-
zation from the outside, these are important considerations. Ideally, an organiza-
tion should be able to run and administer its network as it wishes, while still
being able to connect its network to other outside networks.
Both of these problems can be solved by organizing routers into autonomous sys-
tems (ASs), with each AS consisting of a group of routers that are typically under
the same administrative control (e.g., operated by the same ISP or belonging to the
same company network). Routers within the same AS all run the same routing algo-
rithm (for example, an LS or DV algorithm) and have information about each
otherexactly as was the case in our idealized model in the preceding section. The
routing algorithm running within an autonomous system is called an intra-
autonomous system routing protocol. It will be necessary, of course, to connect
ASs to each other, and thus one or more of the routers in an AS will have the added
task of being responsible for forwarding packets to destinations outside the AS;
these routers are called gateway routers.
Figure 4.32 provides a simple example with three ASs: AS1, AS2, and AS3.
In this figure, the heavy lines represent direct link connections between pairs of
routers. The thinner lines hanging from the routers represent subnets that are
directly connected to the routers. AS1 has four routers1a, 1b, 1c, and 1d
which run the intra-AS routing protocol used within AS1. Thus, each of these
four routers knows how to forward packets along the optimal path to any destina-
tion within AS1. Similarly, autonomous systems AS2 and AS3 each have three
routers. Note that the intra-AS routing protocols running in AS1, AS2, and AS3
need not be the same. Also note that the routers 1b, 1c, 2a, and 3a are all gateway
routers.
It should now be clear how the routers in an AS determine routing paths for
source-destination pairs that are internal to the AS. But there is still a big missing
piece to the end-to-end routing puzzle. How does a router, within some AS, know
how to route a packet to a destination that is outside the AS? Its easy to answer
this question if the AS has only one gateway router that connects to only one
other AS. In this case, because the ASs intra-AS routing algorithm has deter-
mined the least-cost path from each internal router to the gateway router, each
380 CHAPTER 4 THE NETWORK LAYER
internal router knows how it should forward the packet. The gateway router, upon
receiving the packet, forwards the packet on the one link that leads outside the
AS. The AS on the other side of the link then takes over the responsibility
of routing the packet to its ultimate destination. As an example, suppose router
2b in Figure 4.32 receives a packet whose destination is outside of AS2. Router
2b will then forward the packet to either router 2a or 2c, as specified by router
2bs forwarding table, which was configured by AS2s intra-AS routing protocol.
The packet will eventually arrive to the gateway router 2a, which will forward
the packet to 1b. Once the packet has left 2a, AS2s job is done with this one
packet.
So the problem is easy when the source AS has only one link that leads outside
the AS. But what if the source AS has two or more links (through two or more gate-
way routers) that lead outside the AS? Then the problem of knowing where to for-
ward the packet becomes significantly more challenging. For example, consider a
router in AS1 and suppose it receives a packet whose destination is outside the AS.
The router should clearly forward the packet to one of its two gateway routers, 1b or
1c, but which one? To solve this problem, AS1 needs (1) to learn which destinations
are reachable via AS2 and which destinations are reachable via AS3, and (2) to
propagate this reachability information to all the routers within AS1, so that each
router can configure its forwarding table to handle external-AS destinations. These
4.5 ROUTING ALGORITHMS 381
AS1
AS3
3b
3c
3a
1a
1c
1b
1d
AS2
2a
2c
2b
Intra-AS routing
algorithm
Forwarding
table
Inter-AS routing
algorithm
Figure 4.32 An example of interconnected autonomous systems
two tasksobtaining reachability information from neighboring ASs and propagat-
ing the reachability information to all routers internal to the ASare handled by the
inter-AS routing protocol. Since the inter-AS routing protocol involves communi-
cation between two ASs, the two communicating ASs must run the same inter-AS
routing protocol. In fact, in the Internet all ASs run the same inter-AS routing proto-
col, called BGP4, which is discussed in the next section. As shown in Figure 4.32,
each router receives information from an intra-AS routing protocol and an inter-AS
routing protocol, and uses the information from both protocols to configure its for-
warding table.
As an example, consider a subnet x (identified by its CIDRized address), and
suppose that AS1 learns from the inter-AS routing protocol that subnet x is reach-
able from AS3 but is not reachable from AS2. AS1 then propagates this information
to all of its routers. When router 1d learns that subnet x is reachable from AS3, and
hence from gateway 1c, it then determines, from the information provided by the
intra-AS routing protocol, the router interface that is on the least-cost path from
router 1d to gateway router 1c. Say this is interface I. The router 1d can then put the
entry (x, I) into its forwarding table. (This example, and others presented in this sec-
tion, gets the general ideas across but is a simplification of what really happens in
the Internet. In the next section well provide a more detailed description, albeit
more complicated, when we discuss BGP.)
Following up on the previous example, now suppose that AS2 and AS3 con-
nect to other ASs, which are not shown in the diagram. Also suppose that AS1
learns from the inter-AS routing protocol that subnet x is reachable both from AS2,
via gateway 1b, and from AS3, via gateway 1c. AS1 would then propagate this
information to all its routers, including router 1d. In order to configure its forward-
ing table, router 1d would have to determine to which gateway router, 1b or 1c, it
should direct packets that are destined for subnet x. One approach, which is often
employed in practice, is to use hot-potato routing. In hot-potato routing, the AS
gets rid of the packet (the hot potato) as quickly as possible (more precisely, as
inexpensively as possible). This is done by having a router send the packet to the
gateway router that has the smallest router-to-gateway cost among all gateways
with a path to the destination. In the context of the current example, hot-potato
routing, running in 1d, would use information from the intra-AS routing protocol
to determine the path costs to 1b and 1c, and then choose the path with the least
cost. Once this path is chosen, router 1d adds an entry for subnet x in its forward-
ing table. Figure 4.33 summarizes the actions taken at router 1d for adding the new
entry for x to the forwarding table.
When an AS learns about a destination from a neighboring AS, the AS can
advertise this routing information to some of its other neighboring ASs. For example,
suppose AS1 learns from AS2 that subnet x is reachable via AS2. AS1 could then tell
AS3 that x is reachable via AS1. In this manner, if AS3 needs to route a packet
destined to x, AS3 would forward the packet to AS1, which would in turn forward the
packet to AS2. As well see in our discussion of BGP, an AS has quite a bit of
382 CHAPTER 4 THE NETWORK LAYER
flexibility in deciding which destinations it advertises to its neighboring ASs. This
is a policy decision, typically depending more on economic issues than on technical
issues.
Recall from Section 1.5 that the Internet consists of a hierarchy of intercon-
nected ISPs. So what is the relationship between ISPs and ASs? You might think that
the routers in an ISP, and the links that interconnect them, constitute a single AS.
Although this is often the case, many ISPs partition their network into multiple ASs.
For example, some tier-1 ISPs use one AS for their entire network; others break up
their ISP into tens of interconnected ASs.
In summary, the problems of scale and administrative authority are solved by
defining autonomous systems. Within an AS, all routers run the same intra-AS rout-
ing protocol. Among themselves, the ASs run the same inter-AS routing protocol.
The problem of scale is solved because an intra-AS router need only know about
routers within its AS. The problem of administrative authority is solved since an
organization can run whatever intra-AS routing protocol it chooses; however, each
pair of connected ASs needs to run the same inter-AS routing protocol to exchange
reachability information.
In the following section, well examine two intra-AS routing protocols (RIP and
OSPF) and the inter-AS routing protocol (BGP) that are used in todays Internet.
These case studies will nicely round out our study of hierarchical routing.
4.6 Routing in the Internet
Having studied Internet addressing and the IP protocol, we now turn our attention to
the Internets routing protocols; their job is to determine the path taken by a data-
gram between source and destination. Well see that the Internets routing protocols
embody many of the principles we learned earlier in this chapter. The link-state and
distance-vector approaches studied in Sections 4.5.1 and 4.5.2 and the notion of an
autonomous system considered in Section 4.5.3 are all central to how routing is
done in todays Internet.
4.6 ROUTING IN THE INTERNET 383
Learn from inter-AS
protocol that subnet
x is reachable via
multiple gateways.
Use routing info from
intra-AS protocol to
determine costs of
least-cost paths to
each of the gateways.
Hot potato routing:
Choose the gateway
that has the
smallest least cost.
Determine from
forwarding table the
interface I that leads
to least-cost gateway.
Enter (x,I) in
forwarding table.
Figure 4.33 Steps in adding an outside-AS destination in a routers for-
warding table
Recall from Section 4.5.3 that an autonomous system (AS) is a collection of
routers under the same administrative and technical control, and that all run the
same routing protocol among themselves. Each AS, in turn, typically contains mul-
tiple subnets (where we use the term subnet in the precise, addressing sense in Sec-
tion 4.4.2).
4.6.1 Intra-AS Routing in the Internet: RIP
An intra-AS routing protocol is used to determine how routing is performed within
an autonomous system (AS). Intra-AS routing protocols are also known as interior
gateway protocols. Historically, two routing protocols have been used extensively
for routing within an autonomous system in the Internet: the Routing Information
Protocol (RIP) and Open Shortest Path First (OSPF). A routing protocol closely
related to OSPF is the IS-IS protocol [RFC 1142, Perlman 1999]. We first discuss
RIP and then consider OSPF.
RIP was one of the earliest intra-AS Internet routing protocols and is still in
widespread use today. It traces its origins and its name to the Xerox Network Sys-
tems (XNS) architecture. The widespread deployment of RIP was due in great part
to its inclusion in 1982 in the Berkeley Software Distribution (BSD) version of
UNIX supporting TCP/IP. RIP version 1 is defined in [RFC 1058], with a backward-
compatible version 2 defined in [RFC 2453].
RIP is a distance-vector protocol that operates in a manner very close to the
idealized DV protocol we examined in Section 4.5.2. The version of RIP specified
in RFC 1058 uses hop count as a cost metric; that is, each link has a cost of 1. In
the DV algorithm in Section 4.5.2, for simplicity, costs were defined between pairs
of routers. In RIP (and also in OSPF), costs are actually from source router to a des-
tination subnet. RIP uses the term hop, which is the number of subnets traversed
along the shortest path from source router to destination subnet, including the des-
tination subnet. Figure 4.34 illustrates an AS with six leaf subnets. The table in the
figure indicates the number of hops from the source A to each of the leaf subnets.
The maximum cost of a path is limited to 15, thus limiting the use of RIP to
autonomous systems that are fewer than 15 hops in diameter. Recall that in DV
protocols, neighboring routers exchange distance vectors with each other. The
distance vector for any one router is the current estimate of the shortest path
distances from that router to the subnets in the AS. In RIP, routing updates
are exchanged between neighbors approximately every 30 seconds using a
RIP response message. The response message sent by a router or host contains
a list of up to 25 destination subnets within the AS, as well as the senders
distance to each of those subnets. Response messages are also known as RIP
advertisements.
Lets take a look at a simple example of how RIP advertisements work. Con-
sider the portion of an AS shown in Figure 4.35. In this figure, lines connecting the
routers denote subnets. Only selected routers (A, B, C, and D) and subnets (w, x, y,
384 CHAPTER 4 THE NETWORK LAYER
and z) are labeled. Dotted lines indicate that the AS continues on; thus this
autonomous system has many more routers and links than are shown.
Each router maintains a RIP table known as a routing table. A routers routing
table includes both the routers distance vector and the routers forwarding table.
Figure 4.36 shows the routing table for router D. Note that the routing table has
three columns. The first column is for the destination subnet, the second column
indicates the identity of the next router along the shortest path to the destination sub-
net, and the third column indicates the number of hops (that is, the number of sub-
nets that have to be traversed, including the destination subnet) to get to the
destination subnet along the shortest path. For this example, the table indicates that
to send a datagram from router D to destination subnet w, the datagram should first
be forwarded to neighboring router A; the table also indicates that destination sub-
net w is two hops away along the shortest path. Similarly, the table indicates that
subnet z is seven hops away via router B. In principle, a routing table will have one
row for each subnet in the AS, although RIP version 2 allows subnet entries to be
aggregated using route aggregation techniques similar to those we examined in
4.6 ROUTING IN THE INTERNET 385
C D
A
u
Destination Hops
u
v
w
x
y
z
1
2
2
3
3
2
v
w
x
y z
B
Figure 4.34 Number of hops from source router A to various subnets
A
C
D B
z
w x y
Figure 4.35 A portion of an autonomous system
Section 4.4. The table in Figure 4.36, and the subsequent tables to come, are only
partially complete.
Now suppose that 30 seconds later, router D receives from router A the adver-
tisement shown in Figure 4.37. Note that this advertisement is nothing other than
the routing table information from router A! This information indicates, in particu-
lar, that subnet z is only four hops away from router A. Router D, upon receiving this
advertisement, merges the advertisement (Figure 4.37) with the old routing table
(Figure 4.36). In particular, router D learns that there is now a path through router A
to subnet z that is shorter than the path through router B. Thus, router D updates its
routing table to account for the shorter shortest path, as shown in Figure 4.38. How
is it, you might ask, that the shortest path to subnet z has become shorter? Possibly,
the decentralized distance-vector algorithm is still in the process of converging (see
Section 4.5.2), or perhaps new links and/or routers were added to the AS, thus
changing the shortest paths in the AS.
Lets next consider a few of the implementation aspects of RIP. Recall that
RIP routers exchange advertisements approximately every 30 seconds. If a router
does not hear from its neighbor at least once every 180 seconds, that neighbor is
considered to be no longer reachable; that is, either the neighbor has died or the
386 CHAPTER 4 THE NETWORK LAYER
Destination Subnet Next Router Number of Hops to Destination
w A 2
y B 2
z B 7
x 1
. . . . . . . . . . . .
Figure 4.36 Routing table in router D before receiving advertisement
from router A
Destination Subnet Next Router Number of Hops to Destination
z C 4
w 1
x 1
. . . . . . . . . . . .
Figure 4.37 Advertisement from router A
connecting link has gone down. When this happens, RIP modifies the local routing
table and then propagates this information by sending advertisements to its neigh-
boring routers (the ones that are still reachable). A router can also request informa-
tion about its neighbors cost to a given destination using RIPs request message.
Routers send RIP request and response messages to each other over UDP using port
number 520. The UDP segment is carried between routers in a standard IP data-
gram. The fact that RIP uses a transport-layer protocol (UDP) on top of a network-
layer protocol (IP) to implement network-layer functionality (a routing algorithm)
may seem rather convoluted (it is!). Looking a little deeper at how RIP is imple-
mented will clear this up.
Figure 4.39 sketches how RIP is typically implemented in a UNIX system, for
example, a UNIX workstation serving as a router. A process called routed (pronounced
route dee) executes RIP, that is, maintains routing information and exchanges
messages with routed processes running in neighboring routers. Because RIP is
implemented as an application-layer process (albeit a very special one that is able to
4.6 ROUTING IN THE INTERNET 387
Destination Subnet Next Router Number of Hops to Destination
w A 2
y B 2
z A 5
. . . . . . . . . . . .
Figure 4.38 Routing table in router D after receiving advertisement from
router A
Network
(IP)
Transport
(UDP)
Data link
Physical
Forwarding
tables
Routed
Forwarding
tables
Routed
Network
(IP)
Transport
(UDP)
Data link
Physical
Figure 4.39 Implementation of RIP as the routed daemon
manipulate the routing tables within the UNIX kernel), it can send and receive mes-
sages over a standard socket and use a standard transport protocol. As shown, RIP is
implemented as an application-layer protocol (see Chapter 2) running over UDP. If
youre interested in looking at an implementation of RIP (or the OSPF and BGP pro-
tocols that we will study shortly), see [Quagga 2012].
4.6.2 Intra-AS Routing in the Internet: OSPF
Like RIP, OSPF routing is widely used for intra-AS routing in the Internet. OSPF
and its closely related cousin, IS-IS, are typically deployed in upper-tier ISPs
whereas RIP is deployed in lower-tier ISPs and enterprise networks. The Open in
OSPF indicates that the routing protocol specification is publicly available (for
example, as opposed to Ciscos EIGRP protocol). The most recent version of OSPF,
version 2, is defined in RFC 2328, a public document.
OSPF was conceived as the successor to RIP and as such has a number of
advanced features. At its heart, however, OSPF is a link-state protocol that uses
flooding of link-state information and a Dijkstra least-cost path algorithm. With
OSPF, a router constructs a complete topological map (that is, a graph) of the entire
autonomous system. The router then locally runs Dijkstras shortest-path algorithm
to determine a shortest-path tree to all subnets, with itself as the root node. Individ-
ual link costs are configured by the network administrator (see Principles and Prac-
tice: Setting OSPF Weights). The administrator might choose to set all link costs to
1, thus achieving minimum-hop routing, or might choose to set the link weights to
be inversely proportional to link capacity in order to discourage traffic from using
low-bandwidth links. OSPF does not mandate a policy for how link weights are set
(that is the job of the network administrator), but instead provides the mechanisms
(protocol) for determining least-cost path routing for the given set of link weights.
With OSPF, a router broadcasts routing information to all other routers in the
autonomous system, not just to its neighboring routers. A router broadcasts link-
state information whenever there is a change in a links state (for example, a change
in cost or a change in up/down status). It also broadcasts a links state periodically
(at least once every 30 minutes), even if the links state has not changed. RFC 2328
notes that this periodic updating of link state advertisements adds robustness to the
link state algorithm. OSPF advertisements are contained in OSPF messages that
are carried directly by IP, with an upper-layer protocol of 89 for OSPF. Thus, the
OSPF protocol must itself implement functionality such as reliable message transfer
and link-state broadcast. The OSPF protocol also checks that links are operational
(via a HELLO message that is sent to an attached neighbor) and allows an OSPF
router to obtain a neighboring routers database of network-wide link state.
Some of the advances embodied in OSPF include the following:
Security. Exchanges between OSPF routers (for example, link-state updates)
can be authenticated. With authentication, only trusted routers can participate
388 CHAPTER 4 THE NETWORK LAYER
in the OSPF protocol within an AS, thus preventing malicious intruders (or net-
working students taking their newfound knowledge out for a joyride) from
injecting incorrect information into router tables. By default, OSPF packets
between routers are not authenticated and could be forged. Two types of
authentication can be configuredsimple and MD5 (see Chapter 8 for a dis-
cussion on MD5 and authentication in general). With simple authentication, the
same password is configured on each router. When a router sends an OSPF
packet, it includes the password in plaintext. Clearly, simple authentication is
not very secure. MD5 authentication is based on shared secret keys that are
configured in all the routers. For each OSPF packet that it sends, the router
computes the MD5 hash of the content of the OSPF packet appended with the
secret key. (See the discussion of message authentication codes in Chapter 7.)
Then the router includes the resulting hash value in the OSPF packet. The
receiving router, using the preconfigured secret key, will compute an MD5
hash of the packet and compare it with the hash value that the packet carries,
thus verifying the packets authenticity. Sequence numbers are also used with
MD5 authentication to protect against replay attacks.
Multiple same-cost paths. When multiple paths to a destination have the same
cost, OSPF allows multiple paths to be used (that is, a single path need not be
chosen for carrying all traffic when multiple equal-cost paths exist).
Integrated support for unicast and multicast routing. Multicast OSPF (MOSPF)
[RFC 1584] provides simple extensions to OSPF to provide for multicast routing
(a topic we cover in more depth in Section 4.7.2). MOSPF uses the existing
OSPF link database and adds a new type of link-state advertisement to the exist-
ing OSPF link-state broadcast mechanism.
Support for hierarchy within a single routing domain. Perhaps the most signifi-
cant advance in OSPF is the ability to structure an autonomous system hierarchi-
cally. Section 4.5.3 has already looked at the many advantages of hierarchical
routing structures. We cover the implementation of OSPF hierarchical routing in
the remainder of this section.
An OSPF autonomous system can be configured hierarchically into areas. Each
area runs its own OSPF link-state routing algorithm, with each router in an area
broadcasting its link state to all other routers in that area. Within each area, one or
more area border routers are responsible for routing packets outside the area. Lastly,
exactly one OSPF area in the AS is configured to be the backbone area. The primary
role of the backbone area is to route traffic between the other areas in the AS. The
backbone always contains all area border routers in the AS and may contain nonbor-
der routers as well. Inter-area routing within the AS requires that the packet be first
routed to an area border router (intra-area routing), then routed through the back-
bone to the area border router that is in the destination area, and then routed to the
final destination.
4.6 ROUTING IN THE INTERNET 389
OSPF is a relatively complex protocol, and our coverage here has been neces-
sarily brief; [Huitema 1998; Moy 1998; RFC 2328] provide additional details.
4.6.3 Inter-AS Routing: BGP
We just learned how ISPs use RIP and OSPF to determine optimal paths for source-
destination pairs that are internal to the same AS. Lets now examine how paths are
determined for source-destination pairs that span multiple ASs. The Border Gate-
way Protocol version 4, specified in RFC 4271 (see also [RFC 4274), is the de facto
standard inter-AS routing protocol in todays Internet. It is commonly referred to as
BGP4 or simply as BGP. As an inter-AS routing protocol (see Section 4.5.3), BGP
provides each AS a means to
1. Obtain subnet reachability information from neighboring ASs.
2. Propagate the reachability information to all routers internal to the AS.
3. Determine good routes to subnets based on the reachability information and
on AS policy.
390 CHAPTER 4 THE NETWORK LAYER
SETTING OSPF L INK WEIGHTS
Our discussion of link-state routing has implicitly assumed that link weights are set, a rout-
ing algorithm such as OSPF is run, and traffic flows according to the routing tables comput-
ed by the LS algorithm. In terms of cause and effect, the link weights are given (i.e., they
come first) and result (via Dijkstras algorithm) in routing paths that minimize overall cost. In
this viewpoint, link weights reflect the cost of using a link (e.g., if link weights are inversely
proportional to capacity, then the use of high-capacity links would have smaller weights
and thus be more attractive from a routing standpoint) and Disjkstras algorithm serves to
minimize overall cost.
In practice, the cause and effect relationship between link weights and routing paths
may be reversed, with network operators configuring link weights in order to obtain routing
paths that achieve certain traffic engineering goals [Fortz 2000, Fortz 2002]. For example,
suppose a network operator has an estimate of traffic flow entering the network at each
ingress point and destined for each egress point. The operator may then want to put in
place a specific routing of ingress-to-egress flows that minimizes the maximum utilization
over all of the networks links. But with a routing algorithm such as OSPF, the operators
main knobs for tuning the routing of flows through the network are the link weights. Thus,
in order to achieve the goal of minimizing the maximum link utilization, the operator must
find the set of link weights that achieves this goal. This is a reversal of the cause and effect
relationshipthe desired routing of flows is known, and the OSPF link weights must be
found such that the OSPF routing algorithm results in this desired routing of flows.
PRINCIPLES IN PRACTICE
Most importantly, BGP allows each subnet to advertise its existence to the rest of
the Internet. A subnet screams I exist and I am here, and BGP makes sure that all
the ASs in the Internet know about the subnet and how to get there. If it werent for
BGP, each subnet would be isolatedalone and unknown by the rest of the Internet.
BGP Basics
BGP is extremely complex; entire books have been devoted to the subject and many
issues are still not well understood [Yannuzzi 2005]. Furthermore, even after having
read the books and RFCs, you may find it difficult to fully master BGP without hav-
ing practiced BGP for many months (if not years) as a designer or administrator of
an upper-tier ISP. Nevertheless, because BGP is an absolutely critical protocol for
the Internetin essence, it is the protocol that glues the whole thing togetherwe
need to acquire at least a rudimentary understanding of how it works. We begin by
describing how BGP might work in the context of the simple example network we
studied earlier in Figure 4.32. In this description, we build on our discussion of hier-
archical routing in Section 4.5.3; we encourage you to review that material.
In BGP, pairs of routers exchange routing information over semipermanent
TCP connections using port 179. The semi-permanent TCP connections for the net-
work in Figure 4.32 are shown in Figure 4.40. There is typically one such BGP TCP
connection for each link that directly connects two routers in two different ASs;
thus, in Figure 4.40, there is a TCP connection between gateway routers 3a and 1c
and another TCP connection between gateway routers 1b and 2a. There are also
semipermanent BGP TCP connections between routers within an AS. In particular,
Figure 4.40 displays a common configuration of one TCP connection for each pair
of routers internal to an AS, creating a mesh of TCP connections within each AS.
For each TCP connection, the two routers at the end of the connection are called
BGP peers, and the TCP connection along with all the BGP messages sent over the
4.6 ROUTING IN THE INTERNET 391
AS1
AS3
3b
eBGP session
Key:
iBGP session
3c
3a
1a
1c
1b
1d
AS2
2a
2c
2b
Figure 4.40 eBGP and iBGP sessions
VideoNote
Gluing the Internet
together
392 CHAPTER 4 THE NETWORK LAYER
OBTAINING INTERNET PRESENCE: PUTTING THE PUZZLE TOGETHER
Suppose you have just created a small that has a number of servers, including a public
Web server that describes your companys products and services, a mail server from which
your employees obtain their email messages, and a DNS server. Naturally, you would like
the entire world to be able to surf your Web site in order to learn about your exciting prod-
ucts and services. Moreover, you would like your employees to be able to send and
receive email to potential customers throughout the world.
To meet these goals, you first need to obtain Internet connectivity, which is done by
contracting with, and connecting to, a local ISP. Your company will have a gateway
router, which will be connected to a router in your local ISP. This connection might be
a DSL connection through the existing telephone infrastructure, a leased line to the ISPs
router, or one of the many other access solutions described in Chapter 1. Your local
ISP will also provide you with an IP address range, e.g., a /24 address range consist-
ing of 256 addresses. Once you have your physical connectivity and your IP address
range, you will assign one of the IP addresses (in your address range) to your Web
server, one to your mail server, one to your DNS server, one to your gateway router,
and other IP addresses to other servers and networking devices in your companys
network.
In addition to contracting with an ISP, you will also need to contract with an Internet regis-
trar to obtain a domain name for your company, as described in Chapter 2. For example, if
your companys name is, say, Xanadu Inc., you will naturally try to obtain the domain name
xanadu.com. Your company must also obtain presence in the DNS system. Specifically,
because outsiders will want to contact your DNS server to obtain the IP addresses of your
servers, you will also need to provide your registrar with the IP address of your DNS server.
Your registrar will then put an entry for your DNS server (domain name and corresponding IP
address) in the .com top-level-domain servers, as described in Chapter 2. After this step is
completed, any user who knows your domain name (e.g., xanadu.com) will be able to
obtain the IP address of your DNS server via the DNS system.
So that people can discover the IP addresses of your Web server, in your DNS server
you will need to include entries that map the host name of your Web server (e.g.,
www.xanadu.com) to its IP address. You will want to have similar entries for other publicly
available servers in your company, including your mail server. In this manner, if Alice
wants to browse your Web server, the DNS system will contact your DNS server, find the
IP address of your Web server, and give it to Alice. Alice can then establish a TCP
connection directly with your Web server.
However, there still remains one other necessary and crucial step to allow outsiders
from around the world access your Web server. Consider what happens when Alice,
who knows the IP address of your Web server, sends an IP datagram (e.g., a TCP SYN
segment) to that IP address. This datagram will be routed through the Internet, visiting a
series of routers in many different ASes, and eventually reach your Web server. When
PRINCIPLES IN PRACTICE
www.xanadu.com
connection is called a BGP session. Furthermore, a BGP session that spans two ASs
is called an external BGP (eBGP) session, and a BGP session between routers in
the same AS is called an internal BGP (iBGP) session. In Figure 4.40, the eBGP
sessions are shown with the long dashes; the iBGP sessions are shown with the short
dashes. Note that BGP session lines in Figure 4.40 do not always correspond to the
physical links in Figure 4.32.
BGP allows each AS to learn which destinations are reachable via its neighbor-
ing ASs. In BGP, destinations are not hosts but instead are CIDRized prefixes, with
each prefix representing a subnet or a collection of subnets. Thus, for example, sup-
pose there are four subnets attached to AS2: 138.16.64/24, 138.16.65/24,
138.16.66/24, and 138.16.67/24. Then AS2 could aggregate the prefixes for these four
subnets and use BGP to advertise the single prefix to 138.16.64/22 to AS1. As another
example, suppose that only the first three of those four subnets are in AS2 and the
fourth subnet, 138.16.67/24, is in AS3. Then, as described in the Principles and Prac-
tice in Section 4.4.2, because routers use longest-prefix matching for forwarding data-
grams, AS3 could advertise to AS1 the more specific prefix 138.16.67/24 and AS2
could still advertise to AS1 the aggregated prefix 138.16.64/22.
Lets now examine how BGP would distribute prefix reachability information
over the BGP sessions shown in Figure 4.40. As you might expect, using the eBGP
session between the gateway routers 3a and 1c, AS3 sends AS1 the list of prefixes
that are reachable from AS3; and AS1 sends AS3 the list of prefixes that are reach-
able from AS1. Similarly, AS1 and AS2 exchange prefix reachability information
through their gateway routers 1b and 2a. Also as you may expect, when a gateway
router (in any AS) receives eBGP-learned prefixes, the gateway router uses its iBGP
sessions to distribute the prefixes to the other routers in the AS. Thus, all the routers
in AS1 learn about AS3 prefixes, including the gateway router 1b. The gateway
router 1b (in AS1) can therefore re-advertise AS3s prefixes to AS2. When a router
(gateway or not) learns about a new prefix, it creates an entry for the prefix in its
forwarding table, as described in Section 4.5.3.
4.6 ROUTING IN THE INTERNET 393
any one of the routers receives the datagram, it is going to look for an entry in its
forwarding table to determine on which outgoing port it should forward the datagram.
Therefore, each of the routers needs to know about the existence of your companys
/24 prefix (or some aggregate entry). How does a router become aware of your
companys prefix? As we have just seen, it becomes aware of it from BGP! Specifically,
when your company contracts with a local ISP and gets assigned a prefix (i.e., an
address range), your local ISP will use BGP to advertise this prefix to the ISPs to which
it connects. Those ISPs will then, in turn, use BGP to propagate the advertisement.
Eventually, all Internet routers will know about your prefix (or about some aggregate that
includes your prefix) and thus be able to appropriately forward datagrams destined to
your Web and mail servers.
Path Attributes and BGP Routes
Having now a preliminary understanding of BGP, lets get a little deeper into it
(while still brushing some of the less important details under the rug!). In BGP, an
autonomous system is identified by its globally unique autonomous system num-
ber (ASN) [RFC 1930]. (Technically, not every AS has an ASN. In particular, a so-
called stub AS that carries only traffic for which it is a source or destination will not
typically have an ASN; we ignore this technicality in our discussion in order to bet-
ter see the forest for the trees.) AS numbers, like IP addresses, are assigned by
ICANN regional registries [ICANN 2012].
When a router advertises a prefix across a BGP session, it includes with the pre-
fix a number of BGP attributes. In BGP jargon, a prefix along with its attributes is
called a route. Thus, BGP peers advertise routes to each other. Two of the more
important attributes are AS-PATH and NEXT-HOP:
AS-PATH. This attribute contains the ASs through which the advertisement for the
prefix has passed. When a prefix is passed into an AS, the AS adds its ASN to the AS-
PATH attribute. For example, consider Figure 4.40 and suppose that prefix
138.16.64/24 is first advertised from AS2 to AS1; if AS1 then advertises the prefix to
AS3, AS-PATH would be AS2 AS1. Routers use the AS-PATH attribute to detect and
prevent looping advertisements; specifically, if a router sees that its AS is contained
in the path list, it will reject the advertisement. As well soon discuss, routers also use
the AS-PATH attribute in choosing among multiple paths to the same prefix.
Providing the critical link between the inter-AS and intra-AS routing protocols, the
NEXT-HOP attribute has a subtle but important use. The NEXT-HOP is the router
interface that begins the AS-PATH. To gain insight into this attribute, lets again refer
to Figure 4.40. Consider what happens when the gateway router 3a in AS3 advertises
a route to gateway router 1c in AS1 using eBGP. The route includes the advertised
prefix, which well call x, and an AS-PATH to the prefix. This advertisement also
includes the NEXT-HOP, which is the IP address of the router 3a interface that leads
to 1c. (Recall that a router has multiple IP addresses, one for each of its interfaces.)
Now consider what happens when router 1d learns about this route from iBGP. After
learning about this route to x, router 1d may want to forward packets to x along the
route, that is, router 1d may want to include the entry (x, l) in its forwarding table,
where l is its interface that begins the least-cost path from 1d towards the gateway
router 1c. To determine l, 1d provides the IP address in the NEXT-HOP attribute to
its intra-AS routing module. Note that the intra-AS routing algorithm has determined
the least-cost path to all subnets attached to the routers in AS1, including to the sub-
net for the link between 1c and 3a. From this least-cost path from 1d to the 1c-3a sub-
net, 1d determines its router interface l that begins this path and then adds the entry
(x, l) to its forwarding table. Whew! In summary, the NEXT-HOP attribute is used by
routers to properly configure their forwarding tables.
Figure 4.41 illustrates another situation where the NEXT-HOP is needed. In this fig-
ure, AS1 and AS2 are connected by two peering links. A router in AS1 could learn
394 CHAPTER 4 THE NETWORK LAYER
about two different routes to the same prefix x. These two routes could have the same
AS-PATH to x, but could have different NEXT-HOP values corresponding to the dif-
ferent peering links. Using the NEXT-HOP values and the intra-AS routing algo-
rithm, the router can determine the cost of the path to each peering link, and then
apply hot-potato routing (see Section 4.5.3) to determine the appropriate interface.
BGP also includes attributes that allow routers to assign preference metrics to
the routes, and an attribute that indicates how the prefix was inserted into BGP at
the origin AS. For a full discussion of route attributes, see [Griffin 2012; Stewart
1999; Halabi 2000; Feamster 2004; RFC 4271].
When a gateway router receives a route advertisement, it uses its import policy
to decide whether to accept or filter the route and whether to set certain attributes
such as the router preference metrics. The import policy may filter a route because
the AS may not want to send traffic over one of the ASs in the routes AS-PATH.
The gateway router may also filter a route because it already knows of a preferable
route to the same prefix.
BGP Route Selection
As described earlier in this section, BGP uses eBGP and iBGP to distribute routes
to all the routers within ASs. From this distribution, a router may learn about more
than one route to any one prefix, in which case the router must select one of the
4.6 ROUTING IN THE INTERNET 395
Two peering
links between
AS2 and AS1
Router learns
about a
route to x
Router learns about
another route to x
AS2
AS1
Key:
Route advertisements
message for destination x
Figure 4.41 NEXT-HOP attributes in advertisements are used to deter-
mine which peering link to use
possible routes. The input into this route selection process is the set of all routes that
have been learned and accepted by the router. If there are two or more routes to the
same prefix, then BGP sequentially invokes the following elimination rules until one
route remains:
Routes are assigned a local preference value as one of their attributes. The local
preference of a route could have been set by the router or could have been
learned by another router in the same AS. This is a policy decision that is left up
to the ASs network administrator. (We will shortly discuss BGP policy issues in
some detail.) The routes with the highest local preference values are selected.
From the remaining routes (all with the same local preference value), the route with
the shortest AS-PATH is selected. If this rule were the only rule for route selection,
then BGP would be using a DV algorithm for path determination, where the dis-
tance metric uses the number of AS hops rather than the number of router hops.
From the remaining routes (all with the same local preference value and the same
AS-PATH length), the route with the closest NEXT-HOP router is selected. Here,
closest means the router for which the cost of the least-cost path, determined by
the intra-AS algorithm, is the smallest. As discussed in Section 4.5.3, this process
is called hot-potato routing.
If more than one route still remains, the router uses BGP identifiers to select the
route; see [Stewart 1999].
The elimination rules are even more complicated than described above. To avoid
nightmares about BGP, its best to learn about BGP selection rules in small doses!
396 CHAPTER 4 THE NETWORK LAYER
PUTTING IT ALL TOGETHER: HOW DOES AN ENTRY GET INTO A ROUTERS
FORWARDING TABLE?
Recall that an entry in a routers forwarding table consists of a prefix (e.g., 138.16.64/22)
and a corresponding router output port (e.g., port 7). When a packet arrives to the router,
the packets destination IP address is compared with the prefixes in the forwarding table to
find the one with the longest prefix match. The packet is then forwarded (within the router)
to the router port associated with that prefix. Lets now summarize how a routing entry
(prefix and associated port) gets entered into a forwarding table. This simple exercise will
tie together a lot of what we just learned about routing and forwarding. To make things
interesting, lets assume that the prefix is a foreign prefix, that is, it does not belong to
the routers AS but to some other AS.
In order for a prefix to get entered into the routers forwarding table, the router has to
first become aware of the prefix (corresponding to a subnet or an aggregation of sub-
nets). As we have just learned, the router becomes aware of the prefix via a BGP route
PRINCIPLES IN PRACTICE
Routing Policy
Lets illustrate some of the basic concepts of BGP routing policy with a simple exam-
ple. Figure 4.42 shows six interconnected autonomous systems: A, B, C, W, X, and Y.
It is important to note that A, B, C, W, X, and Y are ASs, not routers. Lets assume that
autonomous systems W, X, and Y are stub networks and that A, B, and C are backbone
provider networks. Well also assume that A, B, and C, all peer with each other, and
provide full BGP information to their customer networks. All traffic entering a stub
network must be destined for that network, and all traffic leaving a stub network must
have originated in that network. W and Y are clearly stub networks. X is a multi-
homed stub network, since it is connected to the rest of the network via two different
providers (a scenario that is becoming increasingly common in practice). However,
like W and Y, X itself must be the source/destination of all traffic leaving/entering X.
But how will this stub network behavior be implemented and enforced? How will X
be prevented from forwarding traffic between B and C? This can easily be
advertisement. Such an advertisement may be sent to it over an eBGP session (from a
router in another AS) or over an iBGP session (from a router in the same AS).
After the router becomes aware of the prefix, it needs to determine the appropriate output
port to which datagrams destined to that prefix will be forwarded, before it can enter that
prefix in its forwarding table. If the router receives more than one route advertisement for this
prefix, the router uses the BGP route selection process, as described earlier in this subsection,
to find the best route for the prefix. Suppose such a best route has been selected. As
described earlier, the selected route includes a NEXT-HOP attribute, which is the IP address of
the first router outside the routers AS along this best route. As described above, the router
then uses its intra-AS routing protocol (typically OSPF) to determine the shortest path to the
NEXT-HOP router. The router finally determines the port number to associate with the prefix
by identifying the first link along that shortest path. The router can then (finally!) enter the
prefix-port pair into its forwarding table! The forwarding table computed by the routing
processor (see Figure 4.6) is then pushed to the routers input port line cards.
AW
X
Y
B
Key:
Provider
network
Customer
networkC
Figure 4.42 A simple BGP scenario
4.6 ROUTING IN THE INTERNET 397
398 CHAPTER 4 THE NETWORK LAYER
WHY ARE THERE DIFFERENT INTER-AS AND INTRA-AS
ROUTING PROTOCOLS?
Having now studied the details of specific inter-AS and intra-AS routing protocols deployed
in todays Internet, lets conclude by considering perhaps the most fundamental question we
could ask about these protocols in the first place (hopefully, you have been wondering this
all along, and have not lost the forest for the trees!): Why are different inter-AS and intra-
AS routing protocols used?
The answer to this question gets at the heart of the differences between the goals of
routing within an AS and among ASs:
Policy. Among ASs, policy issues dominate. It may well be important that traffic origi-
nating in a given AS not be able to pass through another specific AS. Similarly, a
given AS may well want to control what transit traffic it carries between other ASs. We
have seen that BGP carries path attributes and provides for controlled distribution of
routing information so that such policy-based routing decisions can be made. Within an
AS, everything is nominally under the same administrative control, and thus policy
issues play a much less important role in choosing routes within the AS.
Scale. The ability of a routing algorithm and its data structures to scale to handle rout-
ing to/among large numbers of networks is a critical issue in inter-AS routing. Within
an AS, scalability is less of a concern. For one thing, if a single administrative domain
becomes too large, it is always possible to divide it into two ASs and perform inter-AS
routing between the two new ASs. (Recall that OSPF allows such a hierarchy to be
built by splitting an AS into areas.)
Performance. Because inter-AS routing is so policy oriented, the quality (for example,
performance) of the routes used is often of secondary concern (that is, a longer or more
costly route that satisfies certain policy criteria may well be taken over a route that is
shorter but does not meet that criteria). Indeed, we saw that among ASs, there is not
even the notion of cost (other than AS hop count) associated with routes. Within a sin-
gle AS, however, such policy concerns are of less importance, allowing routing to focus
more on the level of performance realized on a route.
PRINCIPLES IN PRACTICE
accomplished by controlling the manner in which BGP routes are advertised. In par-
ticular, X will function as a stub network if it advertises (to its neighbors B and C) that
it has no paths to any other destinations except itself. That is, even though X may
know of a path, say XCY, that reaches network Y, it will not advertise this path to B.
Since B is unaware that X has a path to Y, B would never forward traffic destined to Y
(or C) via X. This simple example illustrates how a selective route advertisement pol-
icy can be used to implement customer/provider routing relationships.
Lets next focus on a provider network, say AS B. Suppose that B has learned
(from A) that A has a path AW to W. B can thus install the route BAW into its rout-
ing information base. Clearly, B also wants to advertise the path BAW to its cus-
tomer, X, so that X knows that it can route to W via B. But should B advertise the
path BAW to C? If it does so, then C could route traffic to W via CBAW. If A, B,
and C are all backbone providers, than B might rightly feel that it should not have
to shoulder the burden (and cost!) of carrying transit traffic between A and C. B
might rightly feel that it is As and Cs job (and cost!) to make sure that C can route
to/from As customers via a direct connection between A and C. There are currently
no official standards that govern how backbone ISPs route among themselves. How-
ever, a rule of thumb followed by commercial ISPs is that any traffic flowing across
an ISPs backbone network must have either a source or a destination (or both) in a
network that is a customer of that ISP; otherwise the traffic would be getting a free
ride on the ISPs network. Individual peering agreements (that would govern ques-
tions such as those raised above) are typically negotiated between pairs of ISPs and
are often confidential; [Huston 1999a] provides an interesting discussion of peering
agreements. For a detailed description of how routing policy reflects commercial
relationships among ISPs, see [Gao 2001; Dmitiropoulos 2007]. For a discussion of
BGP routing polices from an ISP standpoint, see [Caesar 2005b].
As noted above, BGP is the de facto standard for inter-AS routing for the pub-
lic Internet. To see the contents of various BGP routing tables (large!) extracted
from routers in tier-1 ISPs, see http://www.routeviews.org. BGP routing tables
often contain tens of thousands of prefixes and corresponding attributes. Statistics
about the size and characteristics of BGP routing tables are presented in [Potaroo
2012].
This completes our brief introduction to BGP. Understanding BGP is important
because it plays a central role in the Internet. We encourage you to see the references
[Griffin 2012; Stewart 1999; Labovitz 1997; Halabi 2000; Huitema 1998; Gao
2001; Feamster 2004; Caesar 2005b; Li 2007] to learn more about BGP.
4.7 Broadcast and Multicast Routing
Thus far in this chapter, our focus has been on routing protocols that support unicast
(i.e., point-to-point) communication, in which a single source node sends a packet
to a single destination node. In this section, we turn our attention to broadcast and
multicast routing protocols. In broadcast routing, the network layer provides a
service of delivering a packet sent from a source node to all other nodes in the
network; multicast routing enables a single source node to send a copy of a packet
to a subset of the other network nodes. In Section 4.7.1 well consider broadcast
routing algorithms and their embodiment in routing protocols. Well examine multi-
cast routing in Section 4.7.2.
4.7 BROADCAST AND MULTICAST ROUTING 399
http://www.routeviews.org
4.7.1 Broadcast Routing Algorithms
Perhaps the most straightforward way to accomplish broadcast communication is
for the sending node to send a separate copy of the packet to each destination, as
shown in Figure 4.43(a). Given N destination nodes, the source node simply makes
N copies of the packet, addresses each copy to a different destination, and then
transmits the N copies to the N destinations using unicast routing. This N-way-
unicast approach to broadcasting is simpleno new network-layer routing proto-
col, packet-duplication, or forwarding functionality is needed. There are, however,
several drawbacks to this approach. The first drawback is its inefficiency. If the
source node is connected to the rest of the network via a single link, then N separate
copies of the (same) packet will traverse this single link. It would clearly be more
efficient to send only a single copy of a packet over this first hop and then have the
node at the other end of the first hop make and forward any additional needed
copies. That is, it would be more efficient for the network nodes themselves (rather
than just the source node) to create duplicate copies of a packet. For example, in
Figure 4.43(b), only a single copy of a packet traverses the R1-R2 link. That packet
is then duplicated at R2, with a single copy being sent over links R2-R3 and R2-R4.
The additional drawbacks of N-way-unicast are perhaps more subtle, but no less
important. An implicit assumption of N-way-unicast is that broadcast recipients, and
their addresses, are known to the sender. But how is this information obtained? Most
likely, additional protocol mechanisms (such as a broadcast membership or
destination-registration protocol) would be required. This would add more overhead
and, importantly, additional complexity to a protocol that had initially seemed quite
simple. A final drawback of N-way-unicast relates to the purposes for which broad-
cast is to be used. In Section 4.5, we learned that link-state routing protocols use
broadcast to disseminate the link-state information that is used to compute unicast
routes. Clearly, in situations where broadcast is used to create and update unicast
routes, it would be unwise (at best!) to rely on the unicast routing infrastructure to
achieve broadcast.
400 CHAPTER 4 THE NETWORK LAYER
R2
R4R3
a. b.
Duplicate
Duplicate creation/transmission
R1
R4R3
Duplicate
R1
R2
Figure 4.43 Source-duplication versus in-network duplication
Given the several drawbacks of N-way-unicast broadcast, approaches in which
the network nodes themselves play an active role in packet duplication, packet for-
warding, and computation of the broadcast routes are clearly of interest. Well
examine several such approaches below and again adopt the graph notation intro-
duced in Section 4.5. We again model the network as a graph, G = (N,E), where N
is a set of nodes and a collection E of edges, where each edge is a pair of nodes from
N. Well be a bit sloppy with our notation and use N to refer to both the set of nodes,
as well as the cardinality (|N|) or size of that set when there is no confusion.
Uncontrolled Flooding
The most obvious technique for achieving broadcast is a flooding approach in
which the source node sends a copy of the packet to all of its neighbors. When a
node receives a broadcast packet, it duplicates the packet and forwards it to all of its
neighbors (except the neighbor from which it received the packet). Clearly, if the
graph is connected, this scheme will eventually deliver a copy of the broadcast
packet to all nodes in the graph. Although this scheme is simple and elegant, it has a
fatal flaw (before you read on, see if you can figure out this fatal flaw): If the graph
has cycles, then one or more copies of each broadcast packet will cycle indefinitely.
For example, in Figure 4.43, R2 will flood to R3, R3 will flood to R4, R4 will flood
to R2, and R2 will flood (again!) to R3, and so on. This simple scenario results in
the endless cycling of two broadcast packets, one clockwise, and one counterclock-
wise. But there can be an even more calamitous fatal flaw: When a node is con-
nected to more than two other nodes, it will create and forward multiple copies of
the broadcast packet, each of which will create multiple copies of itself (at other
nodes with more than two neighbors), and so on. This broadcast storm, resulting
from the endless multiplication of broadcast packets, would eventually result in so
many broadcast packets being created that the network would be rendered useless.
(See the homework questions at the end of the chapter for a problem analyzing the
rate at which such a broadcast storm grows.)
Controlled Flooding
The key to avoiding a broadcast storm is for a node to judiciously choose when
to flood a packet and (e.g., if it has already received and flooded an earlier copy of
a packet) when not to flood a packet. In practice, this can be done in one of several
ways.
In sequence-number-controlled flooding, a source node puts its address (or
other unique identifier) as well as a broadcast sequence number into a broadcast
packet, then sends the packet to all of its neighbors. Each node maintains a list of
the source address and sequence number of each broadcast packet it has already
received, duplicated, and forwarded. When a node receives a broadcast packet, it
first checks whether the packet is in this list. If so, the packet is dropped; if not, the
4.7 BROADCAST AND MULTICAST ROUTING 401
packet is duplicated and forwarded to all the nodes neighbors (except the node from
which the packet has just been received). The Gnutella protocol, discussed in Chap-
ter 2, uses sequence-number-controlled flooding to broadcast queries in its overlay
network. (In Gnutella, message duplication and forwarding is performed at the
application layer rather than at the network layer.)
A second approach to controlled flooding is known as reverse path forwarding
(RPF) [Dalal 1978], also sometimes referred to as reverse path broadcast (RPB). The
idea behind RPF is simple, yet elegant. When a router receives a broadcast packet
with a given source address, it transmits the packet on all of its outgoing links (except
the one on which it was received) only if the packet arrived on the link that is on its
own shortest unicast path back to the source. Otherwise, the router simply discards
the incoming packet without forwarding it on any of its outgoing links. Such a packet
can be dropped because the router knows it either will receive or has already received
a copy of this packet on the link that is on its own shortest path back to the sender.
(You might want to convince yourself that this will, in fact, happen and that looping
and broadcast storms will not occur.) Note that RPF does not use unicast routing to
actually deliver a packet to a destination, nor does it require that a router know the
complete shortest path from itself to the source. RPF need only know the next neigh-
bor on its unicast shortest path to the sender; it uses this neighbors identity only to
determine whether or not to flood a received broadcast packet.
Figure 4.44 illustrates RPF. Suppose that the links drawn with thick lines repre-
sent the least-cost paths from the receivers to the source (A). Node A initially broad-
casts a source-A packet to nodes C and B. Node B will forward the source-A packet
it has received from A (since A is on its least-cost path to A) to both C and D. B will
ignore (drop, without forwarding) any source-A packets it receives from any other
402 CHAPTER 4 THE NETWORK LAYER
A
B
D
G
C
F E
Key:
pkt will be forwarded
pkt not forwarded beyond receiving router
Figure 4.44 Reverse path forwarding
4.7 BROADCAST AND MULTICAST ROUTING 403
a. Broadcast initiated at A b. Broadcast initiated at D
A
B
C
D
G
EF
A
B
C
D
G
EF
Figure 4.45 Broadcast along a spanning tree
nodes (for example, from routers C or D). Let us now consider node C, which will
receive a source-A packet directly from A as well as from B. Since B is not on Cs
own shortest path back to A, C will ignore any source-A packets it receives from B.
On the other hand, when C receives a source-A packet directly from A, it will for-
ward the packet to nodes B, E, and F.
Spanning-Tree Broadcast
While sequence-number-controlled flooding and RPF avoid broadcast storms, they
do not completely avoid the transmission of redundant broadcast packets. For exam-
ple, in Figure 4.44, nodes B, C, D, E, and F receive either one or two redundant
packets. Ideally, every node should receive only one copy of the broadcast packet.
Examining the tree consisting of the nodes connected by thick lines in Figure
4.45(a), you can see that if broadcast packets were forwarded only along links
within this tree, each and every network node would receive exactly one copy of the
broadcast packetexactly the solution we were looking for! This tree is an example
of a spanning treea tree that contains each and every node in a graph. More for-
mally, a spanning tree of a graph G = (N,E) is a graph G = (N,E) such that E is a
subset of E, G is connected, G contains no cycles, and G contains all the original
nodes in G. If each link has an associated cost and the cost of a tree is the sum of the
link costs, then a spanning tree whose cost is the minimum of all of the graphs
spanning trees is called (not surprisingly) a minimum spanning tree.
Thus, another approach to providing broadcast is for the network nodes to first
construct a spanning tree. When a source node wants to send a broadcast packet, it
sends the packet out on all of the incident links that belong to the spanning tree. A
node receiving a broadcast packet then forwards the packet to all its neighbors in the
spanning tree (except the neighbor from which it received the packet). Not only
does spanning tree eliminate redundant broadcast packets, but once in place, the
spanning tree can be used by any node to begin a broadcast, as shown in Figures
4.45(a) and 4.45(b). Note that a node need not be aware of the entire tree; it simply
needs to know which of its neighbors in G are spanning-tree neighbors.
The main complexity associated with the spanning-tree approach is the creation
and maintenance of the spanning tree. Numerous distributed spanning-tree algo-
rithms have been developed [Gallager 1983, Gartner 2003]. We consider only one
simple algorithm here. In the center-based approach to building a spanning tree, a
center node (also known as a rendezvous point or a core) is defined. Nodes then
unicast tree-join messages addressed to the center node. A tree-join message is for-
warded using unicast routing toward the center until it either arrives at a node that
already belongs to the spanning tree or arrives at the center. In either case, the path
that the tree-join message has followed defines the branch of the spanning tree
between the edge node that initiated the tree-join message and the center. One can
think of this new path as being grafted onto the existing spanning tree.
Figure 4.46 illustrates the construction of a center-based spanning tree. Suppose
that node E is selected as the center of the tree. Suppose that node F first joins the tree
and forwards a tree-join message to E. The single link EF becomes the initial span-
ning tree. Node B then joins the spanning tree by sending its tree-join message to E.
Suppose that the unicast path route to E from B is via D. In this case, the tree-join
message results in the path BDE being grafted onto the spanning tree. Node A next
joins the spanning group by forwarding its tree-join message towards E. If As uni-
cast path to E is through B, then since B has already joined the spanning tree, the
arrival of As tree-join message at B will result in the AB link being immediately
grafted onto the spanning tree. Node C joins the spanning tree next by forwarding
its tree-join message directly to E. Finally, because the unicast routing from G to E
404 CHAPTER 4 THE NETWORK LAYER
3
2
4
1
5
a. Stepwise construction of spanning tree b. Constructed spanning tree
A
B
C
D
G
EF
A
B
C
D
G
EF
Figure 4.46 Center-based construction of a spanning tree
must be via node D, when G sends its tree-join message to E, the GD link is grafted
onto the spanning tree at node D.
Broadcast Algorithms in Practice
Broadcast protocols are used in practice at both the application and network layers.
Gnutella [Gnutella 2009] uses application-level broadcast in order to broadcast
queries for content among Gnutella peers. Here, a link between two distributed
application-level peer processes in the Gnutella network is actually a TCP connec-
tion. Gnutella uses a form of sequence-number-controlled flooding in which a 16-
bit identifier and a 16-bit payload descriptor (which identifies the Gnutella message
type) are used to detect whether a received broadcast query has been previously
received, duplicated, and forwarded. Gnutella also uses a time-to-live (TTL) field to
limit the number of hops over which a flooded query will be forwarded. When a
Gnutella process receives and duplicates a query, it decrements the TTL field before
forwarding the query. Thus, a flooded Gnutella query will only reach peers that are
within a given number (the initial value of TTL) of application-level hops from the
query initiator. Gnutellas flooding mechanism is thus sometimes referred to as
limited-scope flooding.
A form of sequence-number-controlled flooding is also used to broadcast link-state
advertisements (LSAs) in the OSPF [RFC 2328, Perlman 1999] routing algorithm, and in
the Intermediate-System-to-Intermediate-System (IS-IS) routing algorithm [RFC
1142, Perlman 1999]. OSPF uses a 32-bit sequence number, as well as a 16-bit age field
to identify LSAs. Recall that an OSPF node broadcasts LSAs for its attached links peri-
odically, when a link cost to a neighbor changes, or when a link goes up/down. LSA
sequence numbers are used to detect duplicate LSAs, but also serve a second important
function in OSPF. With flooding, it is possible for an LSA generated by the source at
time t to arrive after a newer LSA that was generated by the same source at time t + d.
The sequence numbers used by the source node allow an older LSA to be distinguished
from a newer LSA. The age field serves a purpose similar to that of a TTL value. The ini-
tial age field value is set to zero and is incremented at each hop as it is flooded, and is also
incremented as it sits in a routers memory waiting to be flooded. Although we have only
briefly described the LSA flooding algorithm here, we note that designing LSA broadcast
protocols can be very tricky business indeed. [RFC 789; Perlman 1999] describe an inci-
dent in which incorrectly transmitted LSAs by two malfunctioning routers caused an
early version of an LSA flooding algorithm to take down the entire ARPAnet!
4.7.2 Multicast
Weve seen in the previous section that with broadcast service, packets are delivered
to each and every node in the network. In this section we turn our attention to
multicast service, in which a multicast packet is delivered to only a subset of
network nodes. A number of emerging network applications require the delivery of
packets from one or more senders to a group of receivers. These applications include
4.7 BROADCAST AND MULTICAST ROUTING 405
bulk data transfer (for example, the transfer of a software upgrade from the software
developer to users needing the upgrade), streaming continuous media (for example,
the transfer of the audio, video, and text of a live lecture to a set of distributed lec-
ture participants), shared data applications (for example, a whiteboard or teleconfer-
encing application that is shared among many distributed participants), data feeds
(for example, stock quotes), Web cache updating, and interactive gaming (for exam-
ple, distributed interactive virtual environments or multiplayer games).
In multicast communication, we are immediately faced with two problems
how to identify the receivers of a multicast packet and how to address a packet sent
to these receivers. In the case of unicast communication, the IP address of the
receiver (destination) is carried in each IP unicast datagram and identifies the single
recipient; in the case of broadcast, all nodes need to receive the broadcast packet, so
no destination addresses are needed. But in the case of multicast, we now have mul-
tiple receivers. Does it make sense for each multicast packet to carry the IP
addresses of all of the multiple recipients? While this approach might be workable
with a small number of recipients, it would not scale well to the case of hundreds or
thousands of receivers; the amount of addressing information in the datagram would
swamp the amount of data actually carried in the packets payload field. Explicit
identification of the receivers by the sender also requires that the sender know the
identities and addresses of all of the receivers. We will see shortly that there are
cases where this requirement might be undesirable.
For these reasons, in the Internet architecture (and other network architectures
such as ATM [Black 1995]), a multicast packet is addressed using address indirec-
tion. That is, a single identifier is used for the group of receivers, and a copy of the
packet that is addressed to the group using this single identifier is delivered to all of
the multicast receivers associated with that group. In the Internet, the single identifier
that represents a group of receivers is a class D multicast IP address. The group of
receivers associated with a class D address is referred to as a multicast group. The
multicast group abstraction is illustrated in Figure 4.47. Here, four hosts (shown in
shaded color) are associated with the multicast group address of 226.17.30.197 and
will receive all datagrams addressed to that multicast address. The difficulty that we
must still address is the fact that each host has a unique IP unicast address that is com-
pletely independent of the address of the multicast group in which it is participating.
While the multicast group abstraction is simple, it raises a host (pun intended)
of questions. How does a group get started and how does it terminate? How is the
group address chosen? How are new hosts added to the group (either as senders or
receivers)? Can anyone join a group (and send to, or receive from, that group) or is
group membership restricted and, if so, by whom? Do group members know the
identities of the other group members as part of the network-layer protocol? How
do the network nodes interoperate with each other to deliver a multicast datagram to
all group members? For the Internet, the answers to all of these questions involve
the Internet Group Management Protocol [RFC 3376]. So, let us next briefly con-
sider IGMP and then return to these broader questions.
406 CHAPTER 4 THE NETWORK LAYER
Internet Group Management Protocol
The IGMP protocol version 3 [RFC 3376] operates between a host and its directly
attached router (informally, we can think of the directly attached router as the first-
hop router that a host would see on a path to any other host outside its own local
network, or the last-hop router on any path to that host), as shown in Figure 4.48.
Figure 4.48 shows three first-hop multicast routers, each connected to its attached
hosts via one outgoing local interface. This local interface is attached to a LAN in
this example, and while each LAN has multiple attached hosts, at most a few of
these hosts will typically belong to a given multicast group at any given time.
IGMP provides the means for a host to inform its attached router that an application
running on the host wants to join a specific multicast group. Given that the scope of
IGMP interaction is limited to a host and its attached router, another protocol is clearly
required to coordinate the multicast routers (including the attached routers) throughout
4.7 BROADCAST AND MULTICAST ROUTING 407
128.119.40.186
Key:
Router with attached
group member
Router with no attached
group member
128.34.108.63
128.34.108.60
128.59.16.20
mcast group
226.17.30.197
Figure 4.47 The multicast group: A datagram addressed to the group is
delivered to all members of the multicast group
the Internet, so that multicast datagrams are routed to their final destinations. This latter
functionality is accomplished by network-layer multicast routing algorithms, such as
those we will consider shortly. Network-layer multicast in the Internet thus consists of
two complementary components: IGMP and multicast routing protocols.
IGMP has only three message types. Like ICMP, IGMP messages are carried
(encapsulated) within an IP datagram, with an IP protocol number of 2. The mem-
bership_query message is sent by a router to all hosts on an attached interface
(for example, to all hosts on a local area network) to determine the set of all multicast
groups that have been joined by the hosts on that interface. Hosts respond to a mem-
bership_query message with an IGMP membership_report message.
membership_report messages can also be generated by a host when an
application first joins a multicast group without waiting for a membership_query
message from the router. The final type of IGMP message is the leave_group
message. Interestingly, this message is optional. But if it is optional, how does a
router detect when a host leaves the multicast group? The answer to this question is
that the router infers that a host is no longer in the multicast group if it no longer
responds to a membership_query message with the given group address. This is
an example of what is sometimes called soft state in an Internet protocol. In a soft-
state protocol, the state (in this case of IGMP, the fact that there are hosts joined to a
given multicast group) is removed via a timeout event (in this case, via a periodic
membership_query message from the router) if it is not explicitly refreshed (in
this case, by a membership_report message from an attached host).
The term soft state was coined by Clark [Clark 1988], who described the notion
of periodic state refresh messages being sent by an end system, and suggested that
408 CHAPTER 4 THE NETWORK LAYER
Wide-area
multicast
routing
IGMP
IGMP
IGMP
IGMP
Figure 4.48 The two components of network-layer multicast in the
Internet: IGMP and multicast routing protocols
with such refresh messages, state could be lost in a crash and then automatically
restored by subsequent refresh messagesall transparently to the end system and
without invoking any explicit crash-recovery procedures:
. . . the state information would not be critical in maintaining the desired
type of service associated with the flow. Instead, that type of service would
be enforced by the end points, which would periodically send messages to
ensure that the proper type of service was being associated with the flow. In
this way, the state information associated with the flow could be lost in a
crash without permanent disruption of the service features being used. I call
this concept soft state, and it may very well permit us to achieve our pri-
mary goals of survivability and flexibility. . .
It has been argued that soft-state protocols result in simpler control than hard-
state protocols, which not only require state to be explicitly added and removed, but
also require mechanisms to recover from the situation where the entity responsible
for removing state has terminated prematurely or failed. Interesting discussions of
soft state can be found in [Raman 1999; Ji 2003; Lui 2004].
Multicast Routing Algorithms
The multicast routing problem is illustrated in Figure 4.49. Hosts joined to the mul-
ticast group are shaded in color; their immediately attached router is also shaded in
color. As shown in Figure 4.49, only a subset of routers (those with attached hosts that
are joined to the multicast group) actually needs to receive the multicast traffic. In Fig-
ure 4.49, only routers A, B, E, and F need to receive the multicast traffic. Since none
of the hosts attached to router D are joined to the multicast group and since router C
has no attached hosts, neither C nor D needs to receive the multicast group traffic. The
goal of multicast routing, then, is to find a tree of links that connects all of the routers
that have attached hosts belonging to the multicast group. Multicast packets will then
be routed along this tree from the sender to all of the hosts belonging to the multicast
tree. Of course, the tree may contain routers that do not have attached hosts belonging
to the multicast group (for example, in Figure 4.49, it is impossible to connect routers
A, B, E, and F in a tree without involving either router C or D).
In practice, two approaches have been adopted for determining the multicast
routing tree, both of which we have already studied in the context of broadcast
routing, and so we will only mention them in passing here. The two approaches dif-
fer according to whether a single group-shared tree is used to distribute the traffic
for all senders in the group, or whether a source-specific routing tree is constructed
for each individual sender.
Multicast routing using a group-shared tree. As in the case of spanning-tree broad-
cast, multicast routing over a group-shared tree is based on building a tree that
includes all edge routers with attached hosts belonging to the multicast group.
In practice, a center-based approach is used to construct the multicast routing tree,
with edge routers with attached hosts belonging to the multicast group sending
4.7 BROADCAST AND MULTICAST ROUTING 409
(via unicast) join messages addressed to the center node. As in the broadcast case, a
join message is forwarded using unicast routing toward the center until it either
arrives at a router that already belongs to the multicast tree or arrives at the center.
All routers along the path that the join message follows will then forward received
multicast packets to the edge router that initiated the multicast join. A critical ques-
tion for center-based tree multicast routing is the process used to select the center.
Center-selection algorithms are discussed in [Wall 1980; Thaler 1997; Estrin 1997].
Multicast routing using a source-based tree. While group-shared tree multicast
routing constructs a single, shared routing tree to route packets from all senders,
the second approach constructs a multicast routing tree for each source in the
multicast group. In practice, an RPF algorithm (with source node x) is used to
construct a multicast forwarding tree for multicast datagrams originating at
source x. The RPF broadcast algorithm we studied earlier requires a bit of tweak-
ing for use in multicast. To see why, consider router D in Figure 4.50. Under
broadcast RPF, it would forward packets to router G, even though router G has
no attached hosts that are joined to the multicast group. While this is not so bad
for this case where D has only a single downstream router, G, imagine what
would happen if there were thousands of routers downstream from D! Each
of these thousands of routers would receive unwanted multicast packets.
410 CHAPTER 4 THE NETWORK LAYER
A
C
F
B
D
E
Figure 4.49 Multicast hosts, their attached routers, and other routers
(This scenario is not as far-fetched as it might seem. The initial MBone [Casner
1992; Macedonia 1994], the first global multicast network, suffered from pre-
cisely this problem at first.). The solution to the problem of receiving unwanted
multicast packets under RPF is known as pruning. A multicast router that
receives multicast packets and has no attached hosts joined to that group will send
a prune message to its upstream router. If a router receives prune messages from
each of its downstream routers, then it can forward a prune message upstream.
Multicast Routing in the Internet
The first multicast routing protocol used in the Internet was the Distance-Vector Mul-
ticast Routing Protocol (DVMRP) [RFC 1075]. DVMRP implements source-based
trees with reverse path forwarding and pruning. DVMRP uses an RPF algorithm with
pruning, as discussed above. Perhaps the most widely used Internet multicast routing
protocol is the Protocol-Independent Multicast (PIM) routing protocol, which
explicitly recognizes two multicast distribution scenarios. In dense mode [RFC 3973],
multicast group members are densely located; that is, many or most of the routers in
the area need to be involved in routing multicast datagrams. PIM dense mode is a
flood-and-prune reverse path forwarding technique similar in spirit to DVMRP.
4.7 BROADCAST AND MULTICAST ROUTING 411
A
C
F
Key:
pkt will be forwarded
E G
B
S: source
D
pkt not forwarded beyond receiving router
Figure 4.50 Reverse path forwarding, the multicast case
In sparse mode [RFC 4601], the number of routers with attached group mem-
bers is small with respect to the total number of routers; group members are widely
dispersed. PIM sparse mode uses rendezvous points to set up the multicast distri-
bution tree. In source-specific multicast (SSM) [RFC 3569, RFC 4607], only a
single sender is allowed to send traffic into the multicast tree, considerably simpli-
fying tree construction and maintenance.
When PIM and DVMP are used within a domain, the network operator can con-
figure IP multicast routers within the domain, in much the same way that intra-
domain unicast routing protocols such as RIP, IS-IS, and OSPF can be configured.
But what happens when multicast routes are needed between different domains? Is
there a multicast equivalent of the inter-domain BGP protocol? The answer is (liter-
ally) yes. [RFC 4271] defines multiprotocol extensions to BGP to allow it to carry
routing information for other protocols, including multicast information. The Multi-
cast Source Discovery Protocol (MSDP) [RFC 3618, RFC 4611] can be used to con-
nect together rendezvous points in different PIM sparse mode domains. An excellent
overview of the current state of multicast routing in the Internet is [RFC 5110].
Let us close our discussion of IP multicast by noting that IP multicast has yet to
take off in a big way. For interesting discussions of the Internet multicast service
model and deployment issues, see [Diot 2000, Sharma 2003]. Nonetheless, in spite
of the lack of widespread deployment, network-level multicast is far from dead.
Multicast traffic has been carried for many years on Internet 2, and the networks
with which it peers [Internet2 Multicast 2012]. In the United Kingdom, the BBC is
engaged in trials of content distribution via IP multicast [BBC Multicast 2012]. At
the same time, application-level multicast, as we saw with PPLive in Chapter 2 and
in other peer-to-peer systems such as End System Multicast [Chu 2002], provides
multicast distribution of content among peers using application-layer (rather than
network-layer) multicast protocols. Will future multicast services be primarily
implemented in the network layer (in the network core) or in the application layer (at
the networks edge)? While the current craze for content distribution via peer-to-peer
approaches tips the balance in favor of application-layer multicast at least in the near-
term future, progress continues to be made in IP multicast, and sometimes the race
ultimately goes to the slow and steady.
4.8 Summary
In this chapter, we began our journey into the network core. We learned that the
network layer involves each and every host and router in the network. Because of
this, network-layer protocols are among the most challenging in the protocol stack.
We learned that a router may need to process millions of flows of packets
between different source-destination pairs at the same time. To permit a router to
process such a large number of flows, network designers have learned over the years
that the routers tasks should be as simple as possible. Many measures can be taken
412 CHAPTER 4 THE NETWORK LAYER
to make the routers job easier, including using a datagram network layer rather than
a virtual-circuit network layer, using a streamlined and fixed-sized header (as in
IPv6), eliminating fragmentation (also done in IPv6), and providing the one and
only best-effort service. Perhaps the most important trick here is not to keep track of
individual flows, but instead base routing decisions solely on hierarchically struc-
tured destination addresses in the datagrams. It is interesting to note that the postal
service has been using this approach for many years.
In this chapter, we also looked at the underlying principles of routing algorithms.
We learned how routing algorithms abstract the computer network to a graph with
nodes and links. With this abstraction, we can exploit the rich theory of shortest-path
routing in graphs, which has been developed over the past 40 years in the operations
research and algorithms communities. We saw that there are two broad approaches: a
centralized (global) approach, in which each node obtains a complete map of the net-
work and independently applies a shortest-path routing algorithm; and a decentral-
ized approach, in which individual nodes have only a partial picture of the entire
network, yet the nodes work together to deliver packets along the shortest routes. We
also studied how hierarchy is used to deal with the problem of scale by partitioning
large networks into independent administrative domains called autonomous systems
(ASs). Each AS independently routes its datagrams through the AS, just as each
country independently routes its postal mail through the country. We learned how
centralized, decentralized, and hierarchical approaches are embodied in the principal
routing protocols in the Internet: RIP, OSPF, and BGP. We concluded our study of
routing algorithms by considering broadcast and multicast routing.
Having completed our study of the network layer, our journey now takes us one
step further down the protocol stack, namely, to the link layer. Like the network layer,
the link layer is also part of the network core. But we will see in the next chapter that
the link layer has the much more localized task of moving packets between nodes on
the same link or LAN. Although this task may appear on the surface to be trivial com-
pared with that of the network layers tasks, we will see that the link layer involves a
number of important and fascinating issues that can keep us busy for a long time.
Homework Problems and Questions
Chapter 4 Review Questions
SECTIONS 4.14.2
R1. Lets review some of the terminology used in this textbook. Recall that the
name of a transport-layer packet is segment and that the name of a link-layer
packet is frame. What is the name of a network-layer packet? Recall that both
routers and link-layer switches are called packet switches. What is the
fundamental difference between a router and link-layer switch? Recall that
we use the term routers for both datagram networks and VC networks.
HOMEWORK PROBLEMS AND QUESTIONS 413
R2. What are the two most important network-layer functions in a datagram net-
work? What are the three most important network-layer functions in a virtual-
circuit network?
R3. What is the difference between routing and forwarding?
R4. Do the routers in both datagram networks and virtual-circuit networks use for-
warding tables? If so, describe the forwarding tables for both classes of networks.
R5. Describe some hypothetical services that the network layer can provide to a
single packet. Do the same for a flow of packets. Are any of your hypotheti-
cal services provided by the Internets network layer? Are any provided by
ATMs CBR service model? Are any provided by ATMs ABR service
model?
R6. List some applications that would benefit from ATMs CBR service model.
SECTION 4.3
R7. Discuss why each input port in a high-speed router stores a shadow copy of
the forwarding table.
R8. Three types of switching fabrics are discussed in Section 4.3. List and briefly
describe each type. Which, if any, can send multiple packets across the fabric
in parallel?
R9. Describe how packet loss can occur at input ports. Describe how packet loss
at input ports can be eliminated (without using infinite buffers).
R10. Describe how packet loss can occur at output ports. Can this loss be
prevented by increasing the switch fabric speed?
R11. What is HOL blocking? Does it occur in input ports or output ports?
SECTION 4.4
R12. Do routers have IP addresses? If so, how many?
R13. What is the 32-bit binary equivalent of the IP address 223.1.3.27?
R14. Visit a host that uses DHCP to obtain its IP address, network mask, default
router, and IP address of its local DNS server. List these values.
R15. Suppose there are three routers between a source host and a destination host.
Ignoring fragmentation, an IP datagram sent from the source host to the desti-
nation host will travel over how many interfaces? How many forwarding tables
will be indexed to move the datagram from the source to the destination?
R16. Suppose an application generates chunks of 40 bytes of data every 20 msec,
and each chunk gets encapsulated in a TCP segment and then an IP datagram.
What percentage of each datagram will be overhead, and what percentage
will be application data?
R17. Suppose Host A sends Host B a TCP segment encapsulated in an IP datagram.
When Host B receives the datagram, how does the network layer in Host B
414 CHAPTER 4 THE NETWORK LAYER
know it should pass the segment (that is, the payload of the datagram) to TCP
rather than to UDP or to something else?
R18. Suppose you purchase a wireless router and connect it to your cable modem.
Also suppose that your ISP dynamically assigns your connected device (that
is, your wireless router) one IP address. Also suppose that you have five PCs
at home that use 802.11 to wirelessly connect to your wireless router. How
are IP addresses assigned to the five PCs? Does the wireless router use NAT?
Why or why not?
R19. Compare and contrast the IPv4 and the IPv6 header fields. Do they have any
fields in common?
R20. It has been said that when IPv6 tunnels through IPv4 routers, IPv6 treats the
IPv4 tunnels as link-layer protocols. Do you agree with this statement? Why
or why not?
SECTION 4.5
R21. Compare and contrast link-state and distance-vector routing algorithms.
R22. Discuss how a hierarchical organization of the Internet has made it possible
to scale to millions of users.
R23. Is it necessary that every autonomous system use the same intra-AS routing
algorithm? Why or why not?
SECTION 4.6
R24. Consider Figure 4.37. Starting with the original table in D, suppose that D
receives from A the following advertisement:
HOMEWORK PROBLEMS AND QUESTIONS 415
Will the table in D change? If so how?
R25. Compare and contrast the advertisements used by RIP and OSPF.
R26. Fill in the blank: RIP advertisements typically announce the number of hops
to various destinations. BGP updates, on the other hand, announce the
__________ to the various destinations.
R27. Why are different inter-AS and intra-AS protocols used in the Internet?
R28. Why are policy considerations as important for intra-AS protocols, such as
OSPF and RIP, as they are for an inter-AS routing protocol like BGP?
Destination Subnet Next Router Number of Hops to Destination
z C 10
w 1
x 1
. . . . . . . . . . . .
R29. Define and contrast the following terms: subnet, prefix, and BGP route.
R30. How does BGP use the NEXT-HOP attribute? How does it use the AS-PATH
attribute?
R31. Describe how a network administrator of an upper-tier ISP can implement
policy when configuring BGP.
SECTION 4.7
R32. What is an important difference between implementing the broadcast abstrac-
tion via multiple unicasts, and a single network- (router-) supported broad-
cast?
R33. For each of the three general approaches we studied for broadcast communi-
cation (uncontrolled flooding, controlled flooding, and spanning-tree broad-
cast), are the following statements true or false? You may assume that no
packets are lost due to buffer overflow and all packets are delivered on a link
in the order in which they were sent.
a. A node may receive multiple copies of the same packet.
b. A node may forward multiple copies of a packet over the same
outgoing link.
R34. When a host joins a multicast group, must it change its IP address to that of
the multicast group it is joining?
R35. What are the roles played by the IGMP protocol and a wide-area multicast
routing protocol?
R36. What is the difference between a group-shared tree and a source-based tree in
the context of multicast routing?
Problems
P1. In this question, we consider some of the pros and cons of virtual-circuit and
datagram networks.
a. Suppose that routers were subjected to conditions that might cause them
to fail fairly often.Would this argue in favor of a VC or datagram archi-
tecture?Why?
b. Suppose that a source node and a destination require that a fixed amount
of capacity always be available at all routers on the path between the
source and destination node, for the exclusive use of traffic flowing
between this source and destination node. Would this argue in favor of a
VC or datagram architecture?Why?
c. Suppose that the links and routers in the network never fail and that rout-
ing paths used between all source/destination pairs remains constant.In
this scenario, does a VC or datagram architecture have more control traf-
fic overhead?Why?
416 CHAPTER 4 THE NETWORK LAYER
P2. Consider a virtual-circuit network. Suppose the VC number is an 8-bit field.
a. What is the maximum number of virtual circuits that can be carried over a
link?
b. Suppose a central node determines paths and VC numbers at connection
setup. Suppose the same VC number is used on each link along the VCs
path. Describe how the central node might determine the VC number at con-
nection setup. Is it possible that there are fewer VCs in progress than the
maximum as determined in part (a) yet there is no common free VC number?
c. Suppose that different VC numbers are permitted in each link along a
VCs path. During connection setup, after an end-to-end path is determined,
describe how the links can choose their VC numbers and configure their for-
warding tables in a decentralized manner, without reliance on a central node.
P3. A bare-bones forwarding table in a VC network has four columns. What is
the meaning of the values in each of these columns? A bare-bones forwarding
table in a datagram network has two columns. What is the meaning of the
values in each of these columns?
P4. Consider the network below.
a. Suppose that this network is a datagram network.Show the forwarding
table in router A, such that all traffic destined to host H3 is forwarded
through interface 3.
b. Suppose that this network is a datagram network.Can you write down a
forwarding table in router A, such that all traffic from H1 destined to host
H3 is forwarded through interface 3, while all traffic from H2 destined to
host H3 is forwarded through interface 4?(Hint: this is a trick question.)
c. Now suppose that this network is a virtual circuit network and that there is
one ongoing call between H1 and H3, and another ongoing call between
H2 and H3. Write down a forwarding table in router A, such that all traffic
from H1 destined to host H3 is forwarded through interface 3, while all
traffic from H2 destined to host H3 is forwarded through interface 4.
d. Assuming the same scenario as (c), write down the forwarding tables in
nodes B, C, and D.
PROBLEMS 417
B
A
1 3
2 4
2
D
1
2
3
H3
H1
H2
1
1 2
C
P5. Consider a VC network with a 2-bit field for the VC number. Suppose that
the network wants to set up a virtual circuit over four links: link A, link B,
link C, and link D. Suppose that each of these links is currently carrying two
other virtual circuits, and the VC numbers of these other VCs are as follows:
418 CHAPTER 4 THE NETWORK LAYER
Link A Link B Link C Link D
00 01 10 11
01 10 11 00
In answering the following questions, keep in mind that each of the existing
VCs may only be traversing one of the four links.
a. If each VC is required to use the same VC number on all links along its
path, what VC number could be assigned to the new VC?
b. If each VC is permitted to have different VC numbers in the different links
along its path (so that forwarding tables must perform VC number transla-
tion), how many different combinations of four VC numbers (one for each
of the four links) could be used?
P6. In the text we have used the term connection-oriented service to describe a
transport-layer service and connection service for a network-layer service.
Why the subtle shades in terminology?
P7. Suppose two packets arrive to two different input ports of a router at exactly
the same time. Also suppose there are no other packets anywhere in the
router.
a. Suppose the two packets are to be forwarded to two different output ports.
Is it possible to forward the two packets through the switch fabric at the
same time when the fabric uses a shared bus?
b. Suppose the two packets are to be forwarded to two different output ports.
Is it possible to forward the two packets through the switch fabric at the
same time when the fabric uses a crossbar?
c. Suppose the two packets are to be forwarded to the same output port. Is it
possible to forward the two packets through the switch fabric at the same
time when the fabric uses a crossbar?
P8. In Section 4.3, we noted that the maximum queuing delay is (n1)D if the
switching fabric is n times faster than the input line rates. Suppose that all
packets are of the same length, n packets arrive at the same time to the n
input ports, and all n packets want to be forwarded to different output ports.
What is the maximum delay for a packet for the (a) memory, (b) bus, and (c)
crossbar switching fabrics?
P9. Consider the switch shown below. Suppose that all datagrams have the same
fixed length, that the switch operates in a slotted, synchronous manner, and
that in one time slot a datagram can be transferred from an input port to an
output port. The switch fabric is a crossbar so that at most one datagram can
PROBLEMS 419
be transferred to a given output port in a time slot, but different output ports
can receive datagrams from different input ports in a single time slot. What is
the minimal number of time slots needed to transfer the packets shown from
input ports to their output ports, assuming any input queue scheduling order
you want (i.e., it need not have HOL blocking)?What is the largest number
of slots needed, assuming the worst-case scheduling order you can devise,
assuming that a non-empty input queue is never idle?
X Y
Switch
fabric
Output port X
Output port Y
Output port Z
X
YZ
P10. Consider a datagram network using 32-bit host addresses. Suppose a router
has four links, numbered 0 through 3, and packets are to be forwarded to the
link interfaces as follows:
Destination Address RangeLink Interface
11100000 00000000 00000000 00000000
through 0
11100000 00111111 11111111 11111111
11100000 01000000 00000000 00000000
through 1
11100000 01000000 11111111 11111111
11100000 01000001 00000000 00000000
through 2
11100001 01111111 11111111 11111111
otherwise 3
a. Provide a forwarding table that has five entries, uses longest prefix match-
ing, and forwards packets to the correct link interfaces.
b. Describe how your forwarding table determines the appropriate link inter-
face for datagrams with destination addresses:
11001000 10010001 01010001 01010101
11100001 01000000 11000011 00111100
11100001 10000000 00010001 01110111
420 CHAPTER 4 THE NETWORK LAYER
Prefix Match Interface
1 0
10 1
111 2
otherwise 3
For each of the four interfaces, give the associated range of destination host
addresses and the number of addresses in the range.
P13. Consider a router that interconnects three subnets: Subnet 1, Subnet 2, and
Subnet 3. Suppose all of the interfaces in each of these three subnets are
required to have the prefix 223.1.17/24. Also suppose that Subnet 1 is
required to support at least 60 interfaces, Subnet 2 is to support at least 90
interfaces, and Subnet 3 is to support at least 12 interfaces. Provide three net-
work addresses (of the form a.b.c.d/x) that satisfy these constraints.
P14. In Section 4.2.2 an example forwarding table (using longest prefix matching)
is given. Rewrite this forwarding table using the a.b.c.d/x notation instead of
the binary string notation.
P15. In Problem P10 you are asked to provide a forwarding table (using longest
prefix matching). Rewrite this forwarding table using the a.b.c.d/x notation
instead of the binary string notation.
P11. Consider a datagram network using 8-bit host addresses. Suppose a router
uses longest prefix matching and has the following forwarding table:
Prefix Match Interface
00 0
010 1
011 2
10 2
11 3
For each of the four interfaces, give the associated range of destination host
addresses and the number of addresses in the range.
P12. Consider a datagram network using 8-bit host addresses. Suppose a
router uses longest prefix matching and has the following forwarding
table:
P16. Consider a subnet with prefix 128.119.40.128/26. Give an example of one
IP address (of form xxx.xxx.xxx.xxx) that can be assigned to this network.
Suppose an ISP owns the block of addresses of the form 128.119.40.64/26.
Suppose it wants to create four subnets from this block, with each block
having the same number of IP addresses. What are the prefixes (of form
a.b.c.d/x) for the four subnets?
P17. Consider the topology shown in Figure 4.17. Denote the three subnets with
hosts (starting clockwise at 12:00) as Networks A, B, and C. Denote the sub-
nets without hosts as Networks D, E, and F.
a. Assign network addresses to each of these six subnets, with the follow-
ing constraints: All addresses must be allocated from 214.97.254/23;
Subnet A should have enough addresses to support 250 interfaces; Sub-
net B should have enough addresses to support 120 interfaces; and
Subnet C should have enough addresses to support 120 interfaces. Of
course, subnets D, E and F should each be able to support two interfaces.
For each subnet, the assignment should take the form a.b.c.d/x or
a.b.c.d/x e.f.g.h/y.
b. Using your answer to part (a), provide the forwarding tables (using longest
prefix matching) for each of the three routers.
P18. Use the whois service at the American Registry for Internet Numbers
(http://www.arin.net/whois) to determine the IP address blocks for three
universities. Can the whois services be used to determine with certainty the
geographical location of a specific IP address? Use www.maxmind.com to
determine the locations of the Web servers at each of these universities.
P19. Consider sending a 2400-byte datagram into a link that has an MTU of
700 bytes. Suppose the original datagram is stamped with the identifica-
tion number 422. How many fragments are generated? What are the
values in the various fields in the IP datagram(s) generated related to
fragmentation?
P20. Suppose datagrams are limited to 1,500 bytes (including header) between
source Host A and destination Host B. Assuming a 20-byte IP header, how
many datagrams would be required to send an MP3 consisting of 5 million
bytes? Explain how you computed your answer.
P21. Consider the network setup in Figure 4.22. Suppose that the ISP instead
assigns the router the address 24.34.112.235 and that the network address of
the home network is 192.168.1/24.
a. Assign addresses to all interfaces in the home network.
b. Suppose each host has two ongoing TCP connections, all to port 80 at
host 128.119.40.86. Provide the six corresponding entries in the NAT
translation table.
PROBLEMS 421
http://www.arin.net/whois
www.maxmind.com
P22. Suppose you are interested in detecting the number of hosts behind a NAT.
You observe that the IP layer stamps an identification number sequentially on
each IP packet. The identification number of the first IP packet generated by
a host is a random number, and the identification numbers of the subsequent
IP packets are sequentially assigned. Assume all IP packets generated by
hosts behind the NAT are sent to the outside world.
a. Based on this observation, and assuming you can sniff all packets sent by
the NAT to the outside, can you outline a simple technique that detects the
number of unique hosts behind a NAT? Justify your answer.
b. If the identification numbers are not sequentially assigned but randomly
assigned, would your technique work? Justify your answer.
P23. In this problem well explore the impact of NATs on P2P applications.
Suppose a peer with username Arnold discovers through querying that a peer
with username Bernard has a file it wants to download. Also suppose that
Bernard and Arnold are both behind a NAT. Try to devise a technique that
will allow Arnold to establish a TCP connection with Bernard without
application-specific NAT configuration. If you have difficulty devising such
a technique, discuss why.
P24. Looking at Figure 4.27, enumerate the paths from y to u that do not contain
any loops.
P25. Repeat Problem P24 for paths from x to z, z to u, and z to w.
P26. Consider the following network. With the indicated link costs, use Dijkstras
shortest-path algorithm to compute the shortest path from x to all network
nodes. Show how the algorithm works by computing a table similar to
Table 4.3.
x
v
ty
z
u
w
6
12
8
7
8
3
6
4
3
2
4
3
422 CHAPTER 4 THE NETWORK LAYER
VideoNote
Dijkstras algorithm:
discussion and example
P27. Consider the network shown in Problem P26. Using Dijkstras algorithm, and
showing your work using a table similar to Table 4.3, do the following:
a. Compute the shortest path from t to all network nodes.
b. Compute the shortest path from u to all network nodes.
c. Compute the shortest path from v to all network nodes.
d. Compute the shortest path from w to all network nodes.
e. Compute the shortest path from y to all network nodes.
f. Compute the shortest path from z to all network nodes.
P28. Consider the network shown below, and assume that each node initially
knows the costs to each of its neighbors. Consider the distance-vector
algorithm and show the distance table entries at node z.
P29. Consider a general topology (that is, not the specific network shown above) and a
synchronous version of the distance-vector algorithm. Suppose that at each itera-
tion, a node exchanges its distance vectors with its neighbors and receives their
distance vectors. Assuming that the algorithm begins with each node knowing
only the costs to its immediate neighbors, what is the maximum number of itera-
tions required before the distributed algorithm converges? Justify your answer.
P30. Consider the network fragment shown below. x has only two attached neigh-
bors, w and y. w has a minimum-cost path to destination u (not shown) of 5,
and y has a minimum-cost path to u of 6. The complete paths from w and y to
u (and between w and y) are not shown. All link costs in the network have
strictly positive integer values.
x y
w
2 2
5
u
z
v
y
2 3
6
2
3
1
x
PROBLEMS 423
a. Give xs distance vector for destinations w, y, and u.
b. Give a link-cost change for either c(x,w) or c(x,y) such that x will inform
its neighbors of a new minimum-cost path to u as a result of executing the
distance-vector algorithm.
c. Give a link-cost change for either c(x,w) or c(x,y) such that x will not
inform its neighbors of a new minimum-cost path to u as a result of exe-
cuting the distance-vector algorithm.
P31. Consider the three-node topology shown in Figure 4.30. Rather than having
the link costs shown in Figure 4.30, the link costs are c(x,y) = 3, c(y,z) = 6,
c(z,x) = 4. Compute the distance tables after the initialization step and after
each iteration of a synchronous version of the distance-vector algorithm (as
we did in our earlier discussion of Figure 4.30).
P32. Consider the count-to-infinity problem in the distance vector routing. Will the
count-to-infinity problem occur if we decrease the cost of a link? Why? How
about if we connect two nodes which do not have a link?
P33. Argue that for the distance-vector algorithm in Figure 4.30, each value in the
distance vector D(x) is non-increasing and will eventually stabilize in a finite
number of steps.
P34. Consider Figure 4.31. Suppose there is another router w, connected to router
y and z. The costs of all links are given as follows: c(x,y) = 4, c(x,z) = 50,
c(y,w) = 1, c(z,w) = 1, c(y,z) = 3. Suppose that poisoned reverse is used in the
distance-vector routing algorithm.
a. When the distance vector routing is stabilized, router w, y, and z inform their
distances to x to each other. What distance values do they tell each other?
b. Now suppose that the link cost between x and y increases to 60. Will there
be a count-to-infinity problem even if poisoned reverse is used? Why or
why not? If there is a count-to-infinity problem, then how many iterations
are needed for the distance-vector routing to reach a stable state again?
Justify your answer.
c. How do you modify c(y,z) such that there is no count-to-infinity problem
at all if c(y,x) changes from 4 to 60?
P35. Describe how loops in paths can be detected in BGP.
P36. Will a BGP router always choose the loop-free route with the shortest AS-
path length? Justify your answer.
P37. Consider the network shown below. Suppose AS3 and AS2 are running OSPF
for their intra-AS routing protocol. Suppose AS1 and AS4 are running RIP
for their intra-AS routing protocol. Suppose eBGP and iBGP are used for the
inter-AS routing protocol. Initially suppose there is no physical link between
AS2 and AS4.
424 CHAPTER 4 THE NETWORK LAYER
a. Router 3c learns about prefix x from which routing protocol: OSPF, RIP,
eBGP, or iBGP?
b. Router 3a learns about x from which routing protocol?
c. Router 1c learns about x from which routing protocol?
d. Router 1d learns about x from which routing protocol?
P38. Referring to the previous problem, once router 1d learns about x it will put an
entry (x, I) in its forwarding table.
a. Will I be equal to I1 or I2 for this entry? Explain why in one sentence.
b. Now suppose that there is a physical link between AS2 and AS4, shown by
the dotted line. Suppose router 1d learns that x is accessible via AS2 as
well as via AS3. Will I be set to I1 or I2? Explain why in one sentence.
c. Now suppose there is another AS, called AS5, which lies on the path
between AS2 and AS4 (not shown in diagram). Suppose router 1d learns
that x is accessible via AS2 AS5 AS4 as well as via AS3 AS4. Will I be set
to I1 or I2? Explain why in one sentence.
P39. Consider the following network. ISP B provides national backbone service
to regional ISP A. ISP C provides national backbone service to regional
ISP D. Each ISP consists of one AS. B and C peer with each other in two
places using BGP. Consider traffic going from A to D. B would prefer to
hand that traffic over to C on the West Coast (so that C would have to
absorb the cost of carrying the traffic cross-country), while C would
prefer to get the traffic via its East Coast peering point with B (so that B
would have carried the traffic across the country). What BGP mechanism
might C use, so that B would hand over A-to-D traffic at its East Coast
AS4
AS3
AS1
AS2
x
4b
4c 4a
3c
3b
3a
1c
1b
1d
1a
I1 I2
2c
2a
2b
PROBLEMS 425
peering point? To answer this question, you will need to dig into the BGP
specification.
P40. In Figure 4.42, consider the path information that reaches stub networks W,
X, and Y. Based on the information available at W and X, what are their
respective views of the network topology? Justify your answer. The topology
view at Y is shown below.
P41. Consider Figure 4.42. B would never forward traffic destined to Y via X
based on BGP routing. But there are some very popular applications for
which data packets go to X first and then flow to Y.Identify one such
application, and describe how data packets follow a path not given by
BGP routing.
P42. In Figure 4.42, suppose that there is another stub network V that is a customer of
ISP A. Suppose that B and C have a peering relationship, and A is a customer of
both B and C. Suppose that A would like to have the traffic destined to W to
come from B only, and the traffic destined to V from either B or C. How should
A advertise its routes to B and C? What AS routes does C receive?
P43. Suppose ASs X and Z are not directly connected but instead are connected by
AS Y. Further suppose that X has a peering agreement with Y, and that Y has
W
Y
X
A
C
Stub network
Ys view of
the topology
ISP B
ISP C
ISP D
ISP A
426 CHAPTER 4 THE NETWORK LAYER
a peering agreement with Z. Finally, suppose that Z wants to transit all of Ys
traffic but does not want to transit Xs traffic. Does BGP allow Z to imple-
ment this policy?
P44. Consider the seven-node network (with nodes labeled t to z) in Problem P26.
Show the minimal-cost tree rooted at z that includes (as end hosts) nodes u, v,
w, and y. Informally argue why your tree is a minimal-cost tree.
P45. Consider the two basic approaches identified for achieving broadcast, unicast
emulation and network-layer (i.e., router-assisted) broadcast, and suppose
spanning-tree broadcast is used to achive network-layer broadcast. Consider
a single sender and 32 receivers. Suppose the sender is connected to the
receivers by a binary tree of routers. What is the cost of sending a broadcast
packet, in the cases of unicast emulation and network-layer broadcast, for this
topology? Here, each time a packet (or copy of a packet) is sent over a single
link, it incurs a unit of cost. What topology for interconnecting the sender,
receivers, and routers will bring the cost of unicast emulation and true net-
work-layer broadcast as far apart as possible? You can choose as many
routers as youd like.
P46. Consider the operation of the reverse path forwarding (RPF) algorithm in Figure
4.44. Using the same topology, find a set of paths from all nodes to the source
node A (and indicate these paths in a graph using thicker-shaded lines as in Fig-
ure 4.44) such that if these paths were the least-cost paths, then node B would
receive a copy of As broadcast message from nodes A, C, and D under RPF.
P47. Consider the topology shown in Figure 4.44. Suppose that all links have unit
cost and that node E is the broadcast source. Using arrows like those shown
in Figure 4.44 indicate links over which packets will be forwarded using
RPF, and links over which packets will not be forwarded, given that node E is
the source.
P48. Repeat Problem P47 using the graph from Problem P26. Assume that z is the
broadcast source, and that the link costs are as shown in Problem P26.
P49. Consider the topology shown in Figure 4.46, and suppose that each link has
unit cost. Suppose node C is chosen as the center in a center-based multicast
routing algorithm. Assuming that each attached router uses its least-cost path
to node C to send join messages to C, draw the resulting center-based routing
tree. Is the resulting tree a minimum-cost tree? Justify your answer.
P50. Repeat Problem P49, using the graph from Problem P26. Assume that the
center node is v.
P51. In Section 4.5.1 we studied Dijkstras link-state routing algorithm for com-
puting the unicast paths that are individually the least-cost paths from the
source to all destinations. The union of these paths might be thought of as
forming a least-unicast-cost path tree (or a shortest unicast path tree, if
all link costs are identical). By constructing a counterexample, show that
the least-cost path tree is not always the same as a minimum spanning tree.
PROBLEMS 427
P52. Consider a network in which all nodes are connected to three other nodes. In
a single time step, a node can receive all transmitted broadcast packets from
its neighbors, duplicate the packets, and send them to all of its neighbors
(except to the node that sent a given packet). At the next time step, neighboring
nodes can receive, duplicate, and forward these packets, and so on. Sup-
pose that uncontrolled flooding is used to provide broadcast in such a
network. At time step t, how many copies of the broadcast packet will be
transmitted, assuming that during time step 1, a single broadcast packet is
transmitted by the source node to its three neighbors.
P53. We saw in Section 4.7 that there is no network-layer protocol that can be used
to identify the hosts participating in a multicast group. Given this, how can
multicast applications learn the identities of the hosts that are participating in
a multicast group?
P54. Design (give a pseudocode description of) an application-level protocol that
maintains the host addresses of all hosts participating in a multicast group.
Specifically identify the network service (unicast or multicast) that is used by
your protocol, and indicate whether your protocol is sending messages in-
band or out-of-band (with respect to the application data flow among the
multicast group participants) and why.
P55. What is the size of the multicast address space? Suppose now that two multi-
cast groups randomly choose a multicast address. What is the probability that
they choose the same address? Suppose now that 1,000 multicast groups are
ongoing at the same time and choose their multicast group addresses at ran-
dom. What is the probability that they interfere with each other?
Socket Programming Assignment
At the end of Chapter 2, there are four socket programming assignments. Below,
you will find a fifth assignment which employs ICMP, a protocol discussed in this
chapter.
Assignment 5: ICMP Ping
Ping is a popular networking application used to test from a remote location whether
a particular host is up and reachable. It is also often used to measure latency
between the client host and the target host. It works by sending ICMP echo
request packets (i.e., ping packets) to the target host and listening for ICMP echo
response replies (i.e., pong packets). Ping measures the RRT, records packet loss,
and calculates a statistical summary of multiple ping-pong exchanges (the mini-
mum, mean, max, and standard deviation of the round-trip times).
428 CHAPTER 4 THE NETWORK LAYER
In this lab, you will write your own Ping application in Python. Your application
will use ICMP. But in order to keep your program simple, you will not exactly follow
the official specification in RFC 1739. Note that you will only need to write the client
side of the program, as the functionality needed on the server side is built into almost
all operating systems. You can find full details of this assignment, as well as impor-
tant snippets of the Python code, at the Web site http://www.awl.com/kurose-ross.
Programming Assignment
In this programming assignment, you will be writing a distributed set of proce-
dures that implements a distributed asynchronous distance-vector routing for the
network shown below.
You are to write the following routines that will execute asynchronously
within the emulated environment provided for this assignment. For node 0, you will
write the routines:
rtinit0(). This routine will be called once at the beginning of the emulation.
rtinit0() has no arguments. It should initialize your distance table in node 0 to
reflect the direct costs of 1, 3, and 7 to nodes 1, 2, and 3, respectively. In the fig-
ure above, all links are bidirectional and the costs in both directions are identi-
cal. After initializing the distance table and any other data structures needed by
your node 0 routines, it should then send its directly connected neighbors (in
this case, 1, 2, and 3) the cost of its minimum-cost paths to all other network
nodes. This minimum-cost information is sent to neighboring nodes in a routing
update packet by calling the routine tolayer2(), as described in the full assign-
ment. The format of the routing update packet is also described in the full
assignment.
rtupdate0(struct rtpkt *rcvdpkt). This routine will be called when node 0
receives a routing packet that was sent to it by one of its directly connected
neighbors. The parameter *rcvdpkt is a pointer to the packet that was received.
rtupdate0() is the heart of the distance-vector algorithm. The values it
receives in a routing update packet from some other node i contain is current
shortest-path costs to all other network nodes. rtupdate0() uses these received
3 2
0
1
7
3
1
2
1
PROGRAMMING ASSIGNMENT 429
http://www.awl.com/kurose-ross
values to update its own distance table (as specified by the distance-vector algo-
rithm). If its own minimum cost to another node changes as a result of the
update, node 0 informs its directly connected neighbors of this change in mini-
mum cost by sending them a routing packet. Recall that in the distance-vector
algorithm, only directly connected nodes will exchange routing packets. Thus,
nodes 1 and 2 will communicate with each other, but nodes 1 and 3 will not
communicate with each other.
Similar routines are defined for nodes 1, 2, and 3. Thus, you will write eight proce-
dures in all: rtinit0(), rtinit1(), rtinit2(), rtinit3(), rtupdate0(), rtupdate1(), rtup-
date2(), and rtupdate3(). These routines will together implement a distributed,
asynchronous computation of the distance tables for the topology and costs shown
in the figure on the preceding page.
You can find the full details of the programming assignment, as well as C code
that you will need to create the simulated hardware/software environment, at
http://www.awl.com/kurose-ross. A Java version of the assignment is also available.
Wireshark Labs
In the companion Web site for this textbook, http://www.awl.com/kurose-ross,
youll find two Wireshark lab assignments. The first lab examines the operation of
the IP protocol, and the IP datagram format in particular. The second lab explores
the use of the ICMP protocol in the ping and traceroute commands.
430 CHAPTER 4 THE NETWORK LAYER
http://www.awl.com/kurose-ross
http://www.awl.com/kurose-ross
What brought you to specialize in networking?
I was working as a programmer at UCLA in the late 1960s. My job was supported by the
US Defense Advanced Research Projects Agency (called ARPA then, called DARPA now). I
was working in the laboratory of Professor Leonard Kleinrock on the Network
Measurement Center of the newly created ARPAnet. The first node of the ARPAnet was
installed at UCLA on September 1, 1969. I was responsible for programming a computer
that was used to capture performance information about the ARPAnet and to report this
information back for comparison with mathematical models and predictions of the perform-
ance of the network.
Several of the other graduate students and I were made responsible for working on
the so-called host-level protocols of the ARPAnetthe procedures and formats that would
allow many different kinds of computers on the network to interact with each other. It was a
fascinating exploration into a new world (for me) of distributed computing and communication.
Did you imagine that IP would become as pervasive as it is today when you first designed
the protocol?
When Bob Kahn and I first worked on this in 1973, I think we were mostly very focused on
the central question: How can we make heterogeneous packet networks interoperate with
one another, assuming we cannot actually change the networks themselves? We hoped that
we could find a way to permit an arbitrary collection of packet-switched networks to be
interconnected in a transparent fashion, so that host computers could communicate end-to-
end without having to do any translations in between. I think we knew that we were dealing
with powerful and expandable technology but I doubt we had a clear image of what the
world would be like with hundreds of millions of computers all interlinked on the Internet.
431
Vinton G. Cerf
Vinton G. Cerf is Vice President and Chief Internet Evangelist for
Google. He served for over 16 years at MCI in various positions,
ending up his tenure there as Senior Vice President for Technology
Strategy. He is widely known as the co-designer of the TCP/IP
protocols and the architecture of the Internet. During his time from
1976 to 1982 at the US Department of Defense Advanced
Research Projects Agency (DARPA), he played a key role leading the
development of Internet and Internet-related data packet and security
techniques. He received the US Presidential Medal of Freedom in
2005 and the US National Medal of Technology in 1997. He
holds a BS in Mathematics from Stanford University and an MS and
PhD in computer science from UCLA.
AN INTERVIEW WITH
What do you now envision for the future of networking and the Internet? What major
challenges/obstacles do you think lie ahead in their development?
I believe the Internet itself and networks in general will continue to proliferate. Already
there is convincing evidence that there will be billions of Internet-enabled devices on the
Internet, including appliances like cell phones, refrigerators, personal digital assistants,
home servers, televisions, as well as the usual array of laptops, servers, and so on. Big chal-
lenges include support for mobility, battery life, capacity of the access links to the network,
and ability to scale the optical core of the network up in an unlimited fashion. Designing an
interplanetary extension of the Internet is a project in which I am deeply engaged at the Jet
Propulsion Laboratory. We will need to cut over from IPv4 [32-bit addresses] to IPv6 [128
bits]. The list is long!
Who has inspired you professionally?
My colleague Bob Kahn; my thesis advisor, Gerald Estrin; my best friend, Steve Crocker
(we met in high school and he introduced me to computers in 1960!); and the thousands of
engineers who continue to evolve the Internet today.
Do you have any advice for students entering the networking/Internet field?
Think outside the limitations of existing systemsimagine what might be possible; but then
do the hard work of figuring out how to get there from the current state of affairs. Dare to
dream: A half dozen colleagues and I at the Jet Propulsion Laboratory have been working
on the design of an interplanetary extension of the terrestrial Internet. It may take decades to
implement this, mission by mission, but to paraphrase: A mans reach should exceed his
grasp, or what are the heavens for?
432
CHAPTER 5
The Link Layer:
Links, Access
Networks, and
LANs
433
In the previous chapter, we learned that the network layer provides a communica-
tion service between any two network hosts. Between the two hosts, datagrams
travel over a series of communication links, some wired and some wireless, starting
at the source host, passing through a series of packet switches (switches and routers)
and ending at the destination host. As we continue down the protocol stack, from the
network layer to the link layer, we naturally wonder how packets are sent across
the individual links that make up the end-to-end communication path. How are the
network-layer datagrams encapsulated in the link-layer frames for transmission over
a single link? Are different link-layer protocols used in the different links along the
communication path? How are transmission conflicts in broadcast links resolved? Is
there addressing at the link layer and, if so, how does the link-layer addressing oper-
ate with the network-layer addressing we learned about in Chapter 4? And what
exactly is the difference between a switch and a router? Well answer these and
other important questions in this chapter.
In discussing the link layer, well see that there are two fundamentally different
types of link-layer channels. The first type are broadcast channels, which connect mul-
tiple hosts in wireless LANs, satellite networks, and hybrid fiber-coaxial cable (HFC)
434 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
access networks. Since many hosts are connected to the same broadcast communica-
tion channel, a so-called medium access protocol is needed to coordinate frame
transmission. In some cases, a central controller may be used to coordinate transmis-
sions; in other cases, the hosts themselves coordinate transmissions. The second type
of link-layer channel is the point-to-point communication link, such as that often
found between two routers connected by a long-distance link, or between a users
office computer and the nearby Ethernet switch to which it is connected. Coordinating
access to a point-to-point link is simpler; the reference material on this books web site
has a detailed discussion of the Point-to-Point Protocol (PPP), which is used in set-
tings ranging from dial-up service over a telephone line to high-speed point-to-point
frame transport over fiber-optic links.
Well explore several important link-layer concepts and technologies in this chap-
ter. Well dive deeper into error detection and correction, a topic we touched on briefly
in Chapter 3. Well consider multiple access networks and switched LANs, including
Ethernetby far the most prevalent wired LAN technology. Well also look at virtual
LANs, and data center networks. Although WiFi, and more generally wireless LANs,
are link-layer topics, well postpone our study of these important topics until Chapter 6.
5.1 Introduction to the Link Layer
Lets begin with some important terminology. Well find it convenient in this chapter to
refer to any device that runs a link-layer (i.e., layer 2) protocol as a node. Nodes include
hosts, routers, switches, and WiFi access points (discussed in Chapter 6). We will
also refer to the communication channels that connect adjacent nodes along the com-
munication path as links. In order for a datagram to be transferred from source host to
destination host, it must be moved over each of the individual links in the end-to-end
path. As an example, in the company network shown at the bottom of Figure 5.1, con-
sider sending a datagram from one of the wireless hosts to one of the servers. This data-
gram will actually pass through six links: a WiFi link between sending host and WiFi
access point, an Ethernet link between the access point and a link-layer switch; a link
between the link-layer switch and the router, a link between the two routers; an
Ethernet link between the router and a link-layer switch; and finally an Ethernet link
between the switch and the server. Over a given link, a transmitting node encapsulates
the datagram in a link-layer frame and transmits the frame into the link.
In order to gain further insight into the link layer and how it relates to the network
layer, lets consider a transportation analogy. Consider a travel agent who is planning a
trip for a tourist traveling from Princeton, New Jersey, to Lausanne, Switzerland. The
travel agent decides that it is most convenient for the tourist to take a limousine from
Princeton to JFK airport, then a plane from JFK airport to Genevas airport, and finally
a train from Genevas airport to Lausannes train station. Once the travel agent makes
the three reservations, it is the responsibility of the Princeton limousine company to get
the tourist from Princeton to JFK; it is the responsibility of the airline company to
5.1 INTRODUCTION TO THE LINK LAYER 435
Figure 5.1 Six link-layer hops between wireless host and server
Mobile Network
National or
Global ISP
Local or
Regional ISP
Enterprise Network
Home Network
get the tourist from JFK to Geneva; and it is the responsibility of the Swiss train service
to get the tourist from Geneva to Lausanne. Each of the three segments of the trip
is direct between two adjacent locations. Note that the three transportation seg-
ments are managed by different companies and use entirely different transportation
modes (limousine, plane, and train). Although the transportation modes are different,
they each provide the basic service of moving passengers from one location to an
adjacent location. In this transportation analogy, the tourist is a datagram, each trans-
portation segment is a link, the transportation mode is a link-layer protocol, and the
travel agent is a routing protocol.
5.1.1 The Services Provided by the Link Layer
Although the basic service of any link layer is to move a datagram from one node to
an adjacent node over a single communication link, the details of the provided serv-
ice can vary from one link-layer protocol to the next. Possible services that can be
offered by a link-layer protocol include:
Framing. Almost all link-layer protocols encapsulate each network-layer data-
gram within a link-layer frame before transmission over the link. A frame con-
sists of a data field, in which the network-layer datagram is inserted, and a
number of header fields. The structure of the frame is specified by the link-layer
protocol. Well see several different frame formats when we examine specific
link-layer protocols in the second half of this chapter.
Link access. A medium access control (MAC) protocol specifies the rules by which
a frame is transmitted onto the link. For point-to-point links that have a single
sender at one end of the link and a single receiver at the other end of the link, the
MAC protocol is simple (or nonexistent)the sender can send a frame whenever
the link is idle. The more interesting case is when multiple nodes share a single
broadcast linkthe so-called multiple access problem. Here, the MAC protocol
serves to coordinate the frame transmissions of the many nodes.
Reliable delivery. When a link-layer protocol provides reliable delivery service, it
guarantees to move each network-layer datagram across the link without error.
Recall that certain transport-layer protocols (such as TCP) also provide a reliable
delivery service. Similar to a transport-layer reliable delivery service, a link-layer
reliable delivery service can be achieved with acknowledgments and retransmis-
sions (see Section 3.4). A link-layer reliable delivery service is often used for links
that are prone to high error rates, such as a wireless link, with the goal of correcting
an error locallyon the link where the error occursrather than forcing an end-to-
end retransmission of the data by a transport- or application-layer protocol. How-
ever, link-layer reliable delivery can be considered an unnecessary overhead for low
bit-error links, including fiber, coax, and many twisted-pair copper links. For this
reason, many wired link-layer protocols do not provide a reliable delivery service.
436 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Error detection and correction. The link-layer hardware in a receiving node can
incorrectly decide that a bit in a frame is zero when it was transmitted as a one,
and vice versa. Such bit errors are introduced by signal attenuation and electro-
magnetic noise. Because there is no need to forward a datagram that has an error,
many link-layer protocols provide a mechanism to detect such bit errors. This is
done by having the transmitting node include error-detection bits in the frame,
and having the receiving node perform an error check. Recall from Chapters 3
and 4 that the Internets transport layer and network layer also provide a limited
form of error detectionthe Internet checksum. Error detection in the link layer
is usually more sophisticated and is implemented in hardware. Error correction
is similar to error detection, except that a receiver not only detects when bit
errors have occurred in the frame but also determines exactly where in the frame
the errors have occurred (and then corrects these errors).
5.1.2 Where Is the Link Layer Implemented?
Before diving into our detailed study of the link layer, lets conclude this introduc-
tion by considering the question of where the link layer is implemented. Well focus
here on an end system, since we learned in Chapter 4 that the link layer is imple-
mented in a routers line card. Is a hosts link layer implemented in hardware or soft-
ware? Is it implemented on a separate card or chip, and how does it interface with
the rest of a hosts hardware and operating system components?
Figure 5.2 shows a typical host architecture. For the most part, the link layer is
implemented in a network adapter, also sometimes known as a network interface
card (NIC). At the heart of the network adapter is the link-layer controller, usually
a single, special-purpose chip that implements many of the link-layer services
(framing, link access, error detection, and so on). Thus, much of a link-layer con-
trollers functionality is implemented in hardware. For example, Intels 8254x con-
troller [Intel 2012] implements the Ethernet protocols well study in Section 5.5; the
Atheros AR5006 [Atheros 2012] controller implements the 802.11 WiFi protocols
well study in Chapter 6. Until the late 1990s, most network adapters were physi-
cally separate cards (such as a PCMCIA card or a plug-in card fitting into a PCs
PCI card slot) but increasingly, network adapters are being integrated onto the hosts
motherboarda so-called LAN-on-motherboard configuration.
On the sending side, the controller takes a datagram that has been created and
stored in host memory by the higher layers of the protocol stack, encapsulates the
datagram in a link-layer frame (filling in the frames various fields), and then
transmits the frame into the communication link, following the link-access proto-
col. On the receiving side, a controller receives the entire frame, and extracts the
network-layer datagram. If the link layer performs error detection, then it is
the sending controller that sets the error-detection bits in the frame header and it
is the receiving controller that performs error detection.
5.1 INTRODUCTION TO THE LINK LAYER 437
Figure 5.2 shows a network adapter attaching to a hosts bus (e.g., a PCI or
PCI-X bus), where it looks much like any other I/O device to the other host com-
ponents. Figure 5.2 also shows that while most of the link layer is implemented in
hardware, part of the link layer is implemented in software that runs on the hosts
CPU. The software components of the link layer implement higher-level link-
layer functionality such as assembling link-layer addressing information and acti-
vating the controller hardware. On the receiving side, link-layer software responds
to controller interrupts (e.g., due to the receipt of one or more frames), handling
error conditions and passing a datagram up to the network layer. Thus, the link
layer is a combination of hardware and softwarethe place in the protocol stack
where software meets hardware. Intel [2012] provides a readable overview (as
well as a detailed description) of the 8254x controller from a software-program-
ming point of view.
5.2 Error-Detection and -Correction Techniques
In the previous section, we noted that bit-level error detection and correction
detecting and correcting the corruption of bits in a link-layer frame sent from one
node to another physically connected neighboring nodeare two services often
438 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Host
Memory
Host bus
(e.g., PCI)
CPU
Controller
Physical
transmission
Network adapter
Link
Physical
Transport
Network
Link
Application
Figure 5.2 Network adapter: its relationship to other host components
and to protocol stack functionality
provided by the link layer. We saw in Chapter 3 that error-detection and -correction
services are also often offered at the transport layer as well. In this section, well
examine a few of the simplest techniques that can be used to detect and, in some
cases, correct such bit errors. A full treatment of the theory and implementation of
this topic is itself the topic of many textbooks (for example, [Schwartz 1980] or
[Bertsekas 1991]), and our treatment here is necessarily brief. Our goal here is to
develop an intuitive feel for the capabilities that error-detection and -correction
techniques provide, and to see how a few simple techniques work and are used in
practice in the link layer.
Figure 5.3 illustrates the setting for our study. At the sending node, data, D, to
be protected against bit errors is augmented with error-detection and -correction bits
(EDC). Typically, the data to be protected includes not only the datagram passed
down from the network layer for transmission across the link, but also link-level
addressing information, sequence numbers, and other fields in the link frame header.
Both D and EDC are sent to the receiving node in a link-level frame. At the receiv-
ing node, a sequence of bits, D and EDC is received. Note that D and EDC may
differ from the original D and EDC as a result of in-transit bit flips.
The receivers challenge is to determine whether or not D is the same as the
original D, given that it has only received D and EDC. The exact wording of the
receivers decision in Figure 5.3 (we ask whether an error is detected, not whether
an error has occurred!) is important. Error-detection and -correction techniques
EDCD
Detected error
Datagram
EDCD
d data bits
Bit error-prone link
all
bits in D
OK
?
N
Y
Datagram
HI
Figure 5.3 Error-detection and -correction scenario
5.2 ERROR-DETECTION AND -CORRECTION TECHNIQUES 439
allow the receiver to sometimes, but not always, detect that bit errors have
occurred. Even with the use of error-detection bits there still may be undetected
bit errors; that is, the receiver may be unaware that the received information con-
tains bit errors. As a consequence, the receiver might deliver a corrupted datagram
to the network layer, or be unaware that the contents of a field in the frames
header has been corrupted. We thus want to choose an error-detection scheme that
keeps the probability of such occurrences small. Generally, more sophisticated
error-detection and-correction techniques (that is, those that have a smaller proba-
bility of allowing undetected bit errors) incur a larger overheadmore computa-
tion is needed to compute and transmit a larger number of error-detection and
-correction bits.
Lets now examine three techniques for detecting errors in the transmitted data
parity checks (to illustrate the basic ideas behind error detection and correction),
checksumming methods (which are more typically used in the transport layer), and
cyclic redundancy checks (which are more typically used in the link layer in an
adapter).
5.2.1 Parity Checks
Perhaps the simplest form of error detection is the use of a single parity bit. Sup-
pose that the information to be sent, D in Figure 5.4, has d bits. In an even parity
scheme, the sender simply includes one additional bit and chooses its value such
that the total number of 1s in the d + 1 bits (the original information plus a parity
bit) is even. For odd parity schemes, the parity bit value is chosen such that there is
an odd number of 1s. Figure 5.4 illustrates an even parity scheme, with the single
parity bit being stored in a separate field.
Receiver operation is also simple with a single parity bit. The receiver need
only count the number of 1s in the received d + 1 bits. If an odd number of 1-
valued bits are found with an even parity scheme, the receiver knows that at least
one bit error has occurred. More precisely, it knows that some odd number of bit
errors have occurred.
But what happens if an even number of bit errors occur? You should convince
yourself that this would result in an undetected error. If the probability of bit
errors is small and errors can be assumed to occur independently from one bit to
the next, the probability of multiple bit errors in a packet would be extremely small.
440 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
0 1 1 1 0 0 0 1 1 0 1 0 1 0 1 1 1
d data bits
Parity
bit
Figure 5.4 One-bit even parity
In this case, a single parity bit might suffice. However, measurements have shown
that, rather than occurring independently, errors are often clustered together in
bursts. Under burst error conditions, the probability of undetected errors in a
frame protected by single-bit parity can approach 50 percent [Spragins 1991].
Clearly, a more robust error-detection scheme is needed (and, fortunately, is used
in practice!). But before examining error-detection schemes that are used in prac-
tice, lets consider a simple generalization of one-bit parity that will provide us
with insight into error-correction techniques.
Figure 5.5 shows a two-dimensional generalization of the single-bit parity
scheme. Here, the d bits in D are divided into i rows and j columns. A parity value is
computed for each row and for each column. The resulting i + j + 1 parity bits com-
prise the link-layer frames error-detection bits.
Suppose now that a single bit error occurs in the original d bits of informa-
tion. With this two-dimensional parity scheme, the parity of both the column
and the row containing the flipped bit will be in error. The receiver can thus not
only detect the fact that a single bit error has occurred, but can use the column
and row indices of the column and row with parity errors to actually identify the
bit that was corrupted and correct that error! Figure 5.5 shows an example in
101011
111100
011101
001010
101011
101100
011101
001010
Row parity
Parity
error
Parity
error
No errors Correctable
single-bit error
d1,1
d2,1
. . .
di,1
di+1,1
. . .
. . .
. . .
. . .
. . .
d1, j
d2, j
. . .
di, j
di+1, j
d1, j+1
d2, j+1
. . .
di, j+1
di+1, j+1
C
o
lu
m
n
p
a
ri
ty
Figure 5.5 Two-dimensional even parity
5.2 ERROR-DETECTION AND -CORRECTION TECHNIQUES 441
which the 1-valued bit in position (2,2) is corrupted and switched to a 0an
error that is both detectable and correctable at the receiver. Although our discus-
sion has focused on the original d bits of information, a single error in the parity
bits themselves is also detectable and correctable. Two-dimensional parity can
also detect (but not correct!) any combination of two errors in a packet. Other
properties of the two-dimensional parity scheme are explored in the problems at
the end of the chapter.
The ability of the receiver to both detect and correct errors is known as forward
error correction (FEC). These techniques are commonly used in audio storage and
playback devices such as audio CDs. In a network setting, FEC techniques can be
used by themselves, or in conjunction with link-layer ARQ techniques similar to
those we examined in Chapter 3. FEC techniques are valuable because they can
decrease the number of sender retransmissions required. Perhaps more important,
they allow for immediate correction of errors at the receiver. This avoids having to
wait for the round-trip propagation delay needed for the sender to receive a NAK
packet and for the retransmitted packet to propagate back to the receivera poten-
tially important advantage for real-time network applications [Rubenstein 1998] or
links (such as deep-space links) with long propagation delays. Research examining
the use of FEC in error-control protocols includes [Biersack 1992; Nonnenmacher
1998; Byers 1998; Shacham 1990].
5.2.2 Checksumming Methods
In checksumming techniques, the d bits of data in Figure 5.4 are treated as a
sequence of k-bit integers. One simple checksumming method is to simply sum
these k-bit integers and use the resulting sum as the error-detection bits. The
Internet checksum is based on this approachbytes of data are treated as 16-bit
integers and summed. The 1s complement of this sum then forms the Internet
checksum that is carried in the segment header. As discussed in Section 3.3, the
receiver checks the checksum by taking the 1s complement of the sum of the
received data (including the checksum) and checking whether the result is all
1 bits. If any of the bits are 0, an error is indicated. RFC 1071 discusses the Internet
checksum algorithm and its implementation in detail. In the TCP and UDP protocols,
the Internet checksum is computed over all fields (header and data fields
included). In IP the checksum is computed over the IP header (since the UDP or
TCP segment has its own checksum). In other protocols, for example, XTP
[Strayer 1992], one checksum is computed over the header and another checksum
is computed over the entire packet.
Checksumming methods require relatively little packet overhead. For example,
the checksums in TCP and UDP use only 16 bits. However, they provide relatively
weak protection against errors as compared with cyclic redundancy check, which is
discussed below and which is often used in the link layer. A natural question at this
point is, Why is checksumming used at the transport layer and cyclic redundancy
442 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
check used at the link layer? Recall that the transport layer is typically implemented
in software in a host as part of the hosts operating system. Because transport-layer
error detection is implemented in software, it is important to have a simple and fast
error-detection scheme such as checksumming. On the other hand, error detection at
the link layer is implemented in dedicated hardware in adapters, which can rapidly
perform the more complex CRC operations. Feldmeier [Feldmeier 1995] presents
fast software implementation techniques for not only weighted checksum codes, but
CRC (see below) and other codes as well.
5.2.3 Cyclic Redundancy Check (CRC)
An error-detection technique used widely in todays computer networks is based
on cyclic redundancy check (CRC) codes. CRC codes are also known as
polynomial codes, since it is possible to view the bit string to be sent as a polyno-
mial whose coefficients are the 0 and 1 values in the bit string, with operations on
the bit string interpreted as polynomial arithmetic.
CRC codes operate as follows. Consider the d-bit piece of data, D, that the
sending node wants to send to the receiving node. The sender and receiver must first
agree on an r + 1 bit pattern, known as a generator, which we will denote as G. We
will require that the most significant (leftmost) bit of G be a 1. The key idea behind
CRC codes is shown in Figure 5.6. For a given piece of data, D, the sender will
choose r additional bits, R, and append them to D such that the resulting d + r bit
pattern (interpreted as a binary number) is exactly divisible by G (i.e., has no
remainder) using modulo-2 arithmetic. The process of error checking with CRCs is
thus simple: The receiver divides the d + r received bits by G. If the remainder is
nonzero, the receiver knows that an error has occurred; otherwise the data is accepted
as being correct.
All CRC calculations are done in modulo-2 arithmetic without carries in
addition or borrows in subtraction. This means that addition and subtraction are
identical, and both are equivalent to the bitwise exclusive-or (XOR) of the
operands. Thus, for example,
1011 XOR 0101 = 1110
1001 XOR 1101 = 0100
d bits r bits
D: Data bits to be sent
D 2rXOR R
R: CRC bits Bit pattern
Mathematical formula
Figure 5.6 CRC
5.2 ERROR-DETECTION AND -CORRECTION TECHNIQUES 443
Also, we similarly have
1011 0101 = 1110
1001 1101 = 0100
Multiplication and division are the same as in base-2 arithmetic, except that any
required addition or subtraction is done without carries or borrows. As in regular
binary arithmetic, multiplication by 2k left shifts a bit pattern by k places. Thus,
given D and R, the quantity D 2r XOR R yields the d + r bit pattern shown
in Figure 5.6. Well use this algebraic characterization of the d + r bit pattern from
Figure 5.6 in our discussion below.
Let us now turn to the crucial question of how the sender computes R. Recall
that we want to find R such that there is an n such that
D 2r XOR R nG
That is, we want to choose R such that G divides into D 2r XOR R without remain-
der. If we XOR (that is, add modulo-2, without carry) R to both sides of the above
equation, we get
D 2r nG XOR R
This equation tells us that if we divide D 2r by G, the value of the remainder is pre-
cisely R. In other words, we can calculate R as
Figure 5.7 illustrates this calculation for the case of D = 101110, d = 6, G = 1001,
and r 3. The 9 bits transmitted in this case are 101110 011. You should check these
calculations for yourself and also check that indeed D 2r = 101011 G XOR R.
International standards have been defined for 8-, 12-, 16-, and 32-bit genera-
tors, G. The CRC-32 32-bit standard, which has been adopted in a number of link-
level IEEE protocols, uses a generator of
GCRC-32 100000100110000010001110110110111
Each of the CRC standards can detect burst errors of fewer than r + 1 bits. (This
means that all consecutive bit errors of r bits or fewer will be detected.) Furthermore,
under appropriate assumptions, a burst of length greater than r + 1 bits is detected with
probability 1 0.5r. Also, each of the CRC standards can detect any odd number of bit
errors. See [Williams 1993] for a discussion of implementing CRC checks. The theory
=
=
R = remainder
D 2r
G
=
=
444 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
behind CRC codes and even more powerful codes is beyond the scope of this text. The
text [Schwartz 1980] provides an excellent introduction to this topic.
5.3 Multiple Access Links and Protocols
In the introduction to this chapter, we noted that there are two types of network links:
point-to-point links and broadcast links. A point-to-point link consists of a single
sender at one end of the link and a single receiver at the other end of the link. Many
link-layer protocols have been designed for point-to-point links; the point-to-point pro-
tocol (PPP) and high-level data link control (HDLC) are two such protocols that well
cover later in this chapter. The second type of link, a broadcast link, can have multiple
sending and receiving nodes all connected to the same, single, shared broadcast chan-
nel. The term broadcast is used here because when any one node transmits a frame, the
channel broadcasts the frame and each of the other nodes receives a copy. Ethernet and
wireless LANs are examples of broadcast link-layer technologies. In this section well
take a step back from specific link-layer protocols and first examine a problem of cen-
tral importance to the link layer: how to coordinate the access of multiple sending and
receiving nodes to a shared broadcast channelthe multiple access problem. Broad-
cast channels are often used in LANs, networks that are geographically concentrated in
a single building (or on a corporate or university campus). Thus, well also look at how
multiple access channels are used in LANs at the end of this section.
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 445
1001 101110000
101011
1001
101
000
1010
1001
110
000
1100
1001
1010
1001
011
G
D
R
Figure 5.7 A sample CRC calculation
We are all familiar with the notion of broadcastingtelevision has been using
it since its invention. But traditional television is a one-way broadcast (that is, one
fixed node transmitting to many receiving nodes), while nodes on a computer net-
work broadcast channel can both send and receive. Perhaps a more apt human anal-
ogy for a broadcast channel is a cocktail party, where many people gather in a large
room (the air providing the broadcast medium) to talk and listen. A second good
analogy is something many readers will be familiar witha classroomwhere
teacher(s) and student(s) similarly share the same, single, broadcast medium. A cen-
tral problem in both scenarios is that of determining who gets to talk (that is, trans-
mit into the channel), and when. As humans, weve evolved an elaborate set of
protocols for sharing the broadcast channel:
Give everyone a chance to speak.
Dont speak until you are spoken to.
Dont monopolize the conversation.
Raise your hand if you have a question.
Dont interrupt when someone is speaking.
Dont fall asleep when someone is talking.
Computer networks similarly have protocolsso-called multiple access
protocolsby which nodes regulate their transmission into the shared broadcast
channel. As shown in Figure 5.8, multiple access protocols are needed in a wide
variety of network settings, including both wired and wireless access networks, and
satellite networks. Although technically each node accesses the broadcast channel
through its adapter, in this section we will refer to the node as the sending and
receiving device. In practice, hundreds or even thousands of nodes can directly
communicate over a broadcast channel.
Because all nodes are capable of transmitting frames, more than two nodes
can transmit frames at the same time. When this happens, all of the nodes receive
multiple frames at the same time; that is, the transmitted frames collide at all of
the receivers. Typically, when there is a collision, none of the receiving nodes can
make any sense of any of the frames that were transmitted; in a sense, the signals
of the colliding frames become inextricably tangled together. Thus, all the frames
involved in the collision are lost, and the broadcast channel is wasted during the
collision interval. Clearly, if many nodes want to transmit frames frequently,
many transmissions will result in collisions, and much of the bandwidth of the
broadcast channel will be wasted.
In order to ensure that the broadcast channel performs useful work when multiple
nodes are active, it is necessary to somehow coordinate the transmissions of the active
nodes. This coordination job is the responsibility of the multiple access protocol. Over
the past 40 years, thousands of papers and hundreds of PhD dissertations have been
written on multiple access protocols; a comprehensive survey of the first 20 years of
446 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
this body of work is [Rom 1990]. Furthermore, active research in multiple access pro-
tocols continues due to the continued emergence of new types of links, particularly
new wireless links.
Over the years, dozens of multiple access protocols have been implemented
in a variety of link-layer technologies. Nevertheless, we can classify just about
any multiple access protocol as belonging to one of three categories: channel
partitioning protocols, random access protocols, and taking-turns protocols.
Well cover these categories of multiple access protocols in the following three
subsections.
Lets conclude this overview by noting that, ideally, a multiple access protocol
for a broadcast channel of rate R bits per second should have the following desirable
characteristics:
1. When only one node has data to send, that node has a throughput of R bps.
2. When M nodes have data to send, each of these nodes has a throughput of
R/M bps. This need not necessarily imply that each of the M nodes always
Shared wire
(for example, cable access network)
Shared wireless
(for example, WiFi)
Satellite Cocktail party
Head
end
Figure 5.8 Various multiple access channels
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 447
has an instantaneous rate of R/M, but rather that each node should have an
average transmission rate of R/M over some suitably defined interval
of time.
3. The protocol is decentralized; that is, there is no master node that represents a
single point of failure for the network.
4. The protocol is simple, so that it is inexpensive to implement.
5.3.1 Channel Partitioning Protocols
Recall from our early discussion back in Section 1.3 that time-division multiplex-
ing (TDM) and frequency-division multiplexing (FDM) are two techniques that
can be used to partition a broadcast channels bandwidth among all nodes sharing
that channel. As an example, suppose the channel supports N nodes and that the
transmission rate of the channel is R bps. TDM divides time into time frames and
further divides each time frame into N time slots. (The TDM time frame should
not be confused with the link-layer unit of data exchanged between sending and
receiving adapters, which is also called a frame. In order to reduce confusion, in
this subsection well refer to the link-layer unit of data exchanged as a packet.)
Each time slot is then assigned to one of the N nodes. Whenever a node has a
packet to send, it transmits the packets bits during its assigned time slot in the
revolving TDM frame. Typically, slot sizes are chosen so that a single packet can
be transmitted during a slot time. Figure 5.9 shows a simple four-node TDM
example. Returning to our cocktail party analogy, a TDM-regulated cocktail party
would allow one partygoer to speak for a fixed period of time, then allow another
partygoer to speak for the same amount of time, and so on. Once everyone had had
a chance to talk, the pattern would repeat.
TDM is appealing because it eliminates collisions and is perfectly fair: Each
node gets a dedicated transmission rate of R/N bps during each frame time. How-
ever, it has two major drawbacks. First, a node is limited to an average rate of
R/N bps even when it is the only node with packets to send. A second drawback
is that a node must always wait for its turn in the transmission sequenceagain,
even when it is the only node with a frame to send. Imagine the partygoer who is
the only one with anything to say (and imagine that this is the even rarer circum-
stance where everyone wants to hear what that one person has to say). Clearly,
TDM would be a poor choice for a multiple access protocol for this particular
party.
While TDM shares the broadcast channel in time, FDM divides the R bps chan-
nel into different frequencies (each with a bandwidth of R/N) and assigns each fre-
quency to one of the N nodes. FDM thus creates N smaller channels of R/N bps out
of the single, larger R bps channel. FDM shares both the advantages and drawbacks
of TDM. It avoids collisions and divides the bandwidth fairly among the N nodes.
However, FDM also shares a principal disadvantage with TDMa node is limited
to a bandwidth of R/N, even when it is the only node with packets to send.
448 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
A third channel partitioning protocol is code division multiple access
(CDMA). While TDM and FDM assign time slots and frequencies, respectively,
to the nodes, CDMA assigns a different code to each node. Each node then uses
its unique code to encode the data bits it sends. If the codes are chosen carefully,
CDMA networks have the wonderful property that different nodes can transmit
simultaneously and yet have their respective receivers correctly receive a senders
encoded data bits (assuming the receiver knows the senders code) in spite of
interfering transmissions by other nodes. CDMA has been used in military sys-
tems for some time (due to its anti-jamming properties) and now has widespread
civilian use, particularly in cellular telephony. Because CDMAs use is so tightly
tied to wireless channels, well save our discussion of the technical details of
CDMA until Chapter 6. For now, it will suffice to know that CDMA codes, like
time slots in TDM and frequencies in FDM, can be allocated to the multiple
access channel users.
5.3.2 Random Access Protocols
The second broad class of multiple access protocols are random access protocols.
In a random access protocol, a transmitting node always transmits at the full rate
of the channel, namely, R bps. When there is a collision, each node involved in
the collision repeatedly retransmits its frame (that is, packet) until its frame gets
4KHz
FDM
TDM
Link
4KHz
Slot
All slots labeled 2 are dedicated
to a specific sender-receiver pair.
Frame
1
2
2 3 4 1 2 3 4 1 2 3 4 1 2 3 4
Key:
Figure 5.9 A four-node TDM and FDM example
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 449
through without a collision. But when a node experiences a collision, it doesnt
necessarily retransmit the frame right away. Instead it waits a random delay
before retransmitting the frame. Each node involved in a collision chooses inde-
pendent random delays. Because the random delays are independently chosen, it
is possible that one of the nodes will pick a delay that is sufficiently less than the
delays of the other colliding nodes and will therefore be able to sneak its frame
into the channel without a collision.
There are dozens if not hundreds of random access protocols described in the
literature [Rom 1990; Bertsekas 1991]. In this section well describe a few of the
most commonly used random access protocolsthe ALOHA protocols [Abramson
1970; Abramson 1985; Abramson 2009] and the carrier sense multiple access
(CSMA) protocols [Kleinrock 1975b]. Ethernet [Metcalfe 1976] is a popular and
widely deployed CSMA protocol.
Slotted ALOHA
Lets begin our study of random access protocols with one of the simplest random
access protocols, the slotted ALOHA protocol. In our description of slotted
ALOHA, we assume the following:
All frames consist of exactly L bits.
Time is divided into slots of size L/R seconds (that is, a slot equals the time to
transmit one frame).
Nodes start to transmit frames only at the beginnings of slots.
The nodes are synchronized so that each node knows when the slots begin.
If two or more frames collide in a slot, then all the nodes detect the collision
event before the slot ends.
Let p be a probability, that is, a number between 0 and 1. The operation of slotted
ALOHA in each node is simple:
When the node has a fresh frame to send, it waits until the beginning of the next
slot and transmits the entire frame in the slot.
If there isnt a collision, the node has successfully transmitted its frame and thus
need not consider retransmitting the frame. (The node can prepare a new frame
for transmission, if it has one.)
If there is a collision, the node detects the collision before the end of the slot. The
node retransmits its frame in each subsequent slot with probability p until the
frame is transmitted without a collision.
By retransmitting with probability p, we mean that the node effectively tosses
a biased coin; the event heads corresponds to retransmit, which occurs with
450 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
probability p. The event tails corresponds to skip the slot and toss the coin again
in the next slot; this occurs with probability (1 p). All nodes involved in the col-
lision toss their coins independently.
Slotted ALOHA would appear to have many advantages. Unlike channel parti-
tioning, slotted ALOHA allows a node to transmit continuously at the full rate, R,
when that node is the only active node. (A node is said to be active if it has frames
to send.) Slotted ALOHA is also highly decentralized, because each node detects
collisions and independently decides when to retransmit. (Slotted ALOHA does,
however, require the slots to be synchronized in the nodes; shortly well discuss an
unslotted version of the ALOHA protocol, as well as CSMA protocols, none of which
require such synchronization.) Slotted ALOHA is also an extremely simple protocol.
Slotted ALOHA works well when there is only one active node, but how effi-
cient is it when there are multiple active nodes? There are two possible efficiency
concerns here. First, as shown in Figure 5.10, when there are multiple active
nodes, a certain fraction of the slots will have collisions and will therefore be
wasted. The second concern is that another fraction of the slots will be empty
because all active nodes refrain from transmitting as a result of the probabilistic
transmission policy. The only unwasted slots will be those in which exactly
one node transmits. A slot in which exactly one node transmits is said to be
a successful slot. The efficiency of a slotted multiple access protocol is defined
to be the long-run fraction of successful slots in the case when there are a large
number of active nodes, each always having a large number of frames to send.
Node 3
Key:
C = Collision slot
E = Empty slot
S = Successful slot
Node 2
Node 1
2 2 2
1 1 1 1
3 3 3
Time
C E C S E C E S S
Figure 5.10 Nodes 1, 2, and 3 collide in the first slot. Node 2 finally
succeeds in the fourth slot, node 1 in the eighth slot, and
node 3 in the ninth slot
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 451
Note that if no form of access control were used, and each node were to immedi-
ately retransmit after each collision, the efficiency would be zero. Slotted ALOHA
clearly increases the efficiency beyond zero, but by how much?
We now proceed to outline the derivation of the maximum efficiency of slot-
ted ALOHA. To keep this derivation simple, lets modify the protocol a little and
assume that each node attempts to transmit a frame in each slot with probability p.
(That is, we assume that each node always has a frame to send and that the node
transmits with probability p for a fresh frame as well as for a frame that has
already suffered a collision.) Suppose there are N nodes. Then the probability that
a given slot is a successful slot is the probability that one of the nodes transmits
and that the remaining N 1 nodes do not transmit. The probability that a given
node transmits is p; the probability that the remaining nodes do not transmit is
(1 p)N1. Therefore the probability a given node has a success is p(1 p)N1.
Because there are N nodes, the probability that any one of the N nodes has a suc-
cess is Np(1 p)N1.
Thus, when there are N active nodes, the efficiency of slotted ALOHA is
Np(1 p)N1. To obtain the maximum efficiency for N active nodes, we have to find
the p* that maximizes this expression. (See the homework problems for a general
outline of this derivation.) And to obtain the maximum efficiency for a large num-
ber of active nodes, we take the limit of Np*(1 p*)N1 as N approaches infinity.
(Again, see the homework problems.) After performing these calculations, well
find that the maximum efficiency of the protocol is given by 1/e 0.37. That is,
when a large number of nodes have many frames to transmit, then (at best) only
37 percent of the slots do useful work. Thus the effective transmission rate of the
channel is not R bps but only 0.37 R bps! A similar analysis also shows that 37 percent
of the slots go empty and 26 percent of slots have collisions. Imagine the poor
network administrator who has purchased a 100-Mbps slotted ALOHA system,
expecting to be able to use the network to transmit data among a large number of
users at an aggregate rate of, say, 80 Mbps! Although the channel is capable of trans-
mitting a given frame at the full channel rate of 100 Mbps, in the long run, the
successful throughput of this channel will be less than 37 Mbps.
Aloha
The slotted ALOHA protocol required that all nodes synchronize their transmis-
sions to start at the beginning of a slot. The first ALOHA protocol [Abramson
1970] was actually an unslotted, fully decentralized protocol. In pure ALOHA,
when a frame first arrives (that is, a network-layer datagram is passed down from
the network layer at the sending node), the node immediately transmits the frame
in its entirety into the broadcast channel. If a transmitted frame experiences a colli-
sion with one or more other transmissions, the node will then immediately (after
completely transmitting its collided frame) retransmit the frame with probability p.
Otherwise, the node waits for a frame transmission time. After this wait, it then
452 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
transmits the frame with probability p, or waits (remaining idle) for another frame
time with probability 1 p.
To determine the maximum efficiency of pure ALOHA, we focus on an indi-
vidual node. Well make the same assumptions as in our slotted ALOHA analysis
and take the frame transmission time to be the unit of time. At any given time, the
probability that a node is transmitting a frame is p. Suppose this frame begins trans-
mission at time t0. As shown in Figure 5.11, in order for this frame to be success-
fully transmitted, no other nodes can begin their transmission in the interval of time
[t0 1, t0]. Such a transmission would overlap with the beginning of the transmis-
sion of node is frame. The probability that all other nodes do not begin a transmis-
sion in this interval is (1 p)N1. Similarly, no other node can begin a transmission
while node i is transmitting, as such a transmission would overlap with the latter
part of node is transmission. The probability that all other nodes do not begin a
transmission in this interval is also (1 p)N1. Thus, the probability that a given
node has a successful transmission is p(1 p)2(N1). By taking limits as in the slotted
ALOHA case, we find that the maximum efficiency of the pure ALOHA protocol is
only 1/(2e)exactly half that of slotted ALOHA. This then is the price to be paid
for a fully decentralized ALOHA protocol.
Carrier Sense Multiple Access (CSMA)
In both slotted and pure ALOHA, a nodes decision to transmit is made independ-
ently of the activity of the other nodes attached to the broadcast channel. In particu-
lar, a node neither pays attention to whether another node happens to be transmitting
when it begins to transmit, nor stops transmitting if another node begins to interfere
with its transmission. In our cocktail party analogy, ALOHA protocols are quite like
Time
Will overlap
with start of
i s frame
t0 1 t0 t0 + 1
Will overlap
with end of
i s frame
Node i frame
Figure 5.11 Interfering transmissions in pure ALOHA
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 453
a boorish partygoer who continues to chatter away regardless of whether other peo-
ple are talking. As humans, we have human protocols that allow us not only to
behave with more civility, but also to decrease the amount of time spent colliding
with each other in conversation and, consequently, to increase the amount of data
we exchange in our conversations. Specifically, there are two important rules for
polite human conversation:
Listen before speaking. If someone else is speaking, wait until they are finished.
In the networking world, this is called carrier sensinga node listens to the
channel before transmitting. If a frame from another node is currently being
transmitted into the channel, a node then waits until it detects no transmissions
for a short amount of time and then begins transmission.
If someone else begins talking at the same time, stop talking. In the networking
world, this is called collision detectiona transmitting node listens to the channel
while it is transmitting. If it detects that another node is transmitting an interfering
454 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
NORM ABRAMSON AND ALOHANET
Norm Abramson, a PhD engineer, had a passion for surfing and an interest in packet
switching. This combination of interests brought him to the University of Hawaii in
1969. Hawaii consists of many mountainous islands, making it difficult to install and
operate land-based networks. When not surfing, Abramson thought about how to
design a network that does packet switching over radio. The network he designed
had one central host and several secondary nodes scattered over the Hawaiian
Islands. The network had two channels, each using a different frequency band. The
downlink channel broadcasted packets from the central host to the secondary hosts;
and the upstream channel sent packets from the secondary hosts to the central host. In
addition to sending informational packets, the central host also sent on the down-
stream channel an acknowledgment for each packet successfully received from the
secondary hosts.
Because the secondary hosts transmitted packets in a decentralized fashion, colli-
sions on the upstream channel inevitably occurred. This observation led Abramson to
devise the pure ALOHA protocol, as described in this chapter. In 1970, with contin-
ued funding from ARPA, Abramson connected his ALOHAnet to the ARPAnet.
Abramsons work is important not only because it was the first example of a radio
packet network, but also because it inspired Bob Metcalfe. A few years later,
Metcalfe modified the ALOHA protocol to create the CSMA/CD protocol and the
Ethernet LAN.
CASE HISTORY
frame, it stops transmitting and waits a random amount of time before repeating
the sense-and-transmit-when-idle cycle.
These two rules are embodied in the family of carrier sense multiple access
(CSMA) and CSMA with collision detection (CSMA/CD) protocols [Kleinrock
1975b; Metcalfe 1976; Lam 1980; Rom 1990]. Many variations on CSMA and
CSMA/CD have been proposed. Here, well consider a few of the most important,
and fundamental, characteristics of CSMA and CSMA/CD.
The first question that you might ask about CSMA is why, if all nodes per-
form carrier sensing, do collisions occur in the first place? After all, a node will
refrain from transmitting whenever it senses that another node is transmitting. The
answer to the question can best be illustrated using space-time diagrams [Molle
1987]. Figure 5.12 shows a space-time diagram of four nodes (A, B, C, D)
attached to a linear broadcast bus. The horizontal axis shows the position of each
node in space; the vertical axis represents time.
A
Time Time
Space
t 0
t 1
B C D
Figure 5.12 Space-time diagram of two CSMA nodes with colliding
transmissions
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 455
At time t0, node B senses the channel is idle, as no other nodes are currently
transmitting. Node B thus begins transmitting, with its bits propagating in both
directions along the broadcast medium. The downward propagation of Bs bits in
Figure 5.12 with increasing time indicates that a nonzero amount of time is needed
for Bs bits actually to propagate (albeit at near the speed of light) along the
broadcast medium. At time t1 (t1 > t0), node D has a frame to send. Although node
B is currently transmitting at time t1, the bits being transmitted by B have yet to
reach D, and thus D senses the channel idle at t1. In accordance with the CSMA
protocol, D thus begins transmitting its frame. A short time later, Bs transmission
begins to interfere with Ds transmission at D. From Figure 5.12, it is evident that
the end-to-end channel propagation delay of a broadcast channelthe time it
takes for a signal to propagate from one of the nodes to anotherwill play a cru-
cial role in determining its performance. The longer this propagation delay, the
larger the chance that a carrier-sensing node is not yet able to sense a transmission
that has already begun at another node in the network.
Carrier Sense Multiple Access with Collision Dection (CSMA/CD)
In Figure 5.12, nodes do not perform collision detection; both B and D continue to
transmit their frames in their entirety even though a collision has occurred. When a
node performs collision detection, it ceases transmission as soon as it detects a
collision. Figure 5.13 shows the same scenario as in Figure 5.12, except that the two
nodes each abort their transmission a short time after detecting a collision. Clearly,
adding collision detection to a multiple access protocol will help protocol perform-
ance by not transmitting a useless, damaged (by interference with a frame from
another node) frame in its entirety.
Before analyzing the CSMA/CD protocol, let us now summarize its operation
from the perspective of an adapter (in a node) attached to a broadcast channel:
1. The adapter obtains a datagram from the network layer, prepares a link-layer
frame, and puts the frame adapter buffer.
2. If the adapter senses that the channel is idle (that is, there is no signal energy
entering the adapter from the channel), it starts to transmit the frame. If, on the
other hand, the adapter senses that the channel is busy, it waits until it senses
no signal energy and then starts to transmit the frame.
3. While transmitting, the adapter monitors for the presence of signal energy
coming from other adapters using the broadcast channel.
4. If the adapter transmits the entire frame without detecting signal energy from
other adapters, the adapter is finished with the frame. If, on the other hand, the
adapter detects signal energy from other adapters while transmitting, it aborts
the transmission (that is, it stops transmitting its frame).
5. After aborting, the adapter waits a random amount of time and then returns
to step 2.
456 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
The need to wait a random (rather than fixed) amount of time is hopefully clearif
two nodes transmitted frames at the same time and then both waited the same fixed
amount of time, theyd continue colliding forever. But what is a good interval
of time from which to choose the random backoff time? If the interval is large and
the number of colliding nodes is small, nodes are likely to wait a large amount
of time (with the channel remaining idle) before repeating the sense-and-transmit-
when-idle step. On the other hand, if the interval is small and the number of collid-
ing nodes is large, its likely that the chosen random values will be nearly the same,
and transmitting nodes will again collide. What wed like is an interval that is short
when the number of colliding nodes is small, and long when the number of collid-
ing nodes is large.
The binary exponential backoff algorithm, used in Ethernet as well as in
DOCSIS cable network multiple access protocols [DOCSIS 2011], elegantly solves
this problem. Specifically, when transmitting a frame that has already experienced
A
Time Time
Collision
detect/abort
time
Space
t 0
t 1
B C D
Figure 5.13 CSMA with collision detection
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 457
n collisions, a node chooses the value of K at random from {0, 1, 2, . . . . 2n 1}. Thus,
the more collisions experienced by a frame, the larger the interval from which K is
chosen. For Ethernet, the actual amount of time a node waits is K 512 bit times (i.e.,
K times the amount of time needed to send 512 bits into the Ethernet) and the maxi-
mum value that n can take is capped at 10.
Lets look at an example. Suppose that a node attempts to transmit a frame for the
first time and while transmitting it detects a collision. The node then chooses K 0
with probability 0.5 or chooses K 1 with probability 0.5. If the node chooses K
0, then it immediately begins sensing the channel. If the node chooses K 1, it waits
512 bit times (e.g., 0.01 microseconds for a 100 Mbps Ethernet) before beginning
the sense-and-transmit-when-idle cycle. After a second collision, K is chosen with
equal probability from {0,1,2,3}. After three collisions, K is chosen with equal prob-
ability from {0,1,2,3,4,5,6,7}. After 10 or more collisions, K is chosen with equal
probability from {0,1,2, . . . , 1023}. Thus, the size of the sets from which K is cho-
sen grows exponentially with the number of collisions; for this reason this algorithm
is referred to as binary exponential backoff.
We also note here that each time a node prepares a new frame for transmission,
it runs the CSMA/CD algorithm, not taking into account any collisions that may
have occurred in the recent past. So it is possible that a node with a new frame will
immediately be able to sneak in a successful transmission while several other nodes
are in the exponential backoff state.
CSMA/CD Efficiency
When only one node has a frame to send, the node can transmit at the full channel
rate (e.g., for Ethernet typical rates are 10 Mbps, 100 Mbps, or 1 Gbps). However, if
many nodes have frames to transmit, the effective transmission rate of the channel
can be much less. We define the efficiency of CSMA/CD to be the long-run fraction
of time during which frames are being transmitted on the channel without collisions
when there is a large number of active nodes, with each node having a large number
of frames to send. In order to present a closed-form approximation of the efficiency
of Ethernet, let dprop denote the maximum time it takes signal energy to propagate
between any two adapters. Let dtrans be the time to transmit a maximum-size frame
(approximately 1.2 msecs for a 10 Mbps Ethernet). A derivation of the efficiency of
CSMA/CD is beyond the scope of this book (see [Lam 1980] and [Bertsekas 1991]).
Here we simply state the following approximation:
We see from this formula that as dprop approaches 0, the efficiency approaches 1. This
matches our intuition that if the propagation delay is zero, colliding nodes will abort
Efficiency =
1
1 + 5dprop>dtrans
=
==
=
458 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
immediately without wasting the channel. Also, as dtrans becomes very large, effi-
ciency approaches 1. This is also intuitive because when a frame grabs the channel,
it will hold on to the channel for a very long time; thus, the channel will be doing
productive work most of the time.
5.3.3 Taking-Turns Protocols
Recall that two desirable properties of a multiple access protocol are (1) when only
one node is active, the active node has a throughput of R bps, and (2) when M nodes
are active, then each active node has a throughput of nearly R/M bps. The ALOHA
and CSMA protocols have this first property but not the second. This has motivated
researchers to create another class of protocolsthe taking-turns protocols. As
with random access protocols, there are dozens of taking-turns protocols, and each
one of these protocols has many variations. Well discuss two of the more important
protocols here. The first one is the polling protocol. The polling protocol requires
one of the nodes to be designated as a master node. The master node polls each of
the nodes in a round-robin fashion. In particular, the master node first sends a mes-
sage to node 1, saying that it (node 1) can transmit up to some maximum number of
frames. After node 1 transmits some frames, the master node tells node 2 it (node 2)
can transmit up to the maximum number of frames. (The master node can determine
when a node has finished sending its frames by observing the lack of a signal on the
channel.) The procedure continues in this manner, with the master node polling each
of the nodes in a cyclic manner.
The polling protocol eliminates the collisions and empty slots that plague
random access protocols. This allows polling to achieve a much higher efficiency.
But it also has a few drawbacks. The first drawback is that the protocol introduces a
polling delaythe amount of time required to notify a node that it can transmit. If,
for example, only one node is active, then the node will transmit at a rate less than
R bps, as the master node must poll each of the inactive nodes in turn each time the
active node has sent its maximum number of frames. The second drawback, which
is potentially more serious, is that if the master node fails, the entire channel
becomes inoperative. The 802.15 protocol and the Bluetooth protocol we will study
in Section 6.3 are examples of polling protocols.
The second taking-turns protocol is the token-passing protocol. In this protocol
there is no master node. A small, special-purpose frame known as a token is exchanged
among the nodes in some fixed order. For example, node 1 might always send the token
to node 2, node 2 might always send the token to node 3, and node N might always send
the token to node 1. When a node receives a token, it holds onto the token only if it has
some frames to transmit; otherwise, it immediately forwards the token to the next node.
If a node does have frames to transmit when it receives the token, it sends up to a max-
imum number of frames and then forwards the token to the next node. Token passing is
decentralized and highly efficient. But it has its problems as well. For example, the fail-
ure of one node can crash the entire channel. Or if a node accidentally neglects to
5.3 MULTIPLE ACCESS LINKS AND PROTOCOLS 459
release the token, then some recovery procedure must be invoked to get the token back
in circulation. Over the years many token-passing protocols have been developed,
including the fiber distributed data interface (FDDI) protocol [Jain 1994] and the IEEE
802.5 token ring protocol [IEEE 802.5 2012], and each one had to address these as well
as other sticky issues.
5.3.4 DOCSIS: The Link-Layer Protocol for Cable
Internet Access
In the previous three subsections, weve learned about three broad classes of multi-
ple access protocols: channel partitioning protocols, random access protocols, and
taking turns protocols. A cable access network will make for an excellent case study
here, as well find aspects of each of these three classes of multiple access protocols
with the cable access network!
Recall from Section 1.2.1, that a cable access network typically connects sev-
eral thousand residential cable modems to a cable modem termination system
(CMTS) at the cable network headend. The Data-Over-Cable Service Interface
Specifications (DOCSIS) [DOCSIS 2011] specifies the cable data network architec-
ture and its protocols. DOCSIS uses FDM to divide the downstream (CMTS to
modem) and upstream (modem to CMTS) network segments into multiple fre-
quency channels. Each downstream channel is 6 MHz wide, with a maximum
throughput of approximately 40 Mbps per channel (although this data rate is seldom
seen at a cable modem in practice); each upstream channel has a maximum channel
width of 6.4 MHz, and a maximum upstream throughput of approximately 30 Mbps.
Each upstream and downstream channel is a broadcast channel. Frames transmitted
on the downstream channel by the CMTS are received by all cable modems receiv-
ing that channel; since there is just a single CMTS transmitting into the downstream
channel, however, there is no multiple access problem. The upstream direction,
however, is more interesting and technically challenging, since multiple cable
modems share the same upstream channel (frequency) to the CMTS, and thus colli-
sions can potentially occur.
As illustrated in Figure 5.14, each upstream channel is divided into intervals of
time (TDM-like), each containing a sequence of mini-slots during which cable
modems can transmit to the CMTS. The CMTS explicitly grants permission to indi-
vidual cable modems to transmit during specific mini-slots. The CMTS accom-
plishes this by sending a control message known as a MAP message on a
downstream channel to specify which cable modem (with data to send) can transmit
during which mini-slot for the interval of time specified in the control message.
Since mini-slots are explicitly allocated to cable modems, the CMTS can ensure
there are no colliding transmissions during a mini-slot.
But how does the CMTS know which cable modems have data to send in the
first place? This is accomplished by having cable modems send mini-slot-request
frames to the CMTS during a special set of interval mini-slots that are dedicated
460 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
for this purpose, as shown in Figure 5.14. These mini-slot-request frames are
transmitted in a random access manner and so may collide with each other. A
cable modem can neither sense whether the upstream channel is busy nor detect
collisions. Instead, the cable modem infers that its mini-slot-request frame experi-
enced a collision if it does not receive a response to the requested allocation in the
next downstream control message. When a collision is inferred, a cable modem
uses binary exponential backoff to defer the retransmission of its mini-slot
-request frame to a future time slot. When there is little traffic on the upstream
channel, a cable modem may actually transmit data frames during slots nominally
assigned for mini-slot-request frames (and thus avoid having to wait for a mini-slot
assignment).
A cable access network thus serves as a terrific example of multiple access pro-
tocols in actionFDM, TDM, random access, and centrally allocated time slots all
within one network!
5.4 Switched Local Area Networks
Having covered broadcast networks and multiple access protocols in the previ-
ous section, lets turn our attention next to switched local networks. Figure 5.15
shows a switched local network connecting three departments, two servers and a
router with four switches. Because these switches operate at the link layer, they
switch link-layer frames (rather than network-layer datagrams), dont recognize
5.4 SWITCHED LOCAL AREA NETWORKS 461
Figure 5.14 Upstream and downstream channels between CMTS and
cable modems
Residences with
cable modems
Minislots
containing
minislot
request frames
Assigned minislots
containing cable
modem upstream
data frames
Cable head end
MAP frame for
interval [t1,t2]
CMTS
Downstream channel i
Upstream channel j
t1 t2
network-layer addresses, and dont use routing algorithms like RIP or OSPF to
determine paths through the network of layer-2 switches. Instead of using IP
addresses, we will soon see that they use link-layer addresses to forward link-
layer frames through the network of switches. Well begin our study of switched
LANs by first covering link-layer addressing (Section 5.4.1). We then examine
the celebrated Ethernet protocol (Section 5.5.2). After examining link-layer
addressing and Ethernet, well look at how link-layer switches operate (Section
5.4.3), and then see (Section 5.4.4) how these switches are often used to build
large-scale LANs.
5.4.1 Link-Layer Addressing and ARP
Hosts and routers have link-layer addresses. Now you might find this surprising,
recalling from Chapter 4 that hosts and routers have network-layer addresses as
well. You might be asking, why in the world do we need to have addresses at both
the network and link layers? In addition to describing the syntax and function of the
link-layer addresses, in this section we hope to shed some light on why the two layers
462 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Mail
server
To external
internet
1 Gbps
1
2 3
4
56
1 Gbps
1 Gbps
Electrical Engineering Computer Science
100 Mbps
(fiber)
100 Mbps
(fiber)
100 Mbps
(fiber)
Mixture of 10 Mbps,
100 Mbps, 1 Gbps,
Cat 5 cable
Web
server
Computer Engineering
Figure 5.15 An institutional network connected together by four switches
of addresses are useful and, in fact, indispensable. Well also cover the Address Res-
olution Protocol (ARP), which provides a mechanism to translate IP addresses to
link-layer addresses.
MAC Addresses
In truth, it is not hosts and routers that have link-layer addresses but rather their
adapters (that is, network interfaces) that have link-layer addresses. A host or
router with multiple network interfaces will thus have multiple link-layer
addresses associated with it, just as it would also have multiple IP addresses asso-
ciated with it. Its important to note, however, that link-layer switches do not have
link-layer addresses associated with their interfaces that connect to hosts and
routers. This is because the job of the link-layer switch is to carry datagrams
between hosts and routers; a switch does this job transparently, that is, without the
host or router having to explicitly address the frame to the intervening switch.
This is illustrated in Figure 5.16. A link-layer address is variously called a LAN
address, a physical address, or a MAC address. Because MAC address seems to
be the most popular term, well henceforth refer to link-layer addresses as MAC
addresses. For most LANs (including Ethernet and 802.11 wireless LANs), the
MAC address is 6 bytes long, giving 248 possible MAC addresses. As shown in
Figure 5.16, these 6-byte addresses are typically expressed in hexadecimal nota-
tion, with each byte of the address expressed as a pair of hexadecimal numbers.
Although MAC addresses were designed to be permanent, it is now possible to
88-B2-2F-54-1A-0F5C-66-AB-90-75-B1
1A-23-F9-CD-06-9B
49-BD-D2-C7-56-2A
Figure 5.16 Each interface connected to a LAN has a unique MAC
address
5.4 SWITCHED LOCAL AREA NETWORKS 463
change an adapters MAC address via software. For the rest of this section, however,
well assume that an adapters MAC address is fixed.
One interesting property of MAC addresses is that no two adapters have the
same address. This might seem surprising given that adapters are manufactured in
many countries by many companies. How does a company manufacturing
adapters in Taiwan make sure that it is using different addresses from a company
manufacturing adapters in Belgium? The answer is that the IEEE manages the
MAC address space. In particular, when a company wants to manufacture
adapters, it purchases a chunk of the address space consisting of 224 addresses for
a nominal fee. IEEE allocates the chunk of 224 addresses by fixing the first 24 bits
of a MAC address and letting the company create unique combinations of the last
24 bits for each adapter.
An adapters MAC address has a flat structure (as opposed to a hierarchical
structure) and doesnt change no matter where the adapter goes. A laptop with an
Ethernet interface always has the same MAC address, no matter where the com-
puter goes. A smartphone with an 802.11 interface always has the same MAC
address, no matter where the smartphone goes. Recall that, in contrast, IP addresses
have a hierarchical structure (that is, a network part and a host part), and a hosts
IP addresses needs to be changed when the host moves, i.e, changes the network
to which it is attached. An adapters MAC address is analogous to a persons
social security number, which also has a flat addressing structure and which
doesnt change no matter where the person goes. An IP address is analogous to a
persons postal address, which is hierarchical and which must be changed when-
ever a person moves. Just as a person may find it useful to have both a postal
address and a social security number, it is useful for a host and router interfaces to
have both a network-layer address and a MAC address.
When an adapter wants to send a frame to some destination adapter, the send-
ing adapter inserts the destination adapters MAC address into the frame and then
sends the frame into the LAN. As we will soon see, a switch occassionally broad-
casts an incoming frame onto all of its interfaces. Well see in Chapter 6 that
802.11 also broadcasts frames. Thus, an adapter may receive a frame that isnt
addressed to it. Thus, when an adapter receives a frame, it will check to see
whether the destination MAC address in the frame matches its own MAC address.
If there is a match, the adapter extracts the enclosed datagram and passes the data-
gram up the protocol stack. If there isnt a match, the adapter discards the frame,
without passing the network-layer datagram up. Thus, the destination only will be
interrupted when the frame is received.
However, sometimes a sending adapter does want all the other adapters on the
LAN to receive and process the frame it is about to send. In this case, the sending
adapter inserts a special MAC broadcast address into the destination address field
of the frame. For LANs that use 6-byte addresses (such as Ethernet and 802.11),
the broadcast address is a string of 48 consecutive 1s (that is, FF-FF-FF-FF-FF-
FF in hexadecimal notation).
464 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Address Resolution Protocol (ARP)
Because there are both network-layer addresses (for example, Internet IP addresses)
and link-layer addresses (that is, MAC addresses), there is a need to translate
between them. For the Internet, this is the job of the Address Resolution Protocol
(ARP) [RFC 826].
To understand the need for a protocol such as ARP, consider the network shown
in Figure 5.17. In this simple example, each host and router has a single IP address
and single MAC address. As usual, IP addresses are shown in dotted-decimal notation
and MAC addresses are shown in hexadecimal notation. For the purposes of this
discussion, we will assume in this section that the switch broadcasts all frames; that
is, whenever a switch receives a frame on one interface, it forwards the frame on all
of its other interfaces. In the next section, we will provide a more accurate explana-
tion of how switches operate.
Now suppose that the host with IP address 222.222.222.220 wants to send an IP
datagram to host 222.222.222.222. In this example, both the source and destination
are in the same subnet, in the addressing sense of Section 4.4.2. To send a datagram,
the source must give its adapter not only the IP datagram but also the MAC address
for destination 222.222.222.222. The sending adapter will then construct a link-
layer frame containing the destinations MAC address and send the frame into
the LAN.
KEEPING THE LAYERS INDEPENDENT
There are several reasons why hosts and router interfaces have MAC addresses in addition
to network-layer addresses. First, LANs are designed for arbitrary network-layer protocols,
not just for IP and the Internet. If adapters were assigned IP addresses rather than neutral
MAC addresses, then adapters would not easily be able to support other network-layer
protocols (for example, IPX or DECnet). Second, if adapters were to use network-layer
addresses instead of MAC addresses, the network-layer address would have to be stored
in the adapter RAM and reconfigured every time the adapter was moved (or powered up).
Another option is to not use any addresses in the adapters and have each adapter pass
the data (typically, an IP datagram) of each frame it receives up the protocol stack. The
network layer could then check for a matching network-layer address. One problem with
this option is that the host would be interrupted by every frame sent on the LAN, including
by frames that were destined for other hosts on the same broadcast LAN. In summary, in
order for the layers to be largely independent building blocks in a network architecture,
different layers need to have their own addressing scheme. We have now seen three types
of addresses: host names for the application layer, IP addresses for the network layer, and
MAC addresses for the link layer.
PRINCIPLES IN PRACTICE
5.4 SWITCHED LOCAL AREA NETWORKS 465
The important question addressed in this section is, How does the sending
host determine the MAC address for the destination host with IP address
222.222.222.222? As you might have guessed, it uses ARP. An ARP module in the
sending host takes any IP address on the same LAN as input, and returns the corre-
sponding MAC address. In the example at hand, sending host 222.222.222.220
provides its ARP module the IP address 222.222.222.222, and the ARP module
returns the corresponding MAC address 49-BD-D2-C7-56-2A.
So we see that ARP resolves an IP address to a MAC address. In many ways it
is analogous to DNS (studied in Section 2.5), which resolves host names to IP
addresses. However, one important difference between the two resolvers is that
DNS resolves host names for hosts anywhere in the Internet, whereas ARP resolves
IP addresses only for hosts and router interfaces on the same subnet. If a node in
California were to try to use ARP to resolve the IP address for a node in Mississippi,
ARP would return with an error.
Now that we have explained what ARP does, lets look at how it works. Each
host and router has an ARP table in its memory, which contains mappings of IP
addresses to MAC addresses. Figure 5.18 shows what an ARP table in host
222.222.222.220 might look like. The ARP table also contains a time-to-live (TTL)
value, which indicates when each mapping will be deleted from the table. Note that
a table does not necessarily contain an entry for every host and router on the subnet;
some may have never been entered into the table, and others may have expired.
A typical expiration time for an entry is 20 minutes from when an entry is placed in
an ARP table.
466 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
IP:222.222.222.221
IP:222.222.222.220
IP:222.222.222.223
IP:222.222.222.222
5C-66-AB-90-75-B1
1A-23-F9-CD-06-9B
49-BD-D2-C7-56-2A
88-B2-2F-54-1A-0F
A
B
C
Figure 5.17 Each interface on a LAN has an IP address and a MAC
address
Now suppose that host 222.222.222.220 wants to send a datagram that is IP-
addressed to another host or router on that subnet. The sending host needs to
obtain the MAC address of the destination given the IP address. This task is easy
if the senders ARP table has an entry for the destination node. But what if the
ARP table doesnt currently have an entry for the destination? In particular, sup-
pose222.222.222.220 wants to send a datagram to 222.222.222.222. In this case,
the sender uses the ARP protocol to resolve the address. First, the sender con-
structs a special packet called an ARP packet. An ARP packet has several fields,
including the sending and receiving IP and MAC addresses. Both ARP query and
response packets have the same format. The purpose of the ARP query packet is
to query all the other hosts and routers on the subnet to determine the MAC
address corresponding to the IP address that is being resolved.
Returning to our example, 222.222.222.220 passes an ARP query packet to
the adapter along with an indication that the adapter should send the packet to the
MAC broadcast address, namely, FF-FF-FF-FF-FF-FF. The adapter encapsulates
the ARP packet in a link-layer frame, uses the broadcast address for the frames
destination address, and transmits the frame into the subnet. Recalling our social
security number/postal address analogy, an ARP query is equivalent to a person
shouting out in a crowded room of cubicles in some company (say, AnyCorp):
What is the social security number of the person whose postal address is Cubicle
13, Room 112, AnyCorp, Palo Alto, California? The frame containing the ARP
query is received by all the other adapters on the subnet, and (because of the
broadcast address) each adapter passes the ARP packet within the frame up to its
ARP module. Each of these ARP modules checks to see if its IP address matches
the destination IP address in the ARP packet. The one with a match sends back to
the querying host a response ARP packet with the desired mapping. The querying
host 222.222.222.220 can then update its ARP table and send its IP datagram,
encapsulated in a link-layer frame whose destination MAC is that of the host or
router responding to the earlier ARP query.
There are a couple of interesting things to note about the ARP protocol. First,
the query ARP message is sent within a broadcast frame, whereas the response ARP
message is sent within a standard frame. Before reading on you should think about
why this is so. Second, ARP is plug-and-play; that is, an ARP table gets built
automaticallyit doesnt have to be configured by a system administrator. And if
IP Address MAC Address TTL
222.222.222.221 88-B2-2F-54-1A-0F 13:45:00
222.222.222.223 5C-66-AB-90-75-B1 13:52:00
Figure 5.18 A possible ARP table in 222.222.222.220
5.4 SWITCHED LOCAL AREA NETWORKS 467
a host becomes disconnected from the subnet, its entry is eventually deleted from
the other ARP tables in the subnet.
Students often wonder if ARP is a link-layer protocol or a network-layer pro-
tocol. As weve seen, an ARP packet is encapsulated within a link-layer frame
and thus lies architecturally above the link layer. However, an ARP packet has
fields containing link-layer addresses and thus is arguably a link-layer protocol,
but it also contains network-layer addresses and thus is also arguably a network-
layer protocol. In the end, ARP is probably best considered a protocol that strad-
dles the boundary between the link and network layersnot fitting neatly into
the simple layered protocol stack we studied in Chapter 1. Such are the complex-
ities of real-world protocols!
Sending a Datagram off the Subnet
It should now be clear how ARP operates when a host wants to send a datagram to
another host on the same subnet. But now lets look at the more complicated situa-
tion when a host on a subnet wants to send a network-layer datagram to a host off
the subnet (that is, across a router onto another subnet). Lets discuss this issue in
the context of Figure 5.19, which shows a simple network consisting of two subnets
interconnected by a router.
There are several interesting things to note about Figure 5.19. Each host has
exactly one IP address and one adapter. But, as discussed in Chapter 4, a router has
an IP address for each of its interfaces. For each router interface there is also an ARP
module (in the router) and an adapter. Because the router in Figure 5.19 has two
interfaces, it has two IP addresses, two ARP modules, and two adapters. Of course,
each adapter in the network has its own MAC address.
Also note that Subnet 1 has the network address 111.111.111/24 and that Sub-
net 2 has the network address 222.222.222/24. Thus all of the interfaces connected
to Subnet 1 have addresses of the form 111.111.111.xxx and all of the interfaces
connected to Subnet 2 have addresses of the form 222.222.222.xxx.
468 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
IP:111.111.111.110IP:111.111.111.111
IP:111.111.111.112
IP:222.222.222.221
IP:222.222.222.222
74-29-9C-E8-FF-55
CC-49-DE-D0-AB-7D
E6-E9-00-17-BB-4B
1A-23-F9-CD-06-9B
IP:222.222.222.220
88-B2-2F-54-1A-0F
49-BD-D2-C7-56-2A
Figure 5.19 Two subnets interconnected by a router
Now lets examine how a host on Subnet 1 would send a datagram to a host
on Subnet 2. Specifically, suppose that host 111.111.111.111 wants to send an IP
datagram to a host 222.222.222.222. The sending host passes the datagram to its
adapter, as usual. But the sending host must also indicate to its adapter an appro-
priate destination MAC address. What MAC address should the adapter use? One
might be tempted to guess that the appropriate MAC address is that of the adapter
for host 222.222.222.222, namely, 49-BD-D2-C7-56-2A. This guess, however,
would be wrong! If the sending adapter were to use that MAC address, then none
of the adapters on Subnet 1 would bother to pass the IP datagram up to its net-
work layer, since the frames destination address would not match the MAC
address of any adapter on Subnet 1. The datagram would just die and go to data-
gram heaven.
If we look carefully at Figure 5.19, we see that in order for a datagram to go
from 111.111.111.111 to a host on Subnet 2, the datagram must first be sent to the
router interface 111.111.111.110, which is the IP address of the first-hop router
on the path to the final destination. Thus, the appropriate MAC address for the
frame is the address of the adapter for router interface 111.111.111.110, namely,
E6-E9-00-17-BB-4B. How does the sending host acquire the MAC address for
111.111.111.110? By using ARP, of course! Once the sending adapter has this
MAC address, it creates a frame (containing the datagram addressed to
222.222.222.222) and sends the frame into Subnet 1. The router adapter on Sub-
net 1 sees that the link-layer frame is addressed to it, and therefore passes the
frame to the network layer of the router. Hooraythe IP datagram has success-
fully been moved from source host to the router! But we are not finished. We still
have to move the datagram from the router to the destination. The router now has
to determine the correct interface on which the datagram is to be forwarded. As
discussed in Chapter 4, this is done by consulting a forwarding table in the router.
The forwarding table tells the router that the datagram is to be forwarded via
router interface 222.222.222.220. This interface then passes the datagram to its
adapter, which encapsulates the datagram in a new frame and sends the frame
into Subnet 2. This time, the destination MAC address of the frame is indeed the
MAC address of the ultimate destination. And how does the router obtain this
destination MAC address? From ARP, of course!
ARP for Ethernet is defined in RFC 826. A nice introduction to ARP is given in
the TCP/IP tutorial, RFC 1180. Well explore ARP in more detail in the homework
problems.
5.4.2 Ethernet
Ethernet has pretty much taken over the wired LAN market. In the 1980s and the
early 1990s, Ethernet faced many challenges from other LAN technologies, includ-
ing token ring, FDDI, and ATM. Some of these other technologies succeeded in
capturing a part of the LAN market for a few years. But since its invention in the
5.4 SWITCHED LOCAL AREA NETWORKS 469
VideoNote
Sending a datagram
between subnets:
link-layer and
network-layer
addressing
mid-1970s, Ethernet has continued to evolve and grow and has held on to its
dominant position. Today, Ethernet is by far the most prevalent wired LAN tech-
nology, and it is likely to remain so for the foreseeable future. One might say that
Ethernet has been to local area networking what the Internet has been to global
networking.
There are many reasons for Ethernets success. First, Ethernet was the first
widely deployed high-speed LAN. Because it was deployed early, network admin-
istrators became intimately familiar with Ethernetits wonders and its quirks
and were reluctant to switch over to other LAN technologies when they came on
the scene. Second, token ring, FDDI, and ATM were more complex and expensive
than Ethernet, which further discouraged network administrators from switching
over. Third, the most compelling reason to switch to another LAN technology
(such as FDDI or ATM) was usually the higher data rate of the new technology;
however, Ethernet always fought back, producing versions that operated at equal
data rates or higher. Switched Ethernet was also introduced in the early 1990s,
which further increased its effective data rates. Finally, because Ethernet has been
so popular, Ethernet hardware (in particular, adapters and switches) has become a
commodity and is remarkably cheap.
The original Ethernet LAN was invented in the mid-1970s by Bob Metcalfe and
David Boggs. The original Ethernet LAN used a coaxial bus to interconnect the
nodes. Bus topologies for Ethernet actually persisted throughout the 1980s and into
the mid-1990s. Ethernet with a bus topology is a broadcast LANall transmitted
frames travel to and are processed by all adapters connected to the bus. Recall that
we covered Ethernets CSMA/CD multiple access protocol with binary exponential
backoff in Section 5.3.2.
By the late 1990s, most companies and universities had replaced their LANs
with Ethernet installations using a hub-based star topology. In such an installation
the hosts (and routers) are directly connected to a hub with twisted-pair copper
wire. A hub is a physical-layer device that acts on individual bits rather than
frames. When a bit, representing a zero or a one, arrives from one interface, the
hub simply re-creates the bit, boosts its energy strength, and transmits the bit onto
all the other interfaces. Thus, Ethernet with a hub-based star topology is also a
broadcast LANwhenever a hub receives a bit from one of its interfaces, it sends
a copy out on all of its other interfaces. In particular, if a hub receives frames from
two different interfaces at the same time, a collision occurs and the nodes that cre-
ated the frames must retransmit.
In the early 2000s Ethernet experienced yet another major evolutionary change.
Ethernet installations continued to use a star topology, but the hub at the center was
replaced with a switch. Well be examining switched Ethernet in depth later in this
chapter. For now, we only mention that a switch is not only collision-less but
is also a bona-fide store-and-forward packet switch; but unlike routers, which operate
up through layer 3, a switch operates only up through layer 2.
470 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Ethernet Frame Structure
We can learn a lot about Ethernet by examining the Ethernet frame, which is shown in
Figure 5.20. To give this discussion about Ethernet frames a tangible context, lets con-
sider sending an IP datagram from one host to another host, with both hosts on the same
Ethernet LAN (for example, the Ethernet LAN in Figure 5.17.) (Although the payload
of our Ethernet frame is an IP datagram, we note that an Ethernet frame can carry
other network-layer packets as well.) Let the sending adapter, adapter A, have the
MAC address AA-AA-AA-AA-AA-AA and the receiving adapter, adapter B, have the
MAC address BB-BB-BB-BB-BB-BB. The sending adapter encapsulates the IP data-
gram within an Ethernet frame and passes the frame to the physical layer. The receiv-
ing adapter receives the frame from the physical layer, extracts the IP datagram, and
passes the IP datagram to the network layer. In this context, lets now examine the six
fields of the Ethernet frame, as shown in Figure 5.20.
Data field (46 to 1,500 bytes). This field carries the IP datagram. The maximum
transmission unit (MTU) of Ethernet is 1,500 bytes. This means that if the IP
datagram exceeds 1,500 bytes, then the host has to fragment the datagram, as dis-
cussed in Section 4.4.1. The minimum size of the data field is 46 bytes. This
means that if the IP datagram is less than 46 bytes, the data field has to be
stuffed to fill it out to 46 bytes. When stuffing is used, the data passed to the
network layer contains the stuffing as well as an IP datagram. The network layer
uses the length field in the IP datagram header to remove the stuffing.
Destination address (6 bytes). This field contains the MAC address of the des-
tination adapter, BB-BB-BB-BB-BB-BB. When adapter B receives an Ether-
net frame whose destination address is either BB-BB-BB-BB-BB-BB or the
MAC broadcast address, it passes the contents of the frames data field to the
network layer; if it receives a frame with any other MAC address, it discards
the frame.
Source address (6 bytes). This field contains the MAC address of the adapter that
transmits the frame onto the LAN, in this example, AA-AA-AA-AA-AA-AA.
Type field (2 bytes). The type field permits Ethernet to multiplex network-layer
protocols. To understand this, we need to keep in mind that hosts can use other
network-layer protocols besides IP. In fact, a given host may support multiple
Preamble CRC
Dest.
address
Source
address
Type
Data
Figure 5.20 Ethernet frame structure
5.4 SWITCHED LOCAL AREA NETWORKS 471
network-layer protocols using different protocols for different applications. For
this reason, when the Ethernet frame arrives at adapter B, adapter B needs to
know to which network-layer protocol it should pass (that is, demultiplex) the
contents of the data field. IP and other network-layer protocols (for example,
Novell IPX or AppleTalk) each have their own, standardized type number. Fur-
thermore, the ARP protocol (discussed in the previous section) has its own type
number, and if the arriving frame contains an ARP packet (i.e., has a type field
of 0806 hexadecimal), the ARP packet will be demultiplexed up to the ARP pro-
tocol. Note that the type field is analogous to the protocol field in the network-
layer datagram and the port-number fields in the transport-layer segment; all of
these fields serve to glue a protocol at one layer to a protocol at the layer above.
Cyclic redundancy check (CRC) (4 bytes). As discussed in Section 5.2.3, the pur-
pose of the CRC field is to allow the receiving adapter, adapter B, to detect bit
errors in the frame.
Preamble (8 bytes). The Ethernet frame begins with an 8-byte preamble field.
Each of the first 7 bytes of the preamble has a value of 10101010; the last byte is
10101011. The first 7 bytes of the preamble serve to wake up the receiving
adapters and to synchronize their clocks to that of the senders clock. Why
should the clocks be out of synchronization? Keep in mind that adapter A aims
to transmit the frame at 10 Mbps, 100 Mbps, or 1 Gbps, depending on the type
of Ethernet LAN. However, because nothing is absolutely perfect, adapter A will
not transmit the frame at exactly the target rate; there will always be some drift
from the target rate, a drift which is not known a priori by the other adapters on
the LAN. A receiving adapter can lock onto adapter As clock simply by locking
onto the bits in the first 7 bytes of the preamble. The last 2 bits of the eighth byte
of the preamble (the first two consecutive 1s) alert adapter B that the important
stuff is about to come.
All of the Ethernet technologies provide connectionless service to the network
layer. That is, when adapter A wants to send a datagram to adapter B, adapter A encap-
sulates the datagram in an Ethernet frame and sends the frame into the LAN, without
first handshaking with adapter B. This layer-2 connectionless service is analogous to
IPs layer-3 datagram service and UDPs layer-4 connectionless service.
Ethernet technologies provide an unreliable service to the network layer.
Specifically, when adapter B receives a frame from adapter A, it runs the frame
through a CRC check, but neither sends an acknowledgment when a frame passes
the CRC check nor sends a negative acknowledgment when a frame fails the CRC
check. When a frame fails the CRC check, adapter B simply discards the frame.
Thus, adapter A has no idea whether its transmitted frame reached adapter B and
passed the CRC check. This lack of reliable transport (at the link layer) helps to
make Ethernet simple and cheap. But it also means that the stream of datagrams
passed to the network layer can have gaps.
472 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
If there are gaps due to discarded Ethernet frames, does the application at Host
B see gaps as well? As we learned in Chapter 3, this depends on whether the appli-
cation is using UDP or TCP. If the application is using UDP, then the application in
Host B will indeed see gaps in the data. On the other hand, if the application is using
TCP, then TCP in Host B will not acknowledge the data contained in discarded
frames, causing TCP in Host A to retransmit. Note that when TCP retransmits data,
the data will eventually return to the Ethernet adapter at which it was discarded.
Thus, in this sense, Ethernet does retransmit data, although Ethernet is unaware of
whether it is transmitting a brand-new datagram with brand-new data, or a datagram
that contains data that has already been transmitted at least once.
Ethernet Technologies
In our discussion above, weve referred to Ethernet as if it were a single protocol stan-
dard. But in fact, Ethernet comes in many different flavors, with somewhat bewilder-
ing acronyms such as 10BASE-T, 10BASE-2, 100BASE-T, 1000BASE-LX, and
BOB METCALFE AND ETHERNET
As a PhD student at Harvard University in the early 1970s, Bob Metcalfe worked on
the ARPAnet at MIT. During his studies, he also became exposed to Abramsons work
on ALOHA and random access protocols. After completing his PhD and just before
beginning a job at Xerox Palo Alto Research Center (Xerox PARC), he visited
Abramson and his University of Hawaii colleagues for three months, getting a first-
hand look at ALOHAnet. At Xerox PARC, Metcalfe became exposed to Alto comput-
ers, which in many ways were the forerunners of the personal computers of the
1980s. Metcalfe saw the need to network these computers in an inexpensive manner.
So armed with his knowledge about ARPAnet, ALOHAnet, and random access proto-
cols, Metcalfealong with colleague David Boggsinvented Ethernet.
Metcalfe and Boggss original Ethernet ran at 2.94 Mbps and linked up to 256
hosts separated by up to one mile. Metcalfe and Boggs succeeded at getting most of
the researchers at Xerox PARC to communicate through their Alto computers.
Metcalfe then forged an alliance between Xerox, Digital, and Intel to establish
Ethernet as a 10 Mbps Ethernet standard, ratified by the IEEE. Xerox did not show
much interest in commercializing Ethernet. In 1979, Metcalfe formed his own com-
pany, 3Com, which developed and commercialized networking technology, including
Ethernet technology. In particular, 3Com developed and marketed Ethernet cards in
the early 1980s for the immensely popular IBM PCs. Metcalfe left 3Com in 1990,
when it had 2,000 employees and $400 million in revenue.
CASE HISTORY
5.4 SWITCHED LOCAL AREA NETWORKS 473
10GBASE-T. These and many other Ethernet technologies have been standardized
over the years by the IEEE 802.3 CSMA/CD (Ethernet) working group [IEEE 802.3
2012]. While these acronyms may appear bewildering, there is actually considerable
order here. The first part of the acronym refers to the speed of the standard: 10, 100,
1000, or 10G, for 10 Megabit (per second), 100 Megabit, Gigabit, and 10 Gigabit
Ethernet, respectively. BASE refers to baseband Ethernet, meaning that the physical
media only carries Ethernet traffic; almost all of the 802.3 standards are for baseband
Ethernet. The final part of the acronym refers to the physical media itself; Ethernet is
both a link-layer and a physical-layer specification and is carried over a variety of
physical media including coaxial cable, copper wire, and fiber. Generally, a T refers
to twisted-pair copper wires.
Historically, an Ethernet was initially conceived of as a segment of coaxial
cable. The early 10BASE-2 and 10BASE-5 standards specify 10 Mbps Ethernet
over two types of coaxial cable, each limited in length to 500 meters. Longer runs
could be obtained by using a repeatera physical-layer device that receives a
signal on the input side, and regenerates the signal on the output side. A coaxial
cable, as in Figure 5.20, corresponds nicely to our view of Ethernet as a broadcast
mediumall frames transmitted by one interface are received at other interfaces,
and Ethernets CDMA/CD protocol nicely solves the multiple access problem.
Nodes simply attach to the cable, and voila, we have a local area network!
Ethernet has passed through a series of evolutionary steps over the years, and
todays Ethernet is very different from the original bus-topology designs using coax-
ial cable. In most installations today, nodes are connected to a switch via point-to-
point segments made of twisted-pair copper wires or fiber-optic cables, as shown in
Figures 5.155.17.
In the mid-1990s, Ethernet was standardized at 100 Mbps, 10 times faster than
10 Mbps Ethernet. The original Ethernet MAC protocol and frame format were pre-
served, but higher-speed physical layers were defined for copper wire (100BASE-T)
and fiber (100BASE-FX, 100BASE-SX, 100BASE-BX). Figure 5.21 shows these
different standards and the common Ethernet MAC protocol and frame format.
474 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Physical
Transport
Network
Link
Application
100BASE-TX
100BASE-T4
100BASE-T2
MAC protocol
and frame format
100BASE-SX
100BASE-FX
100BASE-BX
Figure 5.21 100 Mbps Ethernet standards: a common link layer,
different physical layers
100 Mbps Ethernet is limited to a 100 meter distance over twisted pair, and to sev-
eral kilometers over fiber, allowing Ethernet switches in different buildings to be
connected.
Gigabit Ethernet is an extension to the highly successful 10 Mbps and 100 Mbps
Ethernet standards. Offering a raw data rate of 1,000 Mbps, Gigabit Ethernet main-
tains full compatibility with the huge installed base of Ethernet equipment. The stan-
dard for Gigabit Ethernet, referred to as IEEE 802.3z, does the following:
Uses the standard Ethernet frame format (Figure 5.20) and is backward com-
patible with 10BASE-T and 100BASE-T technologies. This allows for easy
integration of Gigabit Ethernet with the existing installed base of Ethernet
equipment.
Allows for point-to-point links as well as shared broadcast channels. Point-
to-point links use switches while broadcast channels use hubs, as described
earlier. In Gigabit Ethernet jargon, hubs are called buffered distributors.
Uses CSMA/CD for shared broadcast channels. In order to have acceptable effi-
ciency, the maximum distance between nodes must be severely restricted.
Allows for full-duplex operation at 1,000 Mbps in both directions for point-to-
point channels.
Initially operating over optical fiber, Gigabit Ethernet is now able to run over cate-
gory 5 UTP cabling. 10 Gbps Ethernet (10GBASE-T) was standardized in 2007,
providing yet higher Ethernet LAN capacities.
Lets conclude our discussion of Ethernet technology by posing a question that
may have begun troubling you. In the days of bus topologies and hub-based star
topologies, Ethernet was clearly a broadcast link (as defined in Section 5.3) in which
frame collisions occurred when nodes transmitted at the same time. To deal with
these collisions, the Ethernet standard included the CSMA/CD protocol, which is
particularly effective for a wired broadcast LAN spanning a small geographical
region. But if the prevalent use of Ethernet today is a switch-based star topology,
using store-and-forward packet switching, is there really a need anymore for an Eth-
ernet MAC protocol? As well see shortly, a switch coordinates its transmissions and
never forwards more than one frame onto the same interface at any time. Further-
more, modern switches are full-duplex, so that a switch and a node can each send
frames to each other at the same time without interference. In other words, in a
switch-based Ethernet LAN there are no collisions and, therefore, there is no need
for a MAC protocol!
As weve seen, todays Ethernets are very different from the original Ethernet
conceived by Metcalfe and Boggs more than 30 years agospeeds have increased
by three orders of magnitude, Ethernet frames are carried over a variety of media,
switched-Ethernets have become dominant, and now even the MAC protocol is
often unnecessary! Is all of this really still Ethernet? The answer, of course, is yes,
5.4 SWITCHED LOCAL AREA NETWORKS 475
by definition. It is interesting to note, however, that through all of these changes,
there has indeed been one enduring constant that has remained unchanged over
30 yearsEthernets frame format. Perhaps this then is the one true and timeless
centerpiece of the Ethernet standard.
5.4.3 Link-Layer Switches
Up until this point, we have been purposefully vague about what a switch actually
does and how it works. The role of the switch is to receive incoming link-layer frames
and forward them onto outgoing links; well study this forwarding function in detail
in this subsection. Well see that the switch itself is transparent to the hosts and
routers in the subnet; that is, a host/router addresses a frame to another host/router
(rather than addressing the frame to the switch) and happily sends the frame into the
LAN, unaware that a switch will be receiving the frame and forwarding it. The rate at
which frames arrive to any one of the switchs output interfaces may temporarily
exceed the link capacity of that interface. To accommodate this problem, switch out-
put interfaces have buffers, in much the same way that router output interfaces have
buffers for datagrams. Lets now take a closer look at how switches operate.
Forwarding and Filtering
Filtering is the switch function that determines whether a frame should be for-
warded to some interface or should just be dropped. Forwarding is the switch
function that determines the interfaces to which a frame should be directed, and
then moves the frame to those interfaces. Switch filtering and forwarding are
done with a switch table. The switch table contains entries for some, but not nec-
essarily all, of the hosts and routers on a LAN. An entry in the switch table con-
tains (1) a MAC address, (2) the switch interface that leads toward that MAC
address, and (3) the time at which the entry was placed in the table. An example
switch table for the uppermost switch in Figure 5.15 is shown in Figure 5.22.
476 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Figure 5.22 Portion of a switch table for the uppermost switch in
Figure 5.15
Address Interface Time
62-FE-F7-11-89-A3 1 9:32
7C-BA-B2-B4-91-10 3 9:36
. . .
Although this description of frame forwarding may sound similar to our
discussion of datagram forwarding in Chapter 4, well see shortly that there are
important differences. One important difference is that switches forward packets
based on MAC addresses rather than on IP addresses. We will also see that a
switch table is constructed in a very different manner from a routers forwarding
table.
To understand how switch filtering and forwarding work, suppose a frame with
destination address DD-DD-DD-DD-DD-DD arrives at the switch on interface x. The
switch indexes its table with the MAC address DD-DD-DD-DD-DD-DD. There are
three possible cases:
There is no entry in the table for DD-DD-DD-DD-DD-DD. In this case, the switch
forwards copies of the frame to the output buffers preceding all interfaces except
for interface x. In other words, if there is no entry for the destination address, the
switch broadcasts the frame.
There is an entry in the table, associating DD-DD-DD-DD-DD-DD with inter-
face x. In this case, the frame is coming from a LAN segment that contains
adapter DD-DD-DD-DD-DD-DD. There being no need to forward the frame to
any of the other interfaces, the switch performs the filtering function by discard-
ing the frame.
There is an entry in the table, associating DD-DD-DD-DD-DD-DD with inter-
face yx. In this case, the frame needs to be forwarded to the LAN segment
attached to interface y. The switch performs its forwarding function by putting
the frame in an output buffer that precedes interface y.
Lets walk through these rules for the uppermost switch in Figure 5.15 and its
switch table in Figure 5.22. Suppose that a frame with destination address 62-FE-
F7-11-89-A3 arrives at the switch from interface 1. The switch examines its table
and sees that the destination is on the LAN segment connected to interface 1 (that
is, Electrical Engineering). This means that the frame has already been broadcast on
the LAN segment that contains the destination. The switch therefore filters (that is,
discards) the frame. Now suppose a frame with the same destination address arrives
from interface 2. The switch again examines its table and sees that the destination is
in the direction of interface 1; it therefore forwards the frame to the output buffer
preceding interface 1. It should be clear from this example that as long as the switch
table is complete and accurate, the switch forwards frames towards destinations
without any broadcasting.
In this sense, a switch is smarter than a hub. But how does this switch table
get configured in the first place? Are there link-layer equivalents to network-
layer routing protocols? Or must an overworked manager manually configure the
switch table?
5.4 SWITCHED LOCAL AREA NETWORKS 477
Self-Learning
A switch has the wonderful property (particularly for the already-overworked net-
work administrator) that its table is built automatically, dynamically, and
autonomouslywithout any intervention from a network administrator or from a
configuration protocol. In other words, switches are self-learning. This capability is
accomplished as follows:
1. The switch table is initially empty.
2. For each incoming frame received on an interface, the switch stores in its table
(1) the MAC address in the frames source address field, (2) the interface from
which the frame arrived, and (3) the current time. In this manner the switch
records in its table the LAN segment on which the sender resides. If every host
in the LAN eventually sends a frame, then every host will eventually get
recorded in the table.
3. The switch deletes an address in the table if no frames are received with that
address as the source address after some period of time (the aging time). In
this manner, if a PC is replaced by another PC (with a different adapter), the
MAC address of the original PC will eventually be purged from the switch
table.
Lets walk through the self-learning property for the uppermost switch in
Figure 5.15 and its corresponding switch table in Figure 5.22. Suppose at time 9:39
a frame with source address 01-12-23-34-45-56 arrives from interface 2. Suppose
that this address is not in the switch table. Then the switch adds a new entry to the
table, as shown in Figure 5.23.
Continuing with this same example, suppose that the aging time for this switch
is 60 minutes, and no frames with source address 62-FE-F7-11-89-A3 arrive to the
switch between 9:32 and 10:32. Then at time 10:32, the switch removes this address
from its table.
478 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Figure 5.23 Switch learns about the location of an adapter with address
01-12-23-34-45-56
Address Interface Time
01-12-23-34-45-56 2 9:39
62-FE-F7-11-89-A3 1 9:32
7C-BA-B2-B4-91-10 3 9:36
. . .
Switches are plug-and-play devices because they require no intervention
from a network administrator or user. A network administrator wanting to install
a switch need do nothing more than connect the LAN segments to the switch
interfaces. The administrator need not configure the switch tables at the time of
installation or when a host is removed from one of the LAN segments. Switches
are also full-duplex, meaning any switch interface can send and receive at the
same time.
Properties of Link-Layer Switching
Having described the basic operation of a link-layer switch, lets now consider their
features and properties. We can identify several advantages of using switches, rather
than broadcast links such as buses or hub-based star topologies:
Elimination of collisions. In a LAN built from switches (and without hubs), there
is no wasted bandwidth due to collisions! The switches buffer frames and never
transmit more than one frame on a segment at any one time. As with a router, the
maximum aggregate throughput of a switch is the sum of all the switch interface
rates. Thus, switches provide a significant performance improvement over LANs
with broadcast links.
Heterogeneous links. Because a switch isolates one link from another, the differ-
ent links in the LAN can operate at different speeds and can run over different
media. For example, the uppermost switch in Figure 5.22 might have three
1 Gbps 1000BASE-T copper links, two 100 Mbps 100BASE-FX fiber links, and
one 100BASE-T copper link. Thus, a switch is ideal for mixing legacy equip-
ment with new equipment.
Management. In addition to providing enhanced security (see sidebar on Focus
on Security), a switch also eases network management. For example, if an
adapter malfunctions and continually sends Ethernet frames (called a jabbering
adapter), a switch can detect the problem and internally disconnect the mal-
functioning adapter. With this feature, the network administrator need not get
out of bed and drive back to work in order to correct the problem. Similarly, a
cable cut disconnects only that host that was using the cut cable to connect to
the switch. In the days of coaxial cable, many a network manager spent hours
walking the line (or more accurately, crawling the floor) to find the cable
break that brought down the entire network. As discussed in Chapter 9 (Net-
work Management), switches also gather statistics on bandwidth usage, collision
rates, and traffic types, and make this information available to the network
manager. This information can be used to debug and correct problems, and to
plan how the LAN should evolve in the future. Researchers are exploring adding
yet more management functionality into Ethernet LANs in prototype deploy-
ments [Casado 2007; Koponen 2011].
5.4 SWITCHED LOCAL AREA NETWORKS 479
Switches Versus Routers
As we learned in Chapter 4, routers are store-and-forward packet switches that for-
ward packets using network-layer addresses. Although a switch is also a store-and-
forward packet switch, it is fundamentally different from a router in that it forwards
packets using MAC addresses. Whereas a router is a layer-3 packet switch, a switch
is a layer-2 packet switch.
Even though switches and routers are fundamentally different, network admin-
istrators must often choose between them when installing an interconnection device.
For example, for the network in Figure 5.15, the network administrator could just as
easily have used a router instead of a switch to connect the department LANs,
servers, and internet gateway router. Indeed, a router would permit interdepartmen-
tal communication without creating collisions. Given that both switches and routers
are candidates for interconnection devices, what are the pros and cons of the two
approaches?
First consider the pros and cons of switches. As mentioned above, switches are
plug-and-play, a property that is cherished by all the overworked network adminis-
trators of the world. Switches can also have relatively high filtering and forwarding
ratesas shown in Figure 5.24, switches have to process frames only up through
layer 2, whereas routers have to process datagrams up through layer 3. On the other
480 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
SNIFFING A SWITCHED LAN: SWITCH POISONING
When a host is connected to a switch, it typically only receives frames that are being
explicity sent to it. For example, consider a switched LAN in Figure 5.17. When host
A sends a frame to host B, and there is an entry for host B in the switch table, then
the switch will forward the frame only to host B. If host C happens to be running a
sniffer, host C will not be able to sniff this A-to-B frame. Thus, in a switched-LAN envi-
ronment (in contrast to a broadcast link environment such as 802.11 LANs or
hubbased Ethernet LANs), it is more difficult for an attacker to sniff frames. However,
because the switch broadcasts frames that have destination addresses that are not in the
switch table, the sniffer at C can still sniff some frames that are not explicitly addressed
to C. Furthermore, a sniffer will be able sniff all Ethernet broadcast frames with broad-
cast destination address FFFFFFFFFFFF. A well-known attack against a switch,
called switch poisoning, is to send tons of packets to the switch with many different
bogus source MAC addresses, thereby filling the switch table with bogus entries and
leaving no room for the MAC addresses of the legitimate hosts. This causes the switch
to broadcast most frames, which can then be picked up by the sniffer [Skoudis 2006].
As this attack is rather involved even for a sophisticated attacker, switches are signifi-
cantly less vulnerable to sniffing than are hubs and wireless LANs.
FOCUS ON SECURITY
hand, to prevent the cycling of broadcast frames, the active topology of a switched
network is restricted to a spanning tree. Also, a large switched network would
require large ARP tables in the hosts and routers and would generate substantial
ARP traffic and processing. Furthermore, switches are susceptible to broadcast
stormsif one host goes haywire and transmits an endless stream of Ethernet
broadcast frames, the switches will forward all of these frames, causing the entire
network to collapse.
Now consider the pros and cons of routers. Because network addressing is
often hierarchical (and not flat, as is MAC addressing), packets do not normally
cycle through routers even when the network has redundant paths. (However,
packets can cycle when router tables are misconfigured; but as we learned in
Chapter 4, IP uses a special datagram header field to limit the cycling.) Thus,
packets are not restricted to a spanning tree and can use the best path between
source and destination. Because routers do not have the spanning tree restriction,
they have allowed the Internet to be built with a rich topology that includes, for
example, multiple active links between Europe and North America. Another fea-
ture of routers is that they provide firewall protection against layer-2 broadcast
storms. Perhaps the most significant drawback of routers, though, is that they are
not plug-and-playthey and the hosts that connect to them need their IP
addresses to be configured. Also, routers often have a larger per-packet processing
time than switches, because they have to process up through the layer-3 fields.
Finally, there are two different ways to pronounce the word router, either as
rootor or as rowter, and people waste a lot of time arguing over the proper
pronunciation [Perlman 1999].
Given that both switches and routers have their pros and cons (as summarized
in Table 5.1), when should an institutional network (for example, a university cam-
pus network or a corporate campus network) use switches, and when should it use
Host
Application
Host
Transport
Network
Link
Physical
Link
Physical
Network
Switch Router
Link
Physical
Application
Transport
Network
Link
Physical
Figure 5.24 Packet processing in switches, routers, and hosts
5.4 SWITCHED LOCAL AREA NETWORKS 481
routers? Typically, small networks consisting of a few hundred hosts have a few
LAN segments. Switches suffice for these small networks, as they localize traffic and
increase aggregate throughput without requiring any configuration of IP addresses.
But larger networks consisting of thousands of hosts typically include routers within
the network (in addition to switches). The routers provide a more robust isolation of
traffic, control broadcast storms, and use more intelligent routes among the hosts
in the network.
For more discussion of the pros and cons of switched versus routed networks,
as well as a discussion of how switched LAN technology can be extended to accom-
modate two orders of magnitude more hosts than todays Ethernets, see [Meyers
2004; Kim 2008].
5.4.4 Virtual Local Area Networks (VLANs)
In our earlier discussion of Figure 5.15, we noted that modern institutional LANs
are often configured hierarchically, with each workgroup (department) having its
own switched LAN connected to the switched LANs of other groups via a switch
hierarchy. While such a configuration works well in an ideal world, the real
world is often far from ideal. Three drawbacks can be identified in the configura-
tion in Figure 5.15:
Lack of traffic isolation. Although the hierarchy localizes group traffic to
within a single switch, broadcast traffic (e.g., frames carrying ARP and DHCP
messages or frames whose destination has not yet been learned by a self-
learning switch) must still traverse the entire institutional network. Limiting
the scope of such broadcast traffic would improve LAN performance. Perhaps
more importantly, it also may be desirable to limit LAN broadcast traffic for
security/privacy reasons. For example, if one group contains the companys
executive management team and another group contains disgruntled employ-
ees running Wireshark packet sniffers, the network manager may well prefer
that the executives traffic never even reaches employee hosts. This type of
isolation could be provided by replacing the center switch in Figure 5.15 with
482 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
Hubs Routers Switches
Traffic isolation No Yes Yes
Plug and play Yes No Yes
Optimal routing No Yes No
Table 5.1 Comparison of the typical features of popular interconnection
devices
a router. Well see shortly that this isolation also can be achieved via a
switched (layer 2) solution
Inefficient use of switches. If instead of three groups, the institution had 10
groups, then 10 first-level switches would be required. If each group were
small, say less than 10 people, then a single 96-port switch would likely be
large enough to accommodate everyone, but this single switch would not pro-
vide traffic isolation.
Managing users. If an employee moves between groups, the physical cabling
must be changed to connect the employee to a different switch in Figure 5.15.
Employees belonging to two groups make the problem even harder.
Fortunately, each of these difficulties can be handled by a switch that supports
virtual local area networks (VLANs). As the name suggests, a switch that sup-
ports VLANs allows multiple virtual local area networks to be defined over a sin-
gle physical local area network infrastructure. Hosts within a VLAN communicate
with each other as if they (and no other hosts) were connected to the switch. In a
port-based VLAN, the switchs ports (interfaces) are divided into groups by the
network manager. Each group constitutes a VLAN, with the ports in each VLAN
forming a broadcast domain (i.e., broadcast traffic from one port can only reach
other ports in the group). Figure 5.25 shows a single switch with 16 ports. Ports
2 to 8 belong to the EE VLAN, while ports 9 to 15 belong to the CS VLAN (ports
1 and 16 are unassigned). This VLAN solves all of the difficulties noted above
EE and CS VLAN frames are isolated from each other, the two switches in Fig-
ure 5.15 have been replaced by a single switch, and if the user at switch port 8
joins the CS Department, the network operator simply reconfigures the VLAN
software so that port 8 is now associated with the CS VLAN. One can easily
1
Electrical Engineering
(VLAN ports 28)
Computer Science
(VLAN ports 915)
9 15
2 4 8 10 16
Figure 5.25 A single switch with two configured VLANs
5.4 SWITCHED LOCAL AREA NETWORKS 483
imagine how the VLAN switch is configured and operatesthe network manager
declares a port to belong to a given VLAN (with undeclared ports belonging to a
default VLAN) using switch management software, a table of port-to-VLAN
mappings is maintained within the switch; and switch hardware only delivers
frames between ports belonging to the same VLAN.
But by completely isolating the two VLANs, we have introduced a new diffi-
culty! How can traffic from the EE Department be sent to the CS Department?
One way to handle this would be to connect a VLAN switch port (e.g., port 1 in
Figure 5.25) to an external router and configure that port to belong both the EE
and CS VLANs. In this case, even though the EE and CS departments share the
same physical switch, the logical configuration would look as if the EE and CS
departments had separate switches connected via a router. An IP datagram going
from the EE to the CS department would first cross the EE VLAN to reach the
router and then be forwarded by the router back over the CS VLAN to the CS
host. Fortunately, switch vendors make such configurations easy for the network
manager by building a single device that contains both a VLAN switch and a
router, so a separate external router is not needed. A homework problem at the end
of the chapter explores this scenario in more detail.
Returning again to Figure 5.15, lets now suppose that rather than having a
separate Computer Engineering department, some EE and CS faculty are housed
in a separate building, where (of course!) they need network access, and (of
course!) theyd like to be part of their departments VLAN. Figure 5.26 shows a
second 8-port switch, where the switch ports have been defined as belonging to
the EE or the CS VLAN, as needed. But how should these two switches be inter-
connected? One easy solution would be to define a port belonging to the CS
VLAN on each switch (similarly for the EE VLAN) and to connect these ports to
each other, as shown in Figure 5.26(a). This solution doesnt scale, however, since
N VLANS would require N ports on each switch simply to interconnect the two
switches.
A more scalable approach to interconnecting VLAN switches is known as
VLAN trunking. In the VLAN trunking approach shown in Figure 5.26(b), a spe-
cial port on each switch (port 16 on the left switch and port 1 on the right switch) is
configured as a trunk port to interconnect the two VLAN switches. The trunk port
belongs to all VLANs, and frames sent to any VLAN are forwarded over the trunk
link to the other switch. But this raises yet another question: How does a switch
know that a frame arriving on a trunk port belongs to a particular VLAN? The IEEE
has defined an extended Ethernet frame format, 802.1Q, for frames crossing a
VLAN trunk. As shown in Figure 5.27, the 802.1Q frame consists of the standard
Ethernet frame with a four-byte VLAN tag added into the header that carries the
identity of the VLAN to which the frame belongs. The VLAN tag is added into a
frame by the switch at the sending side of a VLAN trunk, parsed, and removed by
the switch at the receiving side of the trunk. The VLAN tag itself consists of a 2-byte
484 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
1
16
1
8
1
Electrical Engineering
(VLAN ports 28)
b.
a.
Electrical Engineering
(VLAN ports 2, 3, 6)
Trunk
link
Computer Science
(VLAN ports 915)
9 15
2 4 8 10 16
1
2
3
4
5
6 8
7
Computer Science
(VLAN ports 4, 5, 7)
Figure 5.26 Connecting two VLAN switches with two VLANs: (a) two
cables (b) trunked
Preamble CRC
Dest.
address
Source
address
Type
Data
Preamble CRC
Dest.
address
Source
address
Type
Tag Control Information
Tag Protocol Identifier
Recomputed
CRT
Data
Figure 5.27 Original Ethernet frame (top), 802.1Q-tagged Ethernet
VLAN frame (below)
5.4 SWITCHED LOCAL AREA NETWORKS 485
Tag Protocol Identifier (TPID) field (with a fixed hexadecimal value of 81-00), a
2-byte Tag Control Information field that contains a 12-bit VLAN identifier field,
and a 3-bit priority field that is similar in intent to the IP datagram TOS field.
In this discussion, weve only briefly touched on VLANs and have focused
on port-based VLANs. We should also mention that VLANs can be defined in
several other ways. In MAC-based VLANs, the network manager specifies the
set of MAC addresses that belong to each VLAN; whenever a device attaches to
a port, the port is connected into the appropriate VLAN based on the MAC
address of the device. VLANs can also be defined based on network-layer proto-
cols (e.g., IPv4, IPv6, or Appletalk) and other criteria. See the 802.1Q standard
[IEEE 802.1q 2005] for more details.
5.5 Link Virtualization: A Network as a Link
Layer
Because this chapter concerns link-layer protocols, and given that were now near-
ing the chapters end, lets reflect on how our understanding of the term link has
evolved. We began this chapter by viewing the link as a physical wire connecting
two communicating hosts. In studying multiple access protocols, we saw that multi-
ple hosts could be connected by a shared wire and that the wire connecting the
hosts could be radio spectra or other media. This led us to consider the link a bit
more abstractly as a channel, rather than as a wire. In our study of Ethernet LANs
(Figure 5.15) we saw that the interconnecting media could actually be a rather com-
plex switched infrastructure. Throughout this evolution, however, the hosts them-
selves maintained the view that the interconnecting medium was simply a link-layer
channel connecting two or more hosts. We saw, for example, that an Ethernet host
can be blissfully unaware of whether it is connected to other LAN hosts by a single
short LAN segment (Figure 5.17) or by a geographically dispersed switched LAN
(Figure 5.15) or by a VLAN (Figure 5.26).
In the case of a dialup modem connection between two hosts, the link connect-
ing the two hosts is actually the telephone networka logically separate, global
telecommunications network with its own switches, links, and protocol stacks for
data transfer and signaling. From the Internet link-layer point of view, however, the
dial-up connection through the telephone network is viewed as a simple wire. In
this sense, the Internet virtualizes the telephone network, viewing the telephone net-
work as a link-layer technology providing link-layer connectivity between two
Internet hosts. You may recall from our discussion of overlay networks in Chapter 2
that an overlay network similarly views the Internet as a means for providing con-
nectivity between overlay nodes, seeking to overlay the Internet in the same way
that the Internet overlays the telephone network.
486 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
In this section, well consider Multiprotocol Label Switching (MPLS) net-
works. Unlike the circuit-switched telephone network, MPLS is a packet-switched,
virtual-circuit network in its own right. It has its own packet formats and forward-
ing behaviors. Thus, from a pedagogical viewpoint, a discussion of MPLS fits well
into a study of either the network layer or the link layer. From an Internet viewpoint,
however, we can consider MPLS, like the telephone network and switched-
Ethernets, as a link-layer technology that serves to interconnect IP devices. Thus,
well consider MPLS in our discussion of the link layer. Frame-relay and ATM
networks can also be used to interconnect IP devices, though they represent a
slightly older (but still deployed) technology and will not be covered here; see the
very readable book [Goralski 1999] for details. Our treatment of MPLS will be nec-
essarily brief, as entire books could be (and have been) written on these networks.
We recommend [Davie 2000] for details on MPLS. Well focus here primarily on
how MPLS servers interconnect to IP devices, although well dive a bit deeper into
the underlying technologies as well.
5.5.1 Multiprotocol Label Switching (MPLS)
Multiprotocol Label Switching (MPLS) evolved from a number of industry efforts in
the mid-to-late 1990s to improve the forwarding speed of IP routers by adopting a
key concept from the world of virtual-circuit networks: a fixed-length label. The goal
was not to abandon the destination-based IP datagram-forwarding infrastructure for
one based on fixed-length labels and virtual circuits, but to augment it by selectively
labeling datagrams and allowing routers to forward datagrams based on fixed-length
labels (rather than destination IP addresses) when possible. Importantly, these tech-
niques work hand-in-hand with IP, using IP addressing and routing. The IETF uni-
fied these efforts in the MPLS protocol [RFC 3031, RFC 3032], effectively blending
VC techniques into a routed datagram network.
Lets begin our study of MPLS by considering the format of a link-layer frame
that is handled by an MPLS-capable router. Figure 5.28 shows that a link-layer frame
transmitted between MPLS-capable devices has a small MPLS header added
between the layer-2 (e.g., Ethernet) header and layer-3 (i.e., IP) header. RFC 3032
PPP or Ethernet
header
MPLS header IP header Remainder of link-layer frame
Label Exp S TTL
Figure 5.28 MPLS header: Located between link- and network-layer
headers
5.5 LINK VIRTUALIZATION: A NETWORK AS A LINK LAYER 487
defines the format of the MPLS header for such links; headers are defined for ATM
and frame-relayed networks as well in other RFCs. Among the fields in the MPLS
header are the label (which serves the role of the virtual-circuit identifier that we
encountered back in Section 4.2.1), 3 bits reserved for experimental use, a single S bit,
which is used to indicate the end of a series of stacked MPLS headers (an advanced
topic that well not cover here), and a time-to-live field.
Its immediately evident from Figure 5.28 that an MPLS-enhanced frame can
only be sent between routers that are both MPLS capable (since a non-MPLS-
capable router would be quite confused when it found an MPLS header where it had
expected to find the IP header!). An MPLS-capable router is often referred to as a
label-switched router, since it forwards an MPLS frame by looking up the MPLS
label in its forwarding table and then immediately passing the datagram to the
appropriate output interface. Thus, the MPLS-capable router need not extract the
destination IP address and perform a lookup of the longest prefix match in the for-
warding table. But how does a router know if its neighbor is indeed MPLS capable,
and how does a router know what label to associate with the given IP destination?
To answer these questions, well need to take a look at the interaction among a
group of MPLS-capable routers.
In the example in Figure 5.29, routers R1 through R4 are MPLS capable. R5
and R6 are standard IP routers. R1 has advertised to R2 and R3 that it (R1) can route
to destination A, and that a received frame with MPLS label 6 will be forwarded to
destination A. Router R3 has advertised to router R4 that it can route to destinations
488 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
R4
in
label
out
label
10
12
8
A
D
A
0
0
1
dest
out
interface
R6
R5
R3
R2
D
A
0
0 0
1 1
0
R1
in
label
out
label
6
9
A
D
1
0
10
12
dest
out
interface
in
label
out
label
A 06
dest
out
interface
in
label
out
label
6 A 08
dest
out
interface
Figure 5.29 MPLS-enhanced forwarding
A and D, and that incoming frames with MPLS labels 10 and 12, respectively, will
be switched toward those destinations. Router R2 has also advertised to router R4
that it (R2) can reach destination A, and that a received frame with MPLS label 8
will be switched toward A. Note that router R4 is now in the interesting position of
having two MPLS paths to reach A: via interface 0 with outbound MPLS label 10,
and via interface 1 with an MPLS label of 8. The broad picture painted in Figure
5.29 is that IP devices R5, R6, A, and D are connected together via an MPLS infra-
structure (MPLS-capable routers R1, R2, R3, and R4) in much the same way that a
switched LAN or an ATM network can connect together IP devices. And like a
switched LAN or ATM network, the MPLS-capable routers R1 through R4 do so
without ever touching the IP header of a packet.
In our discussion above, weve not specified the specific protocol used to dis-
tribute labels among the MPLS-capable routers, as the details of this signaling are
well beyond the scope of this book. We note, however, that the IETF working group
on MPLS has specified in [RFC 3468] that an extension of the RSVP protocol,
known as RSVP-TE [RFC 3209], will be the focus of its efforts for MPLS signal-
ing. Weve also not discussed how MPLS actually computes the paths for packets
among MPLS capable routers, nor how it gathers link-state information (e.g.,
amount oflink bandwidth unreserved by MPLS) to use in these path computations.
Existing link-state routing algorithms (e.g., OSPF) have been extended to flood this
information to MPLS-capable routers. Interestingly, the actual path computation
algorithms are not standardized, and are currently vendor-specific.
Thus far, the emphasis of our discussion of MPLS has been on the fact that
MPLS performs switching based on labels, without needing to consider the IP
address of a packet. The true advantages of MPLS and the reason for current inter-
est in MPLS, however, lie not in the potential increases in switching speeds, but
rather in the new traffic management capabilities that MPLS enables. As noted
above, R4 has two MPLS paths to A. If forwarding were performed up at the IP
layer on the basis of IP address, the IP routing protocols we studied in Chapter 4
would specify only a single, least-cost path to A. Thus, MPLS provides the ability
to forward packets along routes that would not be possible using standard IP routing
protocols. This is one simple form of traffic engineering using MPLS [RFC 3346;
RFC 3272; RFC 2702; Xiao 2000], in which a network operator can override nor-
mal IP routing and force some of the traffic headed toward a given destination along
one path, and other traffic destined toward the same destination along another path
(whether for policy, performance, or some other reason).
It is also possible to use MPLS for many other purposes as well. It can be used
to perform fast restoration of MPLS forwarding paths, e.g., to reroute traffic over a
precomputed failover path in response to link failure [Kar 2000; Huang 2002; RFC
3469]. Finally, we note that MPLS can, and has, been used to implement so-called
virtual private networks (VPNs). In implementing a VPN for a customer, an ISP
uses its MPLS-enabled network to connect together the customers various net-
works. MPLS can be used to isolate both the resources and addressing used by the
5.5 LINK VIRTUALIZATION: A NETWORK AS A LINK LAYER 489
customers VPN from that of other users crossing the ISPs network; see [DeClercq
2002] for details.
Our discussion of MPLS has been brief, and we encourage you to consult the
references weve mentioned. We note that with so many possible uses for MPLS,
it appears that it is rapidly becoming the Swiss Army knife of Internet traffic
engineering!
5.6 Data Center Networking
In recent years, Internet companies such as Google, Microsoft, Facebook, and Amazon
(as well as their counterparts in Asia and Europe) have built massive data centers, each
housing tens to hundreds of thousands of hosts, and concurrently supporting many
distinct cloud applications (e.g., search, email, social networking, and e-commerce).
Each data center has its own data center network that interconnects its hosts with each
other and interconnects the data center with the Internet. In this section, we provide a
brief introduction to data center networking for cloud applications.
The cost of a large data center is huge, exceeding $12 million per month for a
100,000 host data center [Greenberg 2009a]. Of these costs, about 45 percent can be
attributed to the hosts themselves (which need to be replaced every 34 years); 25
percent to infrastructure, including transformers, uninterruptable power supplies
(UPS) systems, generators for long-term outages, and cooling systems; 15 percent
for electric utility costs for the power draw; and 15 percent for networking, includ-
ing network gear (switches, routers and load balancers), external links, and transit
traffic costs. (In these percentages, costs for equipment are amortized so that a com-
mon cost metric is applied for one-time purchases and ongoing expenses such as
power.) While networking is not the largest cost, networking innovation is the key
to reducing overall cost and maximizing performance [Greenberg 2009a].
The worker bees in a data center are the hosts: They serve content (e.g., Web
pages and videos), store emails and documents, and collectively perform massively
distributed computations (e.g., distributed index computations for search engines).
The hosts in data centers, called blades and resembling pizza boxes, are generally
commodity hosts that include CPU, memory, and disk storage. The hosts are stacked
in racks, with each rack typically having 20 to 40 blades. At the top of each rack
there is a switch, aptly named the Top of Rack (TOR) switch, that interconnects
the hosts in the rack with each other and with other switches in the data center.
Specifically, each host in the rack has a network interface card that connects to its
TOR switch, and each TOR switch has additional ports that can be connected
to other switches. Although today hosts typically have 1 Gbps Ethernet connections
to their TOR switches, 10 Gbps connections may become the norm. Each host is
also assigned its own data-center-internal IP address.
The data center network supports two types of traffic: traffic flowing between
external clients and internal hosts and traffic flowing between internal hosts. To handle
flows between external clients and internal hosts, the data center network includes one
490 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
or more border routers, connecting the data center network to the public Internet. The
data center network therefore interconnects the racks with each other and connects the
racks to the border routers. Figure 5.30 shows an example of a data center network.
Data center network design, the art of designing the interconnection network and pro-
tocols that connect the racks with each other and with the border routers, has become
an important branch of computer networking research in recent years [Al-Fares 2008;
Greenberg 2009a; Greenberg 2009b; Mydotr 2009; Guo 2009; Chen 2010; Abu-Lib-
deh 2010; Alizadeh 2010; Wang 2010; Farrington 2010; Halperin 2011; Wilson 2011;
Mudigonda 2011; Ballani 2011; Curtis 2011; Raiciu 2011].
Load Balancing
A cloud data center, such as a Google or Microsoft data center, provides many appli-
cations concurrently, such as search, email, and video applications. To support
requests from external clients, each application is associated with a publicly visible
IP address to which clients send their requests and from which they receive
responses. Inside the data center, the external requests are first directed to a load
balancer whose job it is to distribute requests to the hosts, balancing the load across
the hosts as a function of their current load. A large data center will often have several
Internet
A
1 2 3 4 5 6 7 8
C
B
Server racks
TOR switches
Tier-2 switches
Tier-1 switches
Access router
Border router
Load
balancer
Figure 5.30 A data center network with a hierarchical topology
5.6 DATA CENTER NETWORKING 491
load balancers, each one devoted to a set of specific cloud applications. Such a load
balancer is sometimes referred to as a layer-4 switch since it makes decisions
based on the destination port number (layer 4) as well as destination IP address in
the packet. Upon receiving a request for a particular application, the load balancer
forwards it to one of the hosts that handles the application. (A host may then invoke
the services of other hosts to help process the request.) When the host finishes pro-
cessing the request, it sends its response back to the load balancer, which in turn
relays the response back to the external client. The load balancer not only balances
the work load across hosts, but also provides a NAT-like function, translating the
public external IP address to the internal IP address of the appropriate host, and then
translating back for packets traveling in the reverse direction back to the clients.
This prevents clients from contacting hosts directly, which has the security benefit
of hiding the internal network structure and preventing clients from directly inter-
acting with the hosts.
Hierarchical Architecture
For a small data center housing only a few thousand hosts, a simple network con-
sisting of a border router, a load balancer, and a few tens of racks all interconnected
by a single Ethernet switch could possibly suffice. But to scale to tens to hundreds
of thousands of hosts, a data center often employs a hierarchy of routers and
switches, such as the topology shown in Figure 5.30. At the top of the hierarchy, the
border router connects to access routers (only two are shown in Figure 5.30, but
there can be many more). Below each access router there are three tiers of switches.
Each access router connects to a top-tier switch, and each top-tier switch connects
to multiple second-tier switches and a load balancer. Each second-tier switch in turn
connects to multiple racks via the racks TOR switches (third-tier switches). All
links typically use Ethernet for their link-layer and physical-layer protocols, with a
mix of copper and fiber cabling. With such a hierarchical design, it is possible to
scale a data center to hundreds of thousands of hosts.
Because it is critical for a cloud application provider to continually provide
applications with high availability, data centers also include redundant network
equipment and redundant links in their designs (not shown in Figure 5.30).
For example, each TOR switch can connect to two tier-2 switches, and each
access router, tier-1 switch, and tier-2 switch can be duplicated and integrated
into the design [Cisco 2012; Greenberg 2009b]. In the hierarchical design in
Figure 5.30, observe that the hosts below each access router form a single sub-
net. In order to localize ARP broadcast traffic, each of these subnets is further
partitioned into smaller VLAN subnets, each comprising a few hundred hosts
[Greenberg 2009a].
Although the conventional hierarchical architecture just described solves the
problem of scale, it suffers from limited host-to-host capacity [Greenberg 2009b]. To
understand this limitation, consider again Figure 5.30, and suppose each host connects
492 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
to its TOR switch with a 1 Gbps link, whereas the links between switches are 10 Gbps
Ethernet links. Two hosts in the same rack can always communicate at a full 1 Gbps,
limited only by the rate of the hosts network interface cards. However, if there are
many simultaneous flows in the data center network, the maximum rate between two
hosts in different racks can be much less. To gain insight into this issue, consider a traf-
fic pattern consisting of 40 simultaneous flows between 40 pairs of hosts in different
racks. Specifically, suppose each of 10 hosts in rack 1 in Figure 5.30 sends a flow to a
corresponding host in rack 5. Similarly, there are ten simultaneous flows between pairs
of hosts in racks 2 and 6, ten simultaneous flows between racks 3 and 7, and ten simul-
taneous flows between racks 4 and 8. If each flow evenly shares a links capacity with
other flows traversing that link, then the 40 flows crossing the 10 Gbps A-to-B link (as
well as the 10 Gbps B-to-C link) will each only receive 10 Gbps / 40 = 250 Mbps,
which is significantly less than the 1 Gbps network interface card rate. The problem
becomes even more acute for flows between hosts that need to travel higher up the
hierarchy. One possible solution to this limitation is to deploy higher-rate switches and
routers. But this would significantly increase the cost of the data center, because
switches and routers with high port speeds are very expensive.
Supporting high-bandwidth host-to-host communication is important because a
key requirement in data centers is flexibility in placement of computation and services
[Greenberg 2009b; Farrington 2010]. For example, a large-scale Internet search
engine may run on thousands of hosts spread across multiple racks with significant
bandwidth requirements between all pairs of hosts. Similarly, a cloud computing serv-
ice such as EC2 may wish to place the multiple virtual machines comprising a cus-
tomers service on the physical hosts with the most capacity irrespective of their
location in the data center. If these physical hosts are spread across multiple racks, net-
work bottlenecks as described above may result in poor performance.
Trends in Data Center Networking
In order to reduce the cost of data centers, and at the same time improve their delay and
throughput performance, Internet cloud giants such as Google, Facebook, Amazon, and
Microsoft are continually deploying new data center network designs. Although these
designs are proprietary, many important trends can nevertheless be identified.
One such trend is to deploy new interconnection architectures and network proto-
cols that overcome the drawbacks of the traditional hierarchical designs. One such
approach is to replace the hierarchy of switches and routers with a fully connected
topology [Al-Fares 2008; Greenberg 2009b; Guo 2009], such as the topology shown in
Figure 5.31. In this design, each tier-1 switch connects to all of the tier-2 switches so
that (1) host-to-host traffic never has to rise above the switch tiers, and (2) with n tier-1
switches, between any two tier-2 switches there are n disjoint paths. Such a design can
significantly improve the host-to-host capacity. To see this, consider again our example
of 40 flows. The topology in Figure 5.31 can handle such a flow pattern since there are
four distinct paths between the first tier-2 switch and the second tier-2 switch, together
5.6 DATA CENTER NETWORKING 493
providing an aggregate capacity of 40 Gbps between the first two tier-2 switches. Such
a design not only alleviates the host-to-host capacity limitation, but also creates a more
flexible computation and service environment in which communication between any
two racks not connected to the same switch is logically equivalent, irrespective of their
locations in the data center.
Another major trend is to employ shipping containerbased modular data centers
(MDCs) [YouTube 2009; Waldrop 2007]. In an MDC, a factory builds, within a stan-
dard 12-meter shipping container, a mini data center and ships the container to the
data center location. Each container has up to a few thousand hosts, stacked in tens of
racks, which are packed closely together. At the data center location, multiple contain-
ers are interconnected with each other and also with the Internet. Once a prefabricated
container is deployed at a data center, it is often difficult to service. Thus, each con-
tainer is designed for graceful performance degradation: as components (servers and
switches) fail over time, the container continues to operate but with degraded perform-
ance. When many components have failed and performance has dropped below a
threshold, the entire container is removed and replaced with a fresh one.
Building a data center out of containers creates new networking challenges.
With an MDC, there are two types of networks: the container-internal networks
within each of the containers and the core network connecting each container [Guo
2009; Farrington 2010]. Within each container, at the scale of up to a few thousand
hosts, it is possible to build a fully connected network (as described above) using
inexpensive commodity Gigabit Ethernet switches. However, the design of the core
network, interconnecting hundreds to thousands of containers while providing high
host-to-host bandwidth across containers for typical workloads, remains a challeng-
ing problem. A hybrid electrical/optical switch architecture for interconnecting the
containers is proposed in [Farrington 2010].
When using highly interconnected topologies, one of the major issues is design-
ing routing algorithms among the switches. One possibility [Greenberg 2009b] is to
494 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
1 2 3 4 5 6 7 8
Server racks
TOR switches
Tier-2 switches
Tier-1 switches
Figure 5.31 Highly-interconnected data network topology
use a form of random routing. Another possibility [Guo 2009] is to deploy multiple
network interface cards in each host, connect each host to multiple low-cost commod-
ity switches, and allow the hosts themselves to intelligently route traffic among the
switches. Variations and extensions of these approaches are currently being deployed
in contemporary data centers. Many more innovations in data center design are likely
to come; interested readers are encouraged to read the many recent papers on data cen-
ter network design.
5.7 Retrospective: A Day in the Life of a Web Page
Request
Now that weve covered the link layer in this chapter, and the network, transport and
application layers in earlier chapters, our journey down the protocol stack is com-
plete! In the very beginning of this book (Section 1.1), we wrote much of this book
is concerned with computer network protocols, and in the first five chapters, weve
certainly seen that this is indeed the case! Before heading into the topical chapters in
second part of this book, wed like to wrap up our journey down the protocol stack
by taking an integrated, holistic view of the protocols weve learned about so far.
One way then to take this big picture view is to identify the many (many!) proto-
cols that are involved in satisfying even the simplest request: downloading a web page.
Figure 5.32 illustrates our setting: a student, Bob, connects a laptop to his schools
Ethernet switch and downloads a web page (say the home page of www.google.com).
As we now know, theres a lot going on under the hood to satisfy this seemingly
simple request. A Wireshark lab at the end of this chapter examines trace files contain-
ing a number of the packets involved in similar scenarios in more detail.
5.7.1 Getting Started: DHCP, UDP, IP, and Ethernet
Lets suppose that Bob boots up his laptop and then connects it to an Ethernet cable
connected to the schools Ethernet switch, which in turn is connected to the schools
router, as shown in Figure 5.32. The schools router is connected to an ISP, in this
example, comcast.net. In this example, comcast.net is providing the DNS service
for the school; thus, the DNS server resides in the Comcast network rather than the
school network. Well assume that the DHCP server is running within the router, as
is often the case.
When Bob first connects his laptop to the network, he cant do anything (e.g.,
download a Web page) without an IP address. Thus, the first network-related action
taken by Bobs laptop is to run the DHCP protocol to obtain an IP address, as well
as other information, from the local DHCP server:
1. The operating system on Bobs laptop creates a DHCP request message (Sec-
tion 4.4.2) and puts this message within a UDP segment (Section 3.3) with
destination port 67 (DHCP server) and source port 68 (DHCP client). The UDP
segment is then placed within an IP datagram (Section 4.4.1) with a broadcast
5.7 RETROSPECTIVE: A DAY IN THE LIFE OF A WEB PAGE REQUEST 495
VideoNote
A day in the life of a
Web page request
www.google.com
IP destination address (255.255.255.255) and a source IP address of 0.0.0.0,
since Bobs laptop doesnt yet have an IP address.
2. The IP datagram containing the DHCP request message is then placed within
an Ethernet frame (Section 5.4.2). The Ethernet frame has a destination MAC
addresses of FF:FF:FF:FF:FF:FF so that the frame will be broadcast to all
devices connected to the switch (hopefully including a DHCP server); the
frames source MAC address is that of Bobs laptop, 00:16:D3:23:68:8A.
3. The broadcast Ethernet frame containing the DHCP request is the first frame
sent by Bobs laptop to the Ethernet switch. The switch broadcasts the incom-
ing frame on all outgoing ports, including the port connected to the router.
4. The router receives the broadcast Ethernet frame containing the DHCP request
on its interface with MAC address 00:22:6B:45:1F:1B and the IP datagram is
extracted from the Ethernet frame. The datagrams broadcast IP destination
address indicates that this IP datagram should be processed by upper layer proto-
cols at this node, so the datagrams payload (a UDP segment) is thus demulti-
plexed (Section 3.2) up to UDP, and the DHCP request message is extracted
from the UDP segment. The DHCP server now has the DHCP request message.
5. Lets suppose that the DHCP server running within the router can allocate IP
addresses in the CIDR (Section 4.4.2) block 68.85.2.0/24. In this example, all
IP addresses used within the school are thus within Comcasts address block.
496 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
00:22:6B:45:1F:1B
68.85.2.1
00:16:D3:23:68:8A
68.85.2.101
comcast.net
DNS server
68.87.71.226
www.google.com
Web server
64.233.169.105
School network
68.80.2.0/24
Comcasts network
68.80.0.0/13
Googles network
64.233.160.0/19
17
813
1824
1417
Figure 5.32 A day in the life of a Web page request: network setting
and actions
www.google.com
Lets suppose the DHCP server allocates address 68.85.2.101 to Bobs laptop.
The DHCP server creates a DHCP ACK message (Section 4.4.2) containing
this IP address, as well as the IP address of the DNS server (68.87.71.226), the
IP address for the default gateway router (68.85.2.1), and the subnet block
(68.85.2.0/24) (equivalently, the network mask). The DHCP message is put
inside a UDP segment, which is put inside an IP datagram, which is put inside
an Ethernet frame. The Ethernet frame has a source MAC address of the
routers interface to the home network (00:22:6B:45:1F:1B) and a destination
MAC address of Bobs laptop (00:16:D3:23:68:8A).
6. The Ethernet frame containing the DHCP ACK is sent (unicast) by the router
to the switch. Because the switch is self-learning (Section 5.4.3) and previ-
ously received an Ethernet frame (containing the DHCP request) from Bobs
laptop, the switch knows to forward a frame addressed to 00:16:D3:23:68:8A
only to the output port leading to Bobs laptop.
7. Bobs laptop receives the Ethernet frame containing the DHCP ACK, extracts
the IP datagram from the Ethernet frame, extracts the UDP segment from the
IP datagram, and extracts the DHCP ACK message from the UDP segment.
Bobs DHCP client then records its IP address and the IP address of its DNS
server. It also installs the address of the default gateway into its IP forwarding
table (Section 4.1). Bobs laptop will send all datagrams with destination
address outside of its subnet 68.85.2.0/24 to the default gateway. At this point,
Bobs laptop has initialized its networking components and is ready to begin
processing the Web page fetch. (Note that only the last two DHCP steps of the
four presented in Chapter 4 are actually necessary.)
5.7.2 Still Getting Started: DNS and ARP
When Bob types the URL for www.google.com into his Web browser, he begins the
long chain of events that will eventually result in Googles home page being dis-
played by his Web browser. Bobs Web browser begins the process by creating a
TCP socket (Section 2.7) that will be used to send the HTTP request (Section 2.2)
to www.google.com. In order to create the socket, Bobs laptop will need to know
the IP address of www.google.com. We learned in Section 2.5, that the DNS proto-
col is used to provide this name-to-IP-address translation service.
8. The operating system on Bobs laptop thus creates a DNS query message (Section
2.5.3), putting the string www.google.com in the question section of the DNS
message. This DNS message is then placed within a UDP segment with a destina-
tion port of 53 (DNS server). The UDP segment is then placed within an IP data-
gram with an IP destination address of 68.87.71.226 (the address of the DNS server
returned in the DHCPACK in step 5) and a source IP address of 68.85.2.101.
9. Bobs laptop then places the datagram containing the DNS query message in
an Ethernet frame. This frame will be sent (addressed, at the link layer) to the
gateway router in Bobs schools network. However, even though Bobs laptop
5.7 RETROSPECTIVE: A DAY IN THE LIFE OF A WEB PAGE REQUEST 497
www.google.com
www.google.com
www.google.com
www.google.com
knows the IP address of the schools gateway router (68.85.2.1) via the DHCP
ACK message in step 5 above, it doesnt know the gateway routers MAC
address. In order to obtain the MAC address of the gateway router, Bobs lap-
top will need to use the ARP protocol (Section 5.4.1).
10. Bobs laptop creates an ARP query message with a target IP address of
68.85.2.1 (the default gateway), places the ARP message within an Ethernet
frame with a broadcast destination address (FF:FF:FF:FF:FF:FF) and sends the
Ethernet frame to the switch, which delivers the frame to all connected
devices, including the gateway router.
11. The gateway router receives the frame containing the ARP query message on the
interface to the school network, and finds that the target IP address of 68.85.2.1 in
the ARP message matches the IP address of its interface. The gateway router thus
prepares an ARP reply, indicating that its MAC address of 00:22:6B:45:1F:1B
corresponds to IP address 68.85.2.1. It places the ARP reply message in an
Ethernet frame, with a destination address of 00:16:D3:23:68:8A (Bobs laptop)
and sends the frame to the switch, which delivers the frame to Bobs laptop.
12. Bobs laptop receives the frame containing the ARP reply message and extracts
the MAC address of the gateway router (00:22:6B:45:1F:1B) from the ARP
reply message.
13. Bobs laptop can now ( finally!) address the Ethernet frame containing the DNS
query to the gateway routers MAC address. Note that the IP datagram in this frame
has an IP destination address of 68.87.71.226 (the DNS server), while the frame has
a destination address of 00:22:6B:45:1F:1B (the gateway router). Bobs laptop
sends this frame to the switch, which delivers the frame to the gateway router.
5.7.3 Still Getting Started: Intra-Domain Routing to the
DNS Server
14. The gateway router receives the frame and extracts the IP datagram containing
the DNS query. The router looks up the destination address of this datagram
(68.87.71.226) and determines from its forwarding table that the datagram should
be sent to the leftmost router in the Comcast network in Figure 5.32. The IP data-
gram is placed inside a link-layer frame appropriate for the link connecting the
schools router to the leftmost Comcast router and the frame is sent over this link.
15. The leftmost router in the Comcast network receives the frame, extracts the IP
datagram, examines the datagrams destination address (68.87.71.226) and
determines the outgoing interface on which to forward the datagram towards
the DNS server from its forwarding table, which has been filled in by Com-
casts intra-domain protocol (such as RIP, OSPF or IS-IS, Section 4.6) as well
as the Internets inter-domain protocol, BGP.
16. Eventually the IP datagram containing the DNS query arrives at the DNS server.
The DNS server extracts the DNS query message, looks up the name
www.google.com in its DNS database (Section 2.5), and finds the DNS resource
498 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
www.google.com
record that contains the IP address (64.233.169.105) for www.google.com.
(assuming that it is currently cached in the DNS server). Recall that this cached
data originated in the authoritative DNS server (Section 2.5.2) for googlecom.
The DNS server forms a DNS reply message containing this hostname-to-IP-
address mapping, and places the DNS reply message in a UDP segment, and the
segment within an IP datagram addressed to Bobs laptop (68.85.2.101). This
datagram will be forwarded back through the Comcast network to the schools
router and from there, via the Ethernet switch to Bobs laptop.
17. Bobs laptop extracts the IP address of the server www.google.com from the
DNS message. Finally, after a lot of work, Bobs laptop is now ready to contact
the www.google.com server!
5.7.4 Web Client-Server Interaction: TCP and HTTP
18. Now that Bobs laptop has the IP address of www.google.com, it can create the
TCP socket (Section 2.7) that will be used to send the HTTP GET message
(Section 2.2.3) to www.google.com. When Bob creates the TCP socket, the TCP
in Bobs laptop must first perform a three-way handshake (Section 3.5.6) with
the TCP in www.google.com. Bobs laptop thus first creates a TCP SYN segment
with destination port 80 (for HTTP), places the TCP segment inside an IP data-
gram with a destination IP address of 64.233.169.105 (www.google.com), places
the datagram inside a frame with a destination MAC address of
00:22:6B:45:1F:1B (the gateway router) and sends the frame to the switch.
19. The routers in the school network, Comcasts network, and Googles network
forward the datagram containing the TCP SYN towards www.google.com,
using the forwarding table in each router, as in steps 1416 above. Recall that
the router forwarding table entries governing forwarding of packets over the
inter-domain link between the Comcast and Google networks are determined
by the BGP protocol (Section 4.6.3).
20. Eventually, the datagram containing the TCP SYN arrives at www.google.com.
The TCP SYN message is extracted from the datagram and demultiplexed to the
welcome socket associated with port 80. A connection socket (Section 2.7) is
created for the TCP connection between the Google HTTP server and Bobs
laptop. A TCP SYNACK (Section 3.5.6) segment is generated, placed inside a
datagram addressed to Bobs laptop, and finally placed inside a link-layer frame
appropriate for the link connecting www.google.com to its first-hop router.
21. The datagram containing the TCP SYNACK segment is forwarded through the
Google, Comcast, and school networks, eventually arriving at the Ethernet card
in Bobs laptop. The datagram is demultiplexed within the operating system to
the TCP socket created in step 18, which enters the connected state.
22. With the socket on Bobs laptop now (finally!) ready to send bytes to www.google
.com, Bobs browser creates the HTTP GET message (Section 2.2.3) containing the
URL to be fetched. The HTTP GET message is then written into the socket, with the
5.7 RETROSPECTIVE: A DAY IN THE LIFE OF A WEB PAGE REQUEST 499
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
www.google.com
GET message becoming the payload of a TCP segment. The TCP segment is placed
in a datagram and sent and delivered to www.google.com as in steps 1820 above.
23. The HTTP server at www.google.com reads the HTTP GET message from the
TCP socket, creates an HTTP response message (Section 2.2), places the
requested Web page content in the body of the HTTP response message, and
sends the message into the TCP socket.
24. The datagram containing the HTTP reply message is forwarded through the Google,
Comcast, and school networks, and arrives at Bobs laptop. Bobs Web browser pro-
gram reads the HTTP response from the socket, extracts the html for the Web page
from the body of the HTTP response, and finally ( finally!) displays the Web page!
Our scenario above has covered a lot of networking ground! If youve understood
most or all of the above example, then youve also covered a lot of ground since you
first read Section 1.1, where we wrote much of this book is concerned with computer
network protocols and you may have wondered what a protocol actually was! As
detailed as the above example might seem, weve omitted a number of possible addi-
tional protocols (e.g., NAT running in the schools gateway router, wireless access to
the schools network, security protocols for accessing the school network or encrypt-
ing segments or datagrams, network management protocols), and considerations (Web
caching, the DNS hierarchy) that one would encounter in the public Internet. Well
cover a number of these topics and more in the second part of this book.
Lastly, we note that our example above was an integrated and holistic, but also
very nuts and bolts, view of many of the protocols that weve studied in the first
part of this book. The example focused more on the how than the why. For a
broader, more reflective view on the design of network protocols in general, see
[Clark 1988, RFC 5218].
5.8 Summary
In this chapter, weve examined the link layerits services, the principles underly-
ing its operation, and a number of important specific protocols that use these princi-
ples in implementing link-layer services.
We saw that the basic service of the link layer is to move a network-layer data-
gram from one node (host, switch, router, WiFi access point) to an adjacent node.
We saw that all link-layer protocols operate by encapsulating a network-layer data-
gram within a link-layer frame before transmitting the frame over the link to the
adjacent node. Beyond this common framing function, however, we learned that dif-
ferent link-layer protocols provide very different link access, delivery, and transmis-
sion services. These differences are due in part to the wide variety of link types over
which link-layer protocols must operate. A simple point-to-point link has a single
sender and receiver communicating over a single wire. A multiple access link is
shared among many senders and receivers; consequently, the link-layer protocol for
a multiple access channel has a protocol (its multiple access protocol) for coordinat-
ing link access. In the case of MPLS, the link connecting two adjacent nodes (for
500 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANS
www.google.com
www.google.com
example, two IP routers that are adjacent in an IP sensethat they are next-hop IP
routers toward some destination) may actually be a network in and of itself. In one
sense, the idea of a network being considered as a link should not seem odd. A tele-
phone link connecting a home modem/computer to a remote modem/router, for
example, is actually a path through a sophisticated and complex telephone network.
Among the principles underlying link-layer communication, we examined error-
detection and -correction techniques, multiple access protocols, link-layer address-
ing, virtualization (VLANs), and the construction of extended switched LANs and
data center networks.Much of the focus today at the link layer is on these switched
networks. In the case of error detection/correction, we examined how it is possible to
add additional bits to a frames header in order to detect, and in some cases correct,
bit-flip errors that might occur when the frame is transmitted over the link. We cov-
ered simple parity and checksumming schemes, as well as the more robust cyclic
redundancy check. We then moved on to the topic of multiple access protocols. We
identified and studied three broad approaches for coordinating access to a broadcast
channel: channel partitioning approaches (TDM, FDM), random access approaches
(the ALOHA protocols and CSMA protocols), and taking-turns approaches (polling
and token passing). We studied the cable access network and found that it uses many
of these multiple access methods. We saw that a consequence of having multiple
nodes share a single broadcast channel was the need to provide node addresses at the
link layer. We learned that link-layer addresses were quite different from network-
layer addresses and that, in the case of the Internet, a special protocol (ARPthe
Address Resolution Protocol) is used to translate between these two forms of
addressing and studied the hugely successful Ethernet protocol in detail. We then
examined how nodes sharing a broadcast channel form a LAN and how multiple
LANs can be connected together to form larger LANsall without the intervention
of network-layer routing to interconnect these local nodes. We also learned how mul-
tiple virtual LANs can be created on a single physical LAN infrastructure.
We ended our study of the link layer by focusing on how MPLS networks pro-
vide link-layer services when they interconnect IP routers and an overview of the net-
work designs for todays massive data centers.We wrapped up this chapter (and
indeed the first five chapters) by identifying the many protocols that are needed to
fetch a simple Web page. Having covered the link layer, our journey down the proto-
col stack is now over! Certainly, the physical layer lies below the link layer, but the
details of the physical layer are probably best left for another course (for example, in
communication theory, rather than computer networking). We have, however,
touched upon several aspects of the physical layer in this chapter and in Chapter 1
(our discussion of physical media in Section 1.2). Well consider the physical layer
again when we study wireless link characteristics in the next chapter.
Although our journey down the protocol stack is over, our study of computer
networking is not yet at an end. In the following four chapters we cover wireless
networking, multimedia networking, network security, and network management.
These four topics do not fit conveniently into any one layer; indeed, each topic
crosscuts many layers. Understanding these topics (billed as advanced topics in
5.8 SUMMARY 501
some networking texts) thus requires a firm foundation in all layers of the protocol
stacka foundation that our study of the link layer has now completed!
Homework Problems and Questions
Chapter 5 Review Questions
SECTIONS 5.15.2
R1. Consider the transportation analogy in Section 5.1.1. If the passenger is
analagous to a datagram, what is analogous to the link layer frame?
R2. If all the links in the Internet were to provide reliable delivery service, would
the TCP reliable delivery service be redundant? Why or why not?
R3. What are some of the possible services that a link-layer protocol can offer to
the network layer? Which of these link-layer services have corresponding
services in IP? In TCP?
SECTION 5.3
R4. Suppose two nodes start to transmit at the same time a packet of length L
over a broadcast channel of rate R. Denote the propagation delay between the
two nodes as dprop. Will there be a collision if dprop < L/R? Why or why not?R5. In Section 5.3, we listed four desirable characteristics of a broadcast channel.Which of these characteristics does slotted ALOHA have? Which of thesecharacteristics does token passing have?R6. In CSMA/CD, after the fifth collision, what is the probability that a nodechooses K = 4? The result K = 4 corresponds to a delay of how many secondson a 10 Mbps Ethernet?R7. Describe polling and token-passing protocols using the analogy of cocktailparty interactions.R8. Why would the token-ring protocol be inefficient if a LAN had a very largeperimeter?SECTION 5.4R9. How big is the MAC address space? The IPv4 address space? The IPv6address space?R10. Suppose nodes A, B, and C each attach to the same broadcast LAN (throughtheir adapters). If A sends thousands of IP datagrams to B with each encapsu-lating frame addressed to the MAC address of B, will Cs adapter processthese frames? If so, will Cs adapter pass the IP datagrams in these frames tothe network layer C? How would your answers change if A sends frames withthe MAC broadcast address?502 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANSR11. Why is an ARP query sent within a broadcast frame? Why is an ARPresponse sent within a frame with a specific destination MAC address?R12. For the network in Figure 5.19, the router has two ARP modules, each withits own ARP table. Is it possible that the same MAC address appears in bothtables?R13. Compare the frame structures for 10BASE-T, 100BASE-T, and Gigabit Eth-ernet. How do they differ?R14. Consider Figure 5.15. How many subnetworks are there, in the addressingsense of Section 4.4?R15. What is the maximum number of VLANs that can be configured on a switchsupporting the 802.1Q protocol? Why?R16. Suppose that N switches supporting K VLAN groups are to be connected viaa trunking protocol. How many ports are needed to connect the switches?Justify your answer.ProblemsP1. Suppose the information content of a packet is the bit pattern 1110 0110 10011101 and an even parity scheme is being used. What would the value of thefield containing the parity bits be for the case of a two-dimensional parityscheme? Your answer should be such that a minimum-length checksum fieldis used.P2. Show (give an example other than the one in Figure 5.5) that two-dimensionalparity checks can correct and detect a single bit error. Show (give an exampleof) a double-bit error that can be detected but not corrected.P3. Suppose the information portion of a packet (D in Figure 5.3) contains 10bytes consisting of the 8-bit unsigned binary ASCII representation of stringNetworking. Compute the Internet checksum for this data.P4. Consider the previous problem, but instead suppose these 10 bytes containa. the binary representation of the numbers 1 through 10.b. the ASCII representation of the letters B through K (uppercase).c. the ASCII representation of the letters b through k (lowercase).Compute the Internet checksum for this data.P5. Consider the 7-bit generator, G=10011, and suppose that D has the value1010101010. What is the value of R?P6. Consider the previous problem, but suppose that D has the valuea. 1001010101.b. 0101101010.c. 1010100000.PROBLEMS 503P7. In this problem, we explore some of the properties of the CRC. For the gen-erator G (=1001) given in Section 5.2.3, answer the following questions.a. Why can it detect any single bit error in data D?b. Can the above G detect any odd number of bit errors? Why?P8. In Section 5.3, we provided an outline of the derivation of the efficiency ofslotted ALOHA. In this problem well complete the derivation.a. Recall that when there are N active nodes, the efficiency of slotted ALOHAis Np(1 p)N1. Find the value of p that maximizes this expression.b. Using the value of p found in (a), find the efficiency of slotted ALOHA byletting N approach infinity. Hint: (1 1/N)N approaches 1/e as Napproaches infinity.P9. Show that the maximum efficiency of pure ALOHA is 1/(2e). Note: Thisproblem is easy if you have completed the problem above!P10. Consider two nodes, A and B, that use the slotted ALOHA protocol to con-tend for a channel. Suppose node A has more data to transmit than node B,and node As retransmission probability pA is greater than node Bs retrans-mission probability, pB.a. Provide a formula for node As average throughput. What is the total effi-ciency of the protocol with these two nodes? b. If pA = 2pB, is node As average throughput twice as large as that of node B?Why or why not? If not, how can you choose pA and pB to make that happen?c. In general, suppose there are N nodes, among which node A has retrans-mission probability 2p and all other nodes have retransmission probabilityp. Provide expressions to compute the average throughputs of node A andof any other node. P11. Suppose four active nodesnodes A, B, C and Dare competing for access toa channel using slotted ALOHA. Assume each node has an infinite numberof packets to send. Each node attempts to transmit in each slot with probabilityp. The first slot is numbered slot 1, the second slot is numbered slot 2, and so on.a. What is the probability that node A succeeds for the first time in slot 5?b. What is the probability that some node (either A, B, C or D) succeeds in slot 4?c. What is the probability that the first success occurs in slot 3?d. What is the efficiency of this four-node system?P12. Graph the efficiency of slotted ALOHA and pure ALOHA as a function of pfor the following values of N:a. N15.b. N25.c. N35.504 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANSSubnet 3EFCSubnet 2DABSubnet 1Figure 5.33 Three subnets, interconnected by routersP13. Consider a broadcast channel with N nodes and a transmission rate of R bps.Suppose the broadcast channel uses polling (with an additional polling node)for multiple access. Suppose the amount of time from when a node completestransmission until the subsequent node is permitted to transmit (that is, thepolling delay) is dpoll. Suppose that within a polling round, a given node isallowed to transmit at most Q bits. What is the maximum throughput of thebroadcast channel?P14. Consider three LANs interconnected by two routers, as shown in Figure 5.33.a. Assign IP addresses to all of the interfaces. For Subnet 1 use addresses ofthe form 192.168.1.xxx; for Subnet 2 uses addresses of the form192.168.2.xxx; and for Subnet 3 use addresses of the form 192.168.3.xxx.b. Assign MAC addresses to all of the adapters.c. Consider sending an IP datagram from Host E to Host B. Suppose all ofthe ARP tables are up to date. Enumerate all the steps, as done for the single-router example in Section 5.4.1.d. Repeat (c), now assuming that the ARP table in the sending host is empty(and the other tables are up to date).P15. Consider Figure 5.33. Now we replace the router between subnets 1 and 2with a switch S1, and label the router between subnets 2 and 3 as R1. PROBLEMS 505a. Consider sending an IP datagram from Host E to Host F. Will Host E askrouter R1 to help forward the datagram? Why? In the Ethernet frame con-taining the IP datagram, what are the source and destination IP and MACaddresses?b. Suppose E would like to send an IP datagram to B, and assume that EsARP cache does not contain Bs MAC address. Will E perform an ARPquery to find Bs MAC address? Why? In the Ethernet frame (containingthe IP datagram destined to B) that is delivered to router R1, what are thesource and destination IP and MAC addresses? c. Suppose Host A would like to send an IP datagram to Host B, and neither AsARP cache contains Bs MAC address nor does Bs ARP cache contain AsMAC address. Further suppose that the switch S1s forwarding table containsentries for Host B and router R1 only. Thus, A will broadcast an ARP requestmessage. What actions will switch S1 perform once it receives the ARPrequest message? Will router R1 also receive this ARP request message? If so,will R1 forward the message to Subnet 3? Once Host B receives this ARPrequest message, it will send back to Host A an ARP response message. Butwill it send an ARP query message to ask for As MAC address? Why? Whatwill switch S1 do once it receives an ARP response message from Host B?P16. Consider the previous problem, but suppose now that the router between sub-nets 2 and 3 is replaced by a switch. Answer questions (a)(c) in the previousproblem in this new context.P17. Recall that with the CSMA/CD protocol, the adapter waits K 512 bit times aftera collision, where K is drawn randomly. For K = 100, how long does the adapterwait until returning to Step 2 for a 10 Mbps broadcast channel? For a 100 Mbpsbroadcast channel?P18. Suppose nodes A and B are on the same 10 Mbps broadcast channel, and thepropagation delay between the two nodes is 325 bit times. Suppose CSMA/CDand Ethernet packets are used for this broadcast channel. Suppose node A beginstransmitting a frame and, before it finishes, node B begins transmitting a frame.Can A finish transmitting before it detects that B has transmitted? Why or whynot? If the answer is yes, then A incorrectly believes that its frame was success-fully transmitted without a collision. Hint: Suppose at time t = 0 bits, A beginstransmitting a frame. In the worst case, A transmits a minimum-sized frame of512 + 64 bit times. So A would finish transmitting the frame at t = 512 + 64 bittimes. Thus, the answer is no, if Bs signal reaches A before bit time t = 512 + 64bits. In the worst case, when does Bs signal reach A?P19. Suppose nodes A and B are on the same 10 Mbps broadcast channel, and the propagation delay between the two nodes is 245 bit times. Suppose Aand B send Ethernet frames at the same time, the frames collide, and then A and B choose different values of K in the CSMA/CD algorithm. Assuming506 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANSVideoNoteSending a datagrambetween subnets: link-layer and network-layeraddressingno other nodes are active, can the retransmissions from A and B collide? Forour purposes, it suffices to work out the following example. Suppose A and Bbegin transmission at t = 0 bit times. They both detect collisions at t = 245 bittimes. Suppose KA = 0 and KB = 1. At what time does B schedule its retrans-mission? At what time does A begin transmission? (Note: The nodes must waitfor an idle channel after returning to Step 2see protocol.) At what time doesAs signal reach B? Does B refrain from transmitting at its scheduled time?P20. In this problem, you will derive the efficiency of a CSMA/CD-like multiple accessprotocol. In this protocol, time is slotted and all adapters are synchronized to theslots. Unlike slotted ALOHA, however, the length of a slot (in seconds) is muchless than a frame time (the time to transmit a frame). Let S be the length of a slot.Suppose all frames are of constant length L = kRS, where R is the transmission rateof the channel and k is a large integer. Suppose there are N nodes, each with aninfinite number of frames to send. We also assume that dprop < S, so that all nodescan detect a collision before the end of a slot time. The protocol is as follows: If, for a given slot, no node has possession of the channel, all nodes contend for the channel; in particular, each node transmits in the slot withprobability p. If exactly one node transmits in the slot, that node takespossession of the channel for the subsequent k 1 slots and transmits itsentire frame. If some node has possession of the channel, all other nodes refrain fromtransmitting until the node that possesses the channel has finished trans-mitting its frame. Once this node has transmitted its frame, all nodes contend for the channel.Note that the channel alternates between two states: the productive state,which lasts exactly k slots, and the nonproductive state, which lasts for a random number of slots. Clearly, the channel efficiency is the ratio of k/(k + x),where x is the expected number of consecutive unproductive slots.a. For fixed N and p, determine the efficiency of this protocol.b. For fixed N, determine the p that maximizes the efficiency.c. Using the p (which is a function of N) found in (b), determine the effi-ciency as N approaches infinity.d. Show that this efficiency approaches 1 as the frame length becomeslarge.P21. Consider Figure 5.33 in problem P14. Provide MAC addresses and IPaddresses for the interfaces at Host A, both routers, and Host F. Suppose HostA sends a datagram to Host F. Give the source and destination MAC addressesin the frame encapsulating this IP datagram as the frame is transmitted (i) fromA to the left router, (ii) from the left router to the right router, (iii) from theright router to F. Also give the source and destination IP addresses in the IPdatagram encapsulated within the frame at each of these points in time.PROBLEMS 507P22. Suppose now that the leftmost router in Figure 5.33 is replaced by a switch.Hosts A, B, C, and D and the right router are all star-connected into thisswitch. Give the source and destination MAC addresses in the frame encap-sulating this IP datagram as the frame is transmitted (i) from A to the switch,(ii) from the switch to the right router, (iii) from the right router to F. Alsogive the source and destination IP addresses in the IP datagram encapsulatedwithin the frame at each of these points in time.P23. Consider Figure 5.15. Suppose that all links are 100 Mbps. What is the maxi-mum total aggregate throughput that can be achieved among the 9 hosts and2 servers in this network? You can assume that any host or serrver can sendto any other host or server. Why?P24. Suppose the three departmental switches in Figure 5.15 are replaced byhubs. All links are 100 Mbps. Now answer the questions posed in problemP23.P25. Suppose that all the switches in Figure 5.15 are replaced by hubs. All linksare 100 Mbps. Now answer the questions posed in problem P23.P26. Lets consider the operation of a learning switch in the context of a net-work in which 6 nodes labeled A through F are star connected into an Eth-ernet switch. Suppose that (i) B sends a frame to E, (ii) E replies with aframe to B, (iii) A sends a frame to B, (iv) B replies with a frame to A. Theswitch table is initially empty. Show the state of the switch table beforeand after each of these events. For each of these events, identify the link(s)on which the transmitted frame will be forwarded, and briefly justify youranswers.P27. In this problem, we explore the use of small packets for Voice-over-IP appli-cations. One of the drawbacks of a small packet size is that a large fraction oflink bandwidth is consumed by overhead bytes. To this end, suppose that thepacket consists of P bytes and 5 bytes of header.a. Consider sending a digitally encoded voice source directly. Suppose thesource is encoded at a constant rate of 128 kbps. Assume each packet isentirely filled before the source sends the packet into the network. Thetime required to fill a packet is the packetization delay. In terms of L,determine the packetization delay in milliseconds.b. Packetization delays greater than 20 msec can cause a noticeable andunpleasant echo. Determine the packetization delay for L = 1,500 bytes(roughly corresponding to a maximum-sized Ethernet packet) and for L = 50 (corresponding to an ATM packet).c. Calculate the store-and-forward delay at a single switch for a link rate ofR = 622 Mbps for L = 1,500 bytes, and for L = 50 bytes.d. Comment on the advantages of using a small packet size.508 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANSP28. Consider the single switch VLAN in Figure 5.25, and assume an externalrouter is connected to switch port 1. Assign IP addresses to the EE and CShosts and router interface. Trace the steps taken at both the network layer andthe link layer to transfer an IP datagram from an EE host to a CS host (Hint:reread the discussion of Figure 5.19 in the text).P29. Consider the MPLS network shown in Figure 5.29, and suppose that routersR5 and R6 are now MPLS enabled. Suppose that we want to perform trafficengineering so that packets from R6 destined for A are switched to A via R6-R4-R3-R1, and packets from R5 destined for A are switched via R5-R4-R2-R1. Show the MPLS tables in R5 and R6, as well as the modifiedtable in R4, that would make this possible.P30. Consider again the same scenario as in the previous problem, but supposethat packets from R6 destined for D are switched via R6-R4-R3, while pack-ets from R5 destined to D are switched via R4-R2-R1-R3. Show the MPLStables in all routers that would make this possible.P31. In this problem, you will put together much of what you have learned aboutInternet protocols. Suppose you walk into a room, connect to Ethernet, andwant to download a Web page. What are all the protocol steps that take place,starting from powering on your PC to getting the Web page? Assume there isnothing in our DNS or browser caches when you power on your PC. (Hint:the steps include the use of Ethernet, DHCP, ARP, DNS, TCP, and HTTP pro-tocols.) Explicitly indicate in your steps how you obtain the IP and MACaddresses of a gateway router.P32. Consider the data center network with hierarchical topology in Figure 5.30.Suppose now there are 80 pairs of flows, with ten flows between the first andninth rack, ten flows between the second and tenth rack, and so on. Furthersuppose that all links in the network are 10 Gbps, except for the linksbetween hosts and TOR switches, which are 1 Gbps.a. Each flow has the same data rate; determine the maximum rate of a flow.b. For the same traffic pattern, determine the maximum rate of a flow for thehighly interconnected topology in Figure 5.31.c. Now suppose there is a similar traffic pattern, but involving 20 hosts oneach hosts and 160 pairs of flows. Determine the maximum flow rates forthe two topologies.P33. Consider the hierarchical network in Figure 5.30 and suppose that the datacenter needs to support email and video distribution among other applica-tions. Suppose four racks of servers are reserved for email and four racks arereserved for video. For each of the applications, all four racks must lie belowa single tier-2 switch since the tier-2 to tier-1 links do not have sufficientbandwidth to support the intra-application traffic. For the email application, PROBLEMS 509510 CHAPTER 5 THE LINK LAYER: LINKS, ACCESS NETWORKS, AND LANSsuppose that for 99.9 percent of the time only three racks are used, and thatthe video application has identical usage patterns.a. For what fraction of time does the email application need to use a fourthrack? How about for the video application?b. Assuming email usage and video usage are independent, for what fractionof time do (equivalently, what is the probability that) both applicationsneed their fourth rack?c. Suppose that it is acceptable for an application to have a shortage ofservers for 0.001 percent of time or less (causing rare periods of perform-ance degradation for users).Discuss how the topology in Figure 5.31 can be used so that only sevenracks are collectively assigned to the two applications (assuming that thetopology can support all the traffic). Wireshark LabsAt the companion Web site for this textbook, http://www.awl.com/kurose-ross,youll find a Wireshark lab that examines the operation of the IEEE 802.3 protocoland the Wireshark frame format. A second Wireshark lab examines packet tracestaken in a home network scenario.http://www.awl.com/kurose-rossSimon S. LamAN INTERVIEW WITH…Simon S. Lam is Professor and Regents Chair in Computer Sciencesat the University of Texas at Austin. From 1971 to 1974, he waswith the ARPA Network Measurement Center at UCLA, where heworked on satellite and radio packet switching. He led a researchgroup that invented secure sockets and prototyped, in 1993, the firstsecure sockets layer named Secure Network Programming, whichwon the 2004 ACM Software System Award. His research interestsare in design and analysis of network protocols and security servic-es. He received his BSEE from Washington State University and hisMS and PhD from UCLA. He was elected to the National Academyof Engineering in 2007.511Why did you decide to specialize in networking?When I arrived at UCLA as a new graduate student in Fall 1969, my intention was to studycontrol theory. Then I took the queuing theory classes of Leonard Kleinrock and was veryimpressed by him. For a while, I was working on adaptive control of queuing systems as a pos-sible thesis topic. In early 1972, Larry Roberts initiated the ARPAnet Satellite System project(later called Packet Satellite). Professor Kleinrock asked me to join the project. The first thingwe did was to introduce a simple, yet realistic, backoff algorithm to the slotted ALOHA proto-col. Shortly thereafter, I found many interesting research problems, such as ALOHAs instab-ility problem and need for adaptive backoff, which would form the core of my thesis.You were active in the early days of the Internet in the 1970s, beginning with your student days at UCLA. What was it like then? Did people have any inkling of what the Internet would become?The atmosphere was really no different from other system-building projects I have seen inindustry and academia. The initially stated goal of the ARPAnet was fairly modest, that is,to provide access to expensive computers from remote locations so that many more scien-tists could use them. However, with the startup of the Packet Satellite project in 1972 andthe Packet Radio project in 1973, ARPAs goal had expanded substantially. By 1973, ARPAwas building three different packet networks at the same time, and it became necessary forVint Cerf and Bob Kahn to develop an interconnection strategy.Back then, all of these progressive developments in networking were viewed (I believe)as logical rather than magical. No one could have envisioned the scale of the Internet andpower of personal computers today. It was a decade before appearance of the first PCs. To putthings in perspective, most students submitted their computer programs as decks of punchedcards for batch processing. Only some students had direct access to computers, which weretypically housed in a restricted area. Modems were slow and still a rarity. As a graduate stu-dent, I had only a phone on my desk, and I used pencil and paper to do most of my work.512Where do you see the field of networking and the Internet heading in the future?In the past, the simplicity of the Internets IP protocol was its greatest strength in vanquish-ing competition and becoming the de facto standard for internetworking. Unlike competi-tors, such as X.25 in the 1980s and ATM in the 1990s, IP can run on top of any link-layernetworking technology, because it offers only a best-effort datagram service. Thus, any packet network can connect to the Internet.Today, IPs greatest strength is actually a shortcoming. IP is like a straitjacket that con-fines the Internets development to specific directions. In recent years, many researchershave redirected their efforts to the application layer only. There is also a great deal ofresearch on wireless ad hoc networks, sensor networks, and satellite networks. These net-works can be viewed either as stand-alone systems or link-layer systems, which can flourishbecause they are outside of the IP straitjacket.Many people are excited about the possibility of P2P systems as a platform for novelInternet applications. However, P2P systems are highly inefficient in their use of Internetresources. A concern of mine is whether the transmission and switching capacity of theInternet core will continue to increase faster than the traffic demand on the Internet as itgrows to interconnect all kinds of devices and support future P2P-enabled applications.Without substantial overprovisioning of capacity, ensuring network stability in the presenceof malicious attacks and congestion will continue to be a significant challenge.The Internets phenomenal growth also requires the allocation of new IP addresses at arapid rate to network operators and enterprises worldwide. At the current rate, the pool of unal-located IPv4 addresses would be depleted in a few years. When that happens, large contiguousblocks of address space can only be allocated from the IPv6 address space. Since adoption ofIPv6 is off to a slow start, due to lack of incentives for early adopters, IPv4 and IPv6 will mostlikely co-exist on the Internet for many years to come. Successful migration from an IPv4-dom-inant Internet to an IPv6-dominant Internet will require a substantial global effort.What is the most challenging part of your job?The most challenging part of my job as a professor is teaching and motivating every studentin my class, and every doctoral student under my supervision, rather than just the highachievers. The very bright and motivated may require a little guidance but not much else. Ioften learn more from these students than they learn from me. Educating and motivating theunderachievers present a major challenge.What impacts do you foresee technology having on learning in the future?Eventually, almost all human knowledge will be accessible through the Internet, which willbe the most powerful tool for learning. This vast knowledge base will have the potential ofleveling the playing field for students all over the world. For example, motivated students inany country will be able to access the best-class Web sites, multimedia lectures, and teach-ing materials. Already, it was said that the IEEE and ACM digital libraries have acceleratedthe development of computer science researchers in China. In time, the Internet will tran-scend all geographic barriers to learning.CHAPTER 6Wireless andMobileNetworks513In the telephony world, the past 15 years have arguably been the golden years of cel-lular telephony. The number of worldwide mobile cellular subscribers increasedfrom 34 million in 1993 to nearly 5.5 billion subscribers by 2011, with the numberof cellular subscribers now surpassing the number of wired telephone lines. Themany advantages of cell phones are evident to allanywhere, anytime, untetheredaccess to the global telephone network via a highly portable lightweight device.With the advent of laptops, palmtops, smartphones, and their promise of anywhere,anytime, untethered access to the global Internet, is a similar explosion in the use ofwireless Internet devices just around the corner?Regardless of the future growth of wireless Internet devices, its already clearthat wireless networks and the mobility-related services they enable are here to stay.From a networking standpoint, the challenges posed by these networks, particularlyat the link layer and the network layer, are so different from traditional wired com-puter networks that an individual chapter devoted to the study of wireless andmobile networks (i.e., this chapter) is appropriate.Well begin this chapter with a discussion of mobile users, wireless links, andnetworks, and their relationship to the larger (typically wired) networks to whichthey connect. Well draw a distinction between the challenges posed by the wirelessnature of the communication links in such networks, and by the mobility that thesewireless links enable. Making this important distinctionbetween wireless and514 CHAPTER 6 WIRELESS AND MOBILE NETWORKSmobilitywill allow us to better isolate, identify, and master the key concepts ineach area. Note that there are indeed many networked environments in which thenetwork nodes are wireless but not mobile (e.g., wireless home or office networkswith stationary workstations and large displays), and that there are limited forms ofmobility that do not require wireless links (e.g., a worker who uses a wired laptop athome, shuts down the laptop, drives to work, and attaches the laptop to the com-panys wired network). Of course, many of the most exciting networked environ-ments are those in which users are both wireless and mobilefor example, ascenario in which a mobile user (say in the back seat of car) maintains a Voice-over-IPcall and multiple ongoing TCP connections while racing down the autobahn at 160 kilometers per hour. It is here, at the intersection of wireless and mobility, thatwell find the most interesting technical challenges!Well begin by illustrating the setting in which well consider wireless commu-nication and mobilitya network in which wireless (and possibly mobile) users areconnected into the larger network infrastructure by a wireless link at the networksedge. Well then consider the characteristics of this wireless link in Section 6.2. Weinclude a brief introduction to code division multiple access (CDMA), a shared-medium access protocol that is often used in wireless networks, in Section 6.2. InSection 6.3, well examine the link-level aspects of the IEEE 802.11 (WiFi) wire-less LAN standard in some depth; well also say a few words about Bluetooth andother wireless personal area networks. In Section 6.4, well provide an overview ofcellular Internet access, including 3G and emerging 4G cellular technologies thatprovide both voice and high-speed Internet access. In Section 6.5, well turn ourattention to mobility, focusing on the problems of locating a mobile user, routing tothe mobile user, and handing off the mobile user who dynamically moves fromone point of attachment to the network to another. Well examine how these mobil-ity services are implemented in the mobile IP standard and in GSM, in Sections 6.6and 6.7, respectively. Finally, well consider the impact of wireless links and mobil-ity on transport-layer protocols and networked applications in Section 6.8.6.1 IntroductionFigure 6.1 shows the setting in which well consider the topics of wireless data com-munication and mobility. Well begin by keeping our discussion general enough tocover a wide range of networks, including both wireless LANs such as IEEE 802.11and cellular networks such as a 3G network; well drill down into a more detaileddiscussion of specific wireless architectures in later sections. We can identify thefollowing elements in a wireless network: Wireless hosts. As in the case of wired networks, hosts are the end-system devicesthat run applications. A wireless host might be a laptop, palmtop, smartphone, ordesktop computer. The hosts themselves may or may not be mobile.6.1 INTRODUCTION 515 Wireless links. A host connects to a base station (defined below) or to anotherwireless host through a wireless communication link. Different wireless linktechnologies have different transmission rates and can transmit over differentdistances. Figure 6.2 shows two key characteristics (coverage area and link rate)of the more popular wireless network standards. (The figure is only meant to pro-vide a rough idea of these characteristics. For example, some of these types ofnetworks are only now being deployed, and some link rates can increase ordecrease beyond the values shown depending on distance, channel conditions,and the number of users in the wireless network.) Well cover these standardsPUBLIC WIFIACCESS: COMING SOON TO A LAMP POST NEAR YOU?WiFi hotspotspublic locations where users can find 802.11 wireless accessarebecoming increasingly common in hotels, airports, and cafs around the world. Mostcollege campuses offer ubiquitous wireless access, and its hard to find a hotel thatdoesnt offer wireless Internet access.Over the past decade a number of cities have designed, deployed, and operatedmunicipal WiFi networks. The vision of providing ubiquitous WiFi access to the commu-nity as a public service (much like streetlights)helping to bridge the digital divide byproviding Internet access to all citizens and to promote economic developmentiscompelling. Many cities around the world, including Philadelphia, Toronto, HongKong, Minneapolis, London, and Auckland, have plans to provide ubiquitous wirelesswithin the city, or have already done so to varying degrees. The goal in Philadelphiawas to turn Philadelphia into the nations largest WiFi hotspot and help to improveeducation, bridge the digital divide, enhance neighborhood development, and reducethe costs of government. The ambitious programan agreement between the city,Wireless Philadelphia (a nonprofit entity), and the Internet Service Provider Earthlinkbuilt an operational network of 802.11b hotspots on streetlamp pole arms and trafficcontrol devices that covered 80 percent of the city. But financial and operational con-cerns caused the network to be sold to a group of private investors in 2008, who latersold the network back to the city in 2010. Other cities, such as Minneapolis, Toronto,Hong Kong, and Auckland, have had success with smaller-scale efforts.The fact that 802.11 networks operate in the unlicensed spectrum (and hence canbe deployed without purchasing expensive spectrum use rights) would seem to makethem financially attractive. However, 802.11 access points (see Section 6.3) havemuch shorter ranges than 3G cellular base stations (see Section 6.4), requiring a larg-er number of deployed endpoints to cover the same geographic region. Cellular datanetworks providing Internet access, on the other hand, operate in the licensed spec-trum. Cellular providers pay billions of dollars for spectrum access rights for their net-works, making cellular data networks a business rather than municipal undertaking.CASE HISTORYlater in the first half of this chapter; well also consider other wireless link char-acteristics (such as their bit error rates and the causes of bit errors) in Section 6.2.In Figure 6.1, wireless links connect wireless hosts located at the edge of the net-work into the larger network infrastructure. We hasten to add that wireless linksare also sometimes used within a network to connect routers, switches, and othernetwork equipment. However, our focus in this chapter will be on the use ofwireless communication at the network edge, as it is here that many of the mostexciting technical challenges, and most of the growth, are occurring. Base station. The base station is a key part of the wireless network infrastructure.Unlike the wireless host and wireless link, a base station has no obvious counterpartin a wired network. A base station is responsible for sending and receiving data (e.g.,packets) to and from a wireless host that is associated with that base station. A basestation will often be responsible for coordinating the transmission of multiple wire-less hosts with which it is associated. When we say a wireless host is associatedwith a base station, we mean that (1) the host is within the wireless communication516 CHAPTER 6 WIRELESS AND MOBILE NETWORKSNetworkinfrastructureKey:Wireless access pointCoverage areaWireless hostWireless host in motionFigure 6.1 Elements of a wireless networkdistance of the base station, and (2) the host uses that base station to relay databetween it (the host) and the larger network. Cell towers in cellular networks andaccess points in 802.11 wireless LANs are examples of base stations.In Figure 6.1, the base station is connected to the larger network (e.g., the Inter-net, corporate or home network, or telephone network), thus functioning as alink-layer relay between the wireless host and the rest of the world with whichthe host communicates.Hosts associated with a base station are often referred to as operating ininfrastructure mode, since all traditional network services (e.g., address assign-ment and routing) are provided by the network to which a host is connected viathe base station. In ad hoc networks, wireless hosts have no such infrastructurewith which to connect. In the absence of such infrastructure, the hosts themselvesmust provide for services such as routing, address assignment, DNS-like nametranslation, and more.When a mobile host moves beyond the range of one base station and into therange of another, it will change its point of attachment into the larger network(i.e., change the base station with which it is associated)a process referred to ashandoff. Such mobility raises many challenging questions. If a host can move,how does one find the mobile hosts current location in the network so that datacan be forwarded to that mobile host? How is addressing performed, given that ahost can be in one of many possible locations? If the host moves during a TCP6.1 INTRODUCTION 517802.11a,g802.11n802.11b802.15.13G: UMTS/WCDMA, CDMA20002G: IS-95, CDMA, GSMIndoor Outdoor Mid rangeoutdoorLong rangeoutdoor1030m 50200m 200m4Km 5Km20Km54 Mbps4 Mbps511 Mbps200 Mbps1 Mbps384 KbpsEnhanced 3G: HSPA4G: LTE802.11a,g point-to-pointFigure 6.2 Link characteristics of selected wireless network standardsconnection or phone call, how is data routed so that the connection continuesuninterrupted? These and many (many!) other questions make wireless andmobile networking an area of exciting networking research. Network infrastructure. This is the larger network with which a wireless hostmay wish to communicate.Having discussed the pieces of a wireless network, we note that thesepieces can be combined in many different ways to form different types of wirelessnetworks. You may find a taxonomy of these types of wireless networks useful asyou read on in this chapter, or read/learn more about wireless networks beyondthis book. At the highest level we can classify wireless networks according to twocriteria: (i) whether a packet in the wireless network crosses exactly one wirelesshop or multiple wireless hops, and (ii) whether there is infrastructure such as a basestation in the network: Single-hop, infrastructure-based. These networks have a base station that isconnected to a larger wired network (e.g., the Internet). Furthermore, all com-munication is between this base station and a wireless host over a single wire-less hop. The 802.11 networks you use in the classroom, caf, or library; andthe 3G cellular data networks that we will learn about shortly all fall in thiscategory. Single-hop, infrastructure-less. In these networks, there is no base station that isconnected to a wireless network. However, as we will see, one of the nodes inthis single-hop network may coordinate the transmissions of the other nodes.Bluetooth networks (which we will study in Section 6.3.6) and 802.11 networksin ad hoc mode are single-hop, infrastructure-less networks. Multi-hop, infrastructure-based. In these networks, a base station is present thatis wired to the larger network. However, some wireless nodes may have to relaytheir communication through other wireless nodes in order to communicate viathe base station. Some wireless sensor networks and so-called wireless meshnetworks fall in this category. Multi-hop, infrastructure-less. There is no base station in these networks, andnodes may have to relay messages among several other nodes in order to reach adestination. Nodes may also be mobile, with connectivity changing amongnodesa class of networks known as mobile ad hoc networks (MANETs). Ifthe mobile nodes are vehicles, the network is a vehicular ad hoc network(VANET). As you might imagine, the development of protocols for such net-works is challenging and is the subject of much ongoing research.In this chapter, well mostly confine ourselves to single-hop networks, and thenmostly to infrastructure-based networks.518 CHAPTER 6 WIRELESS AND MOBILE NETWORKSLets now dig deeper into the technical challenges that arise in wireless andmobile networks. Well begin by first considering the individual wireless link, defer-ring our discussion of mobility until later in this chapter.6.2 Wireless Links and Network CharacteristicsLets begin by considering a simple wired network, say a home network, with awired Ethernet switch (see Section 5.4) interconnecting the hosts. If we replacethe wired Ethernet with a wireless 802.11 network, a wireless network interfacewould replace the hosts wired Ethernet interface, and an access point wouldreplace the Ethernet switch, but virtually no changes would be needed at the net-work layer or above. This suggests that we focus our attention on the link layerwhen looking for important differences between wired and wireless networks.Indeed, we can find a number of important differences between a wired link and awireless link: Decreasing signal strength. Electromagnetic radiation attenuates as it passesthrough matter (e.g., a radio signal passing through a wall). Even in free space,the signal will disperse, resulting in decreased signal strength (sometimesreferred to as path loss) as the distance between sender and receiver increases. Interference from other sources. Radio sources transmitting in the same fre-quency band will interfere with each other. For example, 2.4 GHz wirelessphones and 802.11b wireless LANs transmit in the same frequency band. Thus,the 802.11b wireless LAN user talking on a 2.4 GHz wireless phone can expectthat neither the network nor the phone will perform particularly well. In additionto interference from transmitting sources, electromagnetic noise within the envi-ronment (e.g., a nearby motor, a microwave) can result in interference. Multipath propagation. Multipath propagation occurs when portions of theelectromagnetic wave reflect off objects and the ground, taking paths of differentlengths between a sender and receiver. This results in the blurring of the receivedsignal at the receiver. Moving objects between the sender and receiver can causemultipath propagation to change over time.For a detailed discussion of wireless channel characteristics, models, and measure-ments, see [Anderson 1995].The discussion above suggests that bit errors will be more common in wirelesslinks than in wired links. For this reason, it is perhaps not surprising that wirelesslink protocols (such as the 802.11 protocol well examine in the following section)employ not only powerful CRC error detection codes, but also link-level reliable-data-transfer protocols that retransmit corrupted frames.6.2 WIRELESS LINKS AND NETWORK CHARACTERISTICS 519520 CHAPTER 6 WIRELESS AND MOBILE NETWORKSFigure 6.3 Bit error rate, transmission rate, and SNR10710610510410310210110 20 30 400SNR (dB)BERQAM16(4 Mbps)QAM256(8 Mbps)BPSK(1Mpbs)Having considered the impairments that can occur on a wireless channel, letsnext turn our attention to the host receiving the wireless signal. This host receives anelectromagnetic signal that is a combination of a degraded form of the original signaltransmitted by the sender (degraded due to the attenuation and multipath propagationeffects that we discussed above, among others) and background noise in the environ-ment. The signal-to-noise ratio (SNR) is a relative measure of the strength of thereceived signal (i.e., the information being transmitted) and this noise. The SNR istypically measured in units of decibels (dB), a unit of measure that some think is usedby electrical engineers primarily to confuse computer scientists. The SNR, measuredin dB, is twenty times the ratio of the base-10 logarithm of the amplitude of thereceived signal to the amplitude of the noise. For our purposes here, we need onlyknow that a larger SNR makes it easier for the receiver to extract the transmitted sig-nal from the background noise.Figure 6.3 (adapted from [Holland 2001]) shows the bit error rate (BER)roughly speaking, the probability that a transmitted bit is received in error at thereceiverversus the SNR for three different modulation techniques for encodinginformation for transmission on an idealized wireless channel. The theory of modu-lation and coding, as well as signal extraction and BER, is well beyond the scope ofthis text (see [Schwartz 1980] for a discussion of these topics). Nonetheless, Figure6.3 illustrates several physical-layer characteristics that are important in understand-ing higher-layer wireless communication protocols: For a given modulation scheme, the higher the SNR, the lower the BER. Sincea sender can increase the SNR by increasing its transmission power, a sendercan decrease the probability that a frame is received in error by increasing itstransmission power. Note, however, that there is arguably little practical gainin increasing the power beyond a certain threshold, say to decrease the BERfrom 10-12 to 10-13. There are also disadvantages associated with increasing thetransmission power: More energy must be expended by the sender (an impor-tant concern for battery-powered mobile users), and the senders transmissionsare more likely to interfere with the transmissions of another sender (seeFigure 6.4(b)). For a given SNR, a modulation technique with a higher bit transmission rate(whether in error or not) will have a higher BER. For example, in Figure 6.3,with an SNR of 10 dB, BPSK modulation with a transmission rate of 1 Mbps hasa BER of less than 10-7, while with QAM16 modulation with a transmission rateof 4 Mbps, the BER is 10-1, far too high to be practically useful. However, withan SNR of 20 dB, QAM16 modulation has a transmission rate of 4 Mbps and aBER of 10-7, while BPSK modulation has a transmission rate of only 1 Mbps anda BER that is so low as to be (literally) off the charts. If one can tolerate a BERof 10-7, the higher transmission rate offered by QAM16 would make it the pre-ferred modulation technique in this situation. These considerations give rise tothe final characteristic, described next. Dynamic selection of the physical-layer modulation technique can be used toadapt the modulation technique to channel conditions. The SNR (and hence theBER) may change as a result of mobility or due to changes in the environment.Adaptive modulation and coding are used in cellular data systems and in the802.11 WiFi and 3G cellular data networks that well study in Sections 6.3 and6.4. This allows, for example, the selection of a modulation technique that provides the highest transmission rate possible subject to a constraint on theBER, for given channel characteristics.A higher and time-varying bit error rate is not the only difference between awired and wireless link. Recall that in the case of wired broadcast links, all nodesreceive the transmissions from all other nodes. In the case of wireless links, the situ-ation is not as simple, as shown in Figure 6.4. Suppose that Station A is transmittingto Station B. Suppose also that Station C is transmitting to Station B. With the so-called hidden terminal problem, physical obstructions in the environment (forexample, a mountain or a building) may prevent A and C from hearing each otherstransmissions, even though As and Cs transmissions are indeed interfering at thedestination, B. This is shown in Figure 6.4(a). A second scenario that results inundetectable collisions at the receiver results from the fading of a signals strengthas it propagates through the wireless medium. Figure 6.4(b) illustrates the casewhere A and C are placed such that their signals are not strong enough to detect eachothers transmissions, yet their signals are strong enough to interfere with each otherat station B. As well see in Section 6.3, the hidden terminal problem and fading6.2 WIRELESS LINKS AND NETWORK CHARACTERISTICS 521make multiple access in a wireless network considerably more complex than in awired network.6.2.1 CDMARecall from Chapter 5 that when hosts communicate over a shared medium, a pro-tocol is needed so that the signals sent by multiple senders do not interfere at thereceivers. In Chapter 5 we described three classes of medium access protocols:channel partitioning, random access, and taking turns. Code division multipleaccess (CDMA) belongs to the family of channel partitioning protocols. It is preva-lent in wireless LAN and cellular technologies. Because CDMA is so important inthe wireless world, well take a quick look at CDMA now, before getting into spe-cific wireless access technologies in the subsequent sections.In a CDMA protocol, each bit being sent is encoded by multiplying the bit bya signal (the code) that changes at a much faster rate (known as the chipping rate)than the original sequence of data bits. Figure 6.5 shows a simple, idealizedCDMA encoding/decoding scenario. Suppose that the rate at which original databits reach the CDMA encoder defines the unit of time; that is, each original databit to be transmitted requires a one-bit slot time. Let di be the value of the data bitfor the ith bit slot. For mathematical convenience, we represent a data bit with a 0value as 1. Each bit slot is further subdivided into M mini-slots; in Figure 6.5, M= 8, although in practice M is much larger. The CDMA code used by the senderconsists of a sequence of M values, cm, m = 1, . . . , M, each taking a +1 or 1522 CHAPTER 6 WIRELESS AND MOBILE NETWORKSAACB CLocationb.a.0Signal strengthBFigure 6.4 Hidden terminal problem caused by obstacle (a) and fading (b)value. In the example in Figure 6.5, the M-bit CDMA code being used by thesender is (1, 1, 1, 1, 1, 1, 1, 1).To illustrate how CDMA works, let us focus on the ith data bit, di. For the mthmini-slot of the bit-transmission time of di, the output of the CDMA encoder, Zi,m, isthe value of di multiplied by the mth bit in the assigned CDMA code, cm:(6.1)Zi, m = di cm6.2 WIRELESS LINKS AND NETWORK CHARACTERISTICS 523Figure 6.5 A simple CDMA example: sender encoding, receiver decoding11 1 1-1 -1 -1 -111 1 1-1 -1 -1 -11-1 -1-1 -11 1 1 1-1 -1 -1 -11 1 1Time slot 1received inputTime slot 0received inputCode1-1 -1-1 -11 1 1 1-1 -1 -1 -11 1 1Data bitsCode 11 1 1-1 -1 -1 -111 1 1-1 -1 -1 -1d1 = -1d0 = 1Time slot 1Sender Channel output Zi,mReceiverZi,m di cm=Zi,m cmdMim=1M=Time slot 1channel outputTime slot 0channel outputTime slot 0d1 = -1d0 = 1In a simple world, with no interfering senders, the receiver would receive theencoded bits, Zi,m, and recover the original data bit, di, by computing:(6.2)The reader might want to work through the details of the example in Figure 6.5 tosee that the original data bits are indeed correctly recovered at the receiver usingEquation 6.2.The world is far from ideal, however, and as noted above, CDMA must work inthe presence of interfering senders that are encoding and transmitting their data usinga different assigned code. But how can a CDMA receiver recover a senders originaldata bits when those data bits are being tangled with bits being transmitted by othersenders? CDMA works under the assumption that the interfering transmitted bitsignals are additive. This means, for example, that if three senders send a 1 value, anda fourth sender sends a 1 value during the same mini-slot, then the receivedsignal at all receivers during that mini-slot is a 2 (since 1 1 1 1 = 2). In thepresence of multiple senders, sender s computes its encoded transmissions, Zsi,m, inexactly the same manner as in Equation 6.1. The value received at a receiver duringthe mth mini-slot of the ith bit slot, however, is now the sum of the transmitted bitsfrom all N senders during that mini-slot:Amazingly, if the senders codes are chosen carefully, each receiver can recover thedata sent by a given sender out of the aggregate signal simply by using the senderscode in exactly the same manner as in Equation 6.2:(6.3)as shown in Figure 6.6, for a two-sender CDMA example. The M-bit CDMA code beingused by the upper sender is (1, 1, 1, 1, 1, 1, 1, 1), while the CDMA code being usedby the lower sender is (1, 1, 1, 1, 1, 1, 1, 1). Figure 6.6 illustrates a receiver recoveringthe original data bits from the upper sender. Note that the receiver is able to extractthe data from sender 1 in spite of the interfering transmission from sender 2.Recall our cocktail analogy from Chapter 5. A CDMA protocol is similar to hav-ing partygoers speaking in multiple languages; in such circumstances humans areactually quite good at locking into the conversation in the language they understand,while filtering out the remaining conversations. We see here that CDMA is a partition-ing protocol in that it partitions the codespace (as opposed to time or frequency) andassigns each node a dedicated piece of the codespace.di =1M aMm=1Zi, m* cmZ*i, m = aNs=1Zsi, mdi =1M aMm=1Zi, m cm524 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.2 WIRELESS LINKS AND NETWORK CHARACTERISTICS 525Receiver 111 1 1-1 -1 -1 -111 1 1-1 -1 -1 -1Time slot 1received inputTime slot 0received inputData bitsData bits11 1 1-1 -1 -1 -111 1 1-1 -1 -1 -1CodeSenders1 1 1-11 1 1-11-1 -11 1 1 1 1CodeCode+-22 2 2 2 2-22-22 2 2 2 2-22Channel, Zi,m*Zi,m di cm=Zi,m cmdMim=1M=d1 = -1d0 = 1d1 = 1211*22 2Zi,m di cm=1 1 1d0 = 1211d1 = -1d0 = 111Figure 6.6 A two-sender CDMA exampleOur discussion here of CDMA is necessarily brief; in practice a number ofdifficult issues must be addressed. First, in order for the CDMA receivers to beable to extract a particular senders signal, the CDMA codes must be carefullychosen. Second, our discussion has assumed that the received signal strengthsfrom various senders are the same; in reality this can be difficult to achieve. Thereis a considerable body of literature addressing these and other issues related toCDMA; see [Pickholtz 1982; Viterbi 1995] for details.6.3 WiFi: 802.11 Wireless LANsPervasive in the workplace, the home, educational institutions, cafs, airports, andstreet corners, wireless LANs are now one of the most important access networktechnologies in the Internet today. Although many technologies and standards forwireless LANs were developed in the 1990s, one particular class of standards hasclearly emerged as the winner: the IEEE 802.11 wireless LAN, also known asWiFi. In this section, well take a close look at 802.11 wireless LANs, examiningits frame structure, its medium access protocol, and its internetworking of 802.11LANs with wired Ethernet LANs.There are several 802.11 standards for wireless LAN technology, including802.11b, 802.11a, and 802.11g. Table 6.1 summarizes the main characteristics ofthese standards. 802.11g is by far the most popular technology. A number of dual-mode (802.11a/g) and tri-mode (802.11a/b/g) devices are also available.The three 802.11 standards share many characteristics. They all use the samemedium access protocol, CSMA/CA, which well discuss shortly. All three use thesame frame structure for their link-layer frames as well. All three standards have theability to reduce their transmission rate in order to reach out over greater distances.And all three standards allow for both infrastructure mode and ad hoc mode, aswell also shortly discuss. However, as shown in Table 6.1, the three standards havesome major differences at the physical layer.The 802.11b wireless LAN has a data rate of 11 Mbps and operates in theunlicensed frequency band of 2.42.485 GHz, competing for frequency spectrumwith 2.4 GHz phones and microwave ovens. 802.1la wireless LANs can run at526 CHAPTER 6 WIRELESS AND MOBILE NETWORKSTable 6.1 Summary of IEEE 802.11 standards Standard Frequency Range (United States) Data Rate802.11b 2.42.485 GHz up to 11 Mbps802.11a 5.15.8 GHz up to 54 Mbps802.11g 2.42.485 GHz up to 54 Mbpssignificantly higher bit rates, but do so at higher frequencies. By operating at ahigher frequency, 802.11a LANs have a shorter transmission distance for a givenpower level and suffer more from multipath propagation. 802.11g LANs, operat-ing in the same lower-frequency band as 802.11b and being backwards compati-ble with 802.11b (so one can upgrade 802.11b clients incrementally) yet with thehigher-speed transmission rates of 802.11a, allows users to have their cake andeat it too.A relatively new WiFi standard, 802.11n [IEEE 802.11n 2012], uses multiple-input multiple-output (MIMO) antennas; i.e., two or more antennas on the sending side and two or more antennas on the receiving side that are transmit-ting/receiving different signals [Diggavi 2004]. Depending on the modulationscheme used, transmission rates of several hundred megabits per second are pos-sible with 802.11n.6.3.1 The 802.11 ArchitectureFigure 6.7 illustrates the principal components of the 802.11 wireless LAN archi-tecture. The fundamental building block of the 802.11 architecture is the basicservice set (BSS). A BSS contains one or more wireless stations and a central6.3 WIFI: 802.11 WIRELESS LANS 527InternetSwitch or router APBSS 1 BSS 2 APFigure 6.7 IEEE 802.11 LAN architecturebase station, known as an access point (AP) in 802.11 parlance. Figure 6.7shows the AP in each of two BSSs connecting to an interconnection device (suchas a switch or router), which in turn leads to the Internet. In a typical home net-work, there is one AP and one router (typically integrated together as one unit)that connects the BSS to the Internet.As with Ethernet devices, each 802.11 wireless station has a 6-byte MACaddress that is stored in the firmware of the stations adapter (that is, 802.11 networkinterface card). Each AP also has a MAC address for its wireless interface. As withEthernet, these MAC addresses are administered by IEEE and are (in theory) glob-ally unique.As noted in Section 6.1, wireless LANs that deploy APs are often referred toas infrastructure wireless LANs, with the infrastructure being the APs alongwith the wired Ethernet infrastructure that interconnects the APs and a router.Figure 6.8 shows that IEEE 802.11 stations can also group themselves togetherto form an ad hoc networka network with no central control and with no con-nections to the outside world. Here, the network is formed on the fly, bymobile devices that have found themselves in proximity to each other, that havea need to communicate, and that find no preexisting network infrastructure intheir location. An ad hoc network might be formed when people with laptops gettogether (for example, in a conference room, a train, or a car) and want toexchange data in the absence of a centralized AP. There has been tremendousinterest in ad hoc networking, as communicating portable devices continue toproliferate. In this section, though, well focus our attention on infrastructurewireless LANs.528 CHAPTER 6 WIRELESS AND MOBILE NETWORKSBSSFigure 6.8 An IEEE 802.11 ad hoc networkChannels and AssociationIn 802.11, each wireless station needs to associate with an AP before it can send orreceive network-layer data. Although all of the 802.11 standards use association,well discuss this topic specifically in the context of IEEE 802.11b/g.When a network administrator installs an AP, the administrator assigns a one-or two-word Service Set Identifier (SSID) to the access point. (When you viewavailable networks in Microsoft Windows XP, for example, a list is displayedshowing the SSID of each AP in range.) The administrator must also assign a chan-nel number to the AP. To understand channel numbers, recall that 802.11 operates inthe frequency range of 2.4 GHz to 2.485 GHz. Within this 85 MHz band, 802.11defines 11 partially overlapping channels. Any two channels are non-overlapping ifand only if they are separated by four or more channels. In particular, the set ofchannels 1, 6, and 11 is the only set of three non-overlapping channels. This meansthat an administrator could create a wireless LAN with an aggregate maximumtransmission rate of 33 Mbps by installing three 802.11b APs at the same physicallocation, assigning channels 1, 6, and 11 to the APs, and interconnecting each of theAPs with a switch.Now that we have a basic understanding of 802.11 channels, lets describe aninteresting (and not completely uncommon) situationthat of a WiFi jungle. AWiFi jungle is any physical location where a wireless station receives a sufficientlystrong signal from two or more APs. For example, in many cafs in New York City,a wireless station can pick up a signal from numerous nearby APs. One of the APsmight be managed by the caf, while the other APs might be in residential apart-ments near the caf. Each of these APs would likely be located in a different IP sub-net and would have been independently assigned a channel.Now suppose you enter such a WiFi jungle with your portable computer,seeking wireless Internet access and a blueberry muffin. Suppose there are fiveAPs in the WiFi jungle. To gain Internet access, your wireless station needs to joinexactly one of the subnets and hence needs to associate with exactly one of theAPs. Associating means the wireless station creates a virtual wire between itselfand the AP. Specifically, only the associated AP will send data frames (that is,frames containing data, such as a datagram) to your wireless station, and yourwireless station will send data frames into the Internet only through the associatedAP. But how does your wireless station associate with a particular AP? And morefundamentally, how does your wireless station know which APs, if any, are outthere in the jungle?The 802.11 standard requires that an AP periodically send beacon frames, eachof which includes the APs SSID and MAC address. Your wireless station, knowingthat APs are sending out beacon frames, scans the 11 channels, seeking beaconframes from any APs that may be out there (some of which may be transmitting onthe same channelits a jungle out there!). Having learned about available APs6.3 WIFI: 802.11 WIRELESS LANS 529from the beacon frames, you (or your wireless host) select one of the APs forassociation.The 802.11 standard does not specify an algorithm for selecting which of theavailable APs to associate with; that algorithm is left up to the designers of the 802.11firmware and software in your wireless host. Typically, the host chooses the APwhose beacon frame is received with the highest signal strength. While a high signalstrength is good (see, e.g., Figure 6.3), signal strength is not the only AP characteris-tic that will determine the performance a host receives. In particular, its possible thatthe selected AP may have a strong signal, but may be overloaded with other affiliatedhosts (that will need to share the wireless bandwidth at that AP), while an unloadedAP is not selected due to a slightly weaker signal. A number of alternative ways ofchoosing APs have thus recently been proposed [Vasudevan 2005; Nicholson 2006;Sundaresan 2006]. For an interesting and down-to-earth discussion of how signalstrength is measured, see [Bardwell 2004].The process of scanning channels and listening for beacon frames is known aspassive scanning (see Figure 6.9a). A wireless host can also perform active scan-ning, by broadcasting a probe frame that will be received by all APs within the wire-less hosts range, as shown in Figure 6.9b. APs respond to the probe request framewith a probe response frame. The wireless host can then choose the AP with whichto associate from among the responding APs.After selecting the AP with which to associate, the wireless host sends an asso-ciation request frame to the AP, and the AP responds with an association responseframe. Note that this second request/response handshake is needed with active scan-ning, since an AP responding to the initial probe request frame doesnt know whichof the (possibly many) responding APs the host will choose to associate with, inmuch the same way that a DHCP client can choose from among multiple DHCPservers (see Figure 4.21). Once associated with an AP, the host will want to join thesubnet (in the IP addressing sense of Section 4.4.2) to which the AP belongs. Thus,the host will typically send a DHCP discovery message (see Figure 4.21) into thesubnet via the AP in order to obtain an IP address on the subnet. Once the address isobtained, the rest of the world then views that host simply as another host with an IPaddress in that subnet.In order to create an association with a particular AP, the wireless station maybe required to authenticate itself to the AP. 802.11 wireless LANs provide a numberof alternatives for authentication and access. One approach, used by many compa-nies, is to permit access to a wireless network based on a stations MAC address. Asecond approach, used by many Internet cafs, employs usernames and passwords.In both cases, the AP typically communicates with an authentication server, relay-ing information between the wireless end-point station and the authentication serverusing a protocol such as RADIUS [RFC 2865] or DIAMETER [RFC 3588]. Sepa-rating the authentication server from the AP allows one authentication server toserve many APs, centralizing the (often sensitive) decisions of authentication andaccess within the single server, and keeping AP costs and complexity low. Well see530 CHAPTER 6 WIRELESS AND MOBILE NETWORKSin Section 8.8 that the new IEEE 802.11i protocol defining security aspects of the802.11 protocol family takes precisely this approach.6.3.2 The 802.11 MAC ProtocolOnce a wireless station is associated with an AP, it can start sending and receivingdata frames to and from the access point. But because multiple stations may want totransmit data frames at the same time over the same channel, a multiple access proto-col is needed to coordinate the transmissions. Here, a station is either a wireless sta-tion or an AP. As discussed in Chapter 5 and Section 6.2.1, broadly speaking thereare three classes of multiple access protocols: channel partitioning (includingCDMA), random access, and taking turns. Inspired by the huge success of Ethernetand its random access protocol, the designers of 802.11 chose a random access proto-col for 802.11 wireless LANs. This random access protocol is referred to as CSMAwith collision avoidance, or more succinctly as CSMA/CA. As with EthernetsCSMA/CD, the CSMA in CSMA/CA stands for carrier sense multiple access,meaning that each station senses the channel before transmitting, and refrains fromtransmitting when the channel is sensed busy. Although both Ethernet and 802.11 usecarrier-sensing random access, the two MAC protocols have important differences.6.3 WIFI: 802.11 WIRELESS LANS 5311132H1AP 2AP 1BBS 1a. Passive scanning 1. Beacon frames sent from APs 2. Association Request frame sent:H1 to selected AP 3. Association Response frame sent:Selected AP to H1a. Active scanning 1. Probe Request frame broadcast from H1 2. Probes Response frame sent from APs 3. Association Request frame sent:H1 to selected AP 4. Association Response frame sent:Selected AP to H1BBS 22243H1AP 2AP 1BBS 1 BBS 21Figure 6.9 Active and passive scanning for access pointsFirst, instead of using collision detection, 802.11 uses collision-avoidance tech-niques. Second, because of the relatively high bit error rates of wireless channels,802.11 (unlike Ethernet) uses a link-layer acknowledgment/retransmission (ARQ)scheme. Well describe 802.11s collision-avoidance and link-layer acknowledgmentschemes below.Recall from Sections 5.3.2 and 5.4.2 that with Ethernets collision-detectionalgorithm, an Ethernet station listens to the channel as it transmits. If, while trans-mitting, it detects that another station is also transmitting, it aborts its transmissionand tries to transmit again after waiting a small, random amount of time. Unlike the802.3 Ethernet protocol, the 802.11 MAC protocol does not implement collisiondetection. There are two important reasons for this: The ability to detect collisions requires the ability to send (the stations own sig-nal) and receive (to determine whether another station is also transmitting) at thesame time. Because the strength of the received signal is typically very smallcompared to the strength of the transmitted signal at the 802.11 adapter, it iscostly to build hardware that can detect a collision. More importantly, even if the adapter could transmit and listen at the same time(and presumably abort transmission when it senses a busy channel), the adapterwould still not be able to detect all collisions, due to the hidden terminal problemand fading, as discussed in Section 6.2.Because 802.11wireless LANs do not use collision detection, once a stationbegins to transmit a frame, it transmits the frame in its entirety; that is, once a sta-tion gets started, there is no turning back. As one might expect, transmitting entireframes (particularly long frames) when collisions are prevalent can significantlydegrade a multiple access protocols performance. In order to reduce the likelihoodof collisions, 802.11 employs several collision-avoidance techniques, which wellshortly discuss.Before considering collision avoidance, however, well first need to examine802.11s link-layer acknowledgment scheme. Recall from Section 6.2 that when astation in a wireless LAN sends a frame, the frame may not reach the destinationstation intact for a variety of reasons. To deal with this non-negligible chance of fail-ure, the 802.11 MAC protocol uses link-layer acknowledgments. As shown inFigure 6.10, when the destination station receives a frame that passes the CRC, itwaits a short period of time known as the Short Inter-frame Spacing (SIFS) andthen sends back an acknowledgment frame. If the transmitting station does notreceive an acknowledgment within a given amount of time, it assumes that an errorhas occurred and retransmits the frame, using the CSMA/CA protocol to access thechannel. If an acknowledgment is not received after some fixed number of retrans-missions, the transmitting station gives up and discards the frame.532 CHAPTER 6 WIRELESS AND MOBILE NETWORKSHaving discussed how 802.11 uses link-layer acknowledgments, were now ina position to describe the 802.11 CSMA/CA protocol. Suppose that a station (wire-less station or an AP) has a frame to transmit.1. If initially the station senses the channel idle, it transmits its frame after a shortperiod of time known as the Distributed Inter-frame Space (DIFS); seeFigure 6.10.2. Otherwise, the station chooses a random backoff value using binary exponen-tial backoff (as we encountered inSection 5.3.2) and counts down this valuewhen the channel is sensed idle. While the channel is sensed busy, the countervalue remains frozen.3. When the counter reaches zero (note that this can only occur while the channelis sensed idle), the station transmits the entire frame and then waits for anacknowledgment.6.3 WIFI: 802.11 WIRELESS LANS 533DestinationDIFSSIFSdataackSourceFigure 6.10 802.11 uses link-layer acknowledgments4. If an acknowledgment is received, the transmitting station knows that its framehas been correctly received at the destination station. If the station has anotherframe to send, it begins the CSMA/CA protocol at step 2. If the acknowledg-ment isnt received, the transmitting station reenters the backoff phase in step2, with the random value chosen from a larger interval.Recall that under Ethernets CSMA/CD, multiple access protocol (Section5.3.2), a station begins transmitting as soon as the channel is sensed idle. WithCSMA/CA, however, the station refrains from transmitting while counting down,even when it senses the channel to be idle.Why do CSMA/CD and CDMA/CA takesuch different approaches here?To answer this question, lets consider a scenario in which two stations eachhave a data frame to transmit, but neither station transmits immediately becauseeach senses that a third station is already transmitting. With Ethernets CSMA/CD,the two stations would each transmit as soon as they detect that the third station hasfinished transmitting. This would cause a collision, which isnt a serious issue inCSMA/CD, since both stations would abort their transmissions and thus avoid theuseless transmissions of the remainders of their frames. In 802.11, however, the sit-uation is quite different. Because 802.11 does not detect a collision and abort trans-mission, a frame suffering a collision will be transmitted in its entirety. The goal in802.11 is thus to avoid collisions whenever possible. In 802.11, if the two stationssense the channel busy, they both immediately enter random backoff, hopefullychoosing different backoff values. If these values are indeed different, once thechannel becomes idle, one of the two stations will begin transmitting before theother, and (if the two stations are not hidden from each other) the losing stationwill hear the winning stations signal, freeze its counter, and refrain from trans-mitting until the winning station has completed its transmission. In this manner, acostly collision is avoided. Of course, collisions can still occur with 802.11 in thisscenario: The two stations could be hidden from each other, or the two stationscould choose random backoff values that are close enough that the transmissionfrom the station starting first have yet to reach the second station. Recall that weencountered this problem earlier in our discussion of random access algorithms inthe context of Figure 5.12.Dealing with Hidden Terminals: RTS and CTSThe 802.11 MAC protocol also includes a nifty (but optional) reservation schemethat helps avoid collisions even in the presence of hidden terminals. Lets investi-gate this scheme in the context of Figure 6.11, which shows two wireless stationsand one access point. Both of the wireless stations are within range of the AP(whose coverage is shown as a shaded circle) and both have associated with the AP.However, due to fading, the signal ranges of wireless stations are limited to the534 CHAPTER 6 WIRELESS AND MOBILE NETWORKSinteriors of the shaded circles shown in Figure 6.11. Thus, each of the wireless sta-tions is hidden from the other, although neither is hidden from the AP.Lets now consider why hidden terminals can be problematic. Suppose StationH1 is transmitting a frame and halfway through H1s transmission, Station H2 wantsto send a frame to the AP. H2, not hearing the transmission from H1, will first wait aDIFS interval and then transmit the frame, resulting in a collision. The channel willtherefore be wasted during the entire period of H1s transmission as well as duringH2s transmission.In order to avoid this problem, the IEEE 802.11 protocol allows a station to usea short Request to Send (RTS) control frame and a short Clear to Send (CTS) con-trol frame to reserve access to the channel. When a sender wants to send a DATAframe, it can first send an RTS frame to the AP, indicating the total time required totransmit the DATA frame and the acknowledgment (ACK) frame. When the APreceives the RTS frame, it responds by broadcasting a CTS frame. This CTS frameserves two purposes: It gives the sender explicit permission to send and alsoinstructs the other stations not to send for the reserved duration.Thus, in Figure 6.12, before transmitting a DATA frame, H1 first broadcasts anRTS frame, which is heard by all stations in its circle, including the AP. The AP thenresponds with a CTS frame, which is heard by all stations within its range, includ-ing H1 and H2. Station H2, having heard the CTS, refrains from transmitting for thetime specified in the CTS frame. The RTS, CTS, DATA, and ACK frames are shownin Figure 6.12.6.3 WIFI: 802.11 WIRELESS LANS 535APH1 H2Figure 6.11 Hidden terminal example: H1 is hidden from H2, and vice versaThe use of the RTS and CTS frames can improve performance in two importantways: The hidden station problem is mitigated, since a long DATA frame is transmittedonly after the channel has been reserved.536 CHAPTER 6 WIRELESS AND MOBILE NETWORKSDestination All other nodes Defer access Source DIFSACKSIFSSIFSSIFSDATA CTS CTSACKRTS Figure 6.12 Collision avoidance using the RTS and CTS frames Because the RTS and CTS frames are short, a collision involving an RTS or CTSframe will last only for the duration of the short RTS or CTS frame. Once theRTS and CTS frames are correctly transmitted, the following DATA and ACKframes should be transmitted without collisions.You are encouraged to check out the 802.11 applet in the textbooks companion Website. This interactive applet illustrates the CSMA/CA protocol, including theRTS/CTS exchange sequence.Although the RTS/CTS exchange can help reduce collisions, it also introducesdelay and consumes channel resources. For this reason, the RTS/CTS exchange isonly used (if at all) to reserve the channel for the transmission of a long DATAframe. In practice, each wireless station can set an RTS threshold such that theRTS/CTS sequence is used only when the frame is longer than the threshold. Formany wireless stations, the default RTS threshold value is larger than the maximumframe length, so the RTS/CTS sequence is skipped for all DATA frames sent.Using 802.11 as a Point-to-Point LinkOur discussion so far has focused on the use of 802.11 in a multiple access setting.We should mention that if two nodes each have a directional antenna, they can pointtheir directional antennas at each other and run the 802.11 protocol over what isessentially a point-to-point link. Given the low cost of commodity 802.11 hardware,the use of directional antennas and an increased transmission power allow 802.11 tobe used as an inexpensive means of providing wireless point-to-point connectionsover tens of kilometers distance. [Raman 2007] describes such a multi-hop wirelessnetwork operating in the rural Ganges plains in India that contains point-to-point802.11 links.6.3.3 The IEEE 802.11 FrameAlthough the 802.11 frame shares many similarities with an Ethernet frame, it alsocontains a number of fields that are specific to its use for wireless links. The 802.11frame is shown in Figure 6.13. The numbers above each of the fields in the framerepresent the lengths of the fields in bytes; the numbers above each of the subfieldsin the frame control field represent the lengths of the subfields in bits. Lets nowexamine the fields in the frame as well as some of the more important subfields inthe frames control field.Payload and CRC FieldsAt the heart of the frame is the payload, which typically consists of an IP datagramor an ARP packet. Although the field is permitted to be as long as 2,312 bytes, it is6.3 WIFI: 802.11 WIRELESS LANS 537typically fewer than 1,500 bytes, holding an IP datagram or an ARP packet. As withan Ethernet frame, an 802.11 frame includes a 32-bit cyclic redundancy check(CRC) so that the receiver can detect bit errors in the received frame. As weve seen,bit errors are much more common in wireless LANs than in wired LANs, so theCRC is even more useful here.Address FieldsPerhaps the most striking difference in the 802.11 frame is that it has four addressfields, each of which can hold a 6-byte MAC address. But why four address fields?Doesnt a source MAC field and destination MAC field suffice, as they do forEthernet? It turns out that three address fields are needed for internetworking pur-posesspecifically, for moving the network-layer datagram from a wireless stationthrough an AP to a router interface. The fourth address field is used when APs for-ward frames to each other in ad hoc mode. Since we are only considering infrastruc-ture networks here, lets focus our attention on the first three address fields. The802.11 standard defines these fields as follows: Address 2 is the MAC address of the station that transmits the frame. Thus, if awireless station transmits the frame, that stations MAC address is inserted in theaddress 2 field. Similarly, if an AP transmits the frame, the APs MAC address isinserted in the address 2 field. Address 1 is the MAC address of the wireless station that is to receive theframe. Thus if a mobile wireless station transmits the frame, address 1 containsthe MAC address of the destination AP. Similarly, if an AP transmits the frame,address 1 contains the MAC address of the destination wireless station.538 CHAPTER 6 WIRELESS AND MOBILE NETWORKSFramecontrol22 2 4 1 1 1 1 1 1 1 12 6 6 6 2 6 0-2312 4Frame (numbers indicate field length in bytes):Address1Duration Payload CRCProtocolversionToAPFromAPMorefragPowermgtMoredataAddress2Address3Address4SeqcontrolType Subtype Retry WEP RsvdFrame control field expanded (numbers indicate field length in bits):Figure 6.13 The 802.11 frame To understand address 3, recall that the BSS (consisting of the AP and wireless sta-tions) is part of a subnet, and that this subnet connects to other subnets via somerouter interface. Address 3 contains the MAC address of this router interface.To gain further insight into the purpose of address 3, lets walk through an inter-networking example in the context of Figure 6.14. In this figure, there are two APs,each of which is responsible for a number of wireless stations. Each of the APs hasa direct connection to a router, which in turn connects to the global Internet. Weshould keep in mind that an AP is a link-layer device, and thus neither speaks IPnor understands IP addresses. Consider now moving a datagram from the routerinterface R1 to the wireless Station H1. The router is not aware that there is an APbetween it and H1; from the routers perspective, H1 is just a host in one of the sub-nets to which it (the router) is connected. The router, which knows the IP address of H1 (from the destination address ofthe datagram), uses ARP to determine the MAC address of H1, just as in anordinary Ethernet LAN. After obtaining H1s MAC address, router interface R1encapsulates the datagram within an Ethernet frame. The source address field ofthis frame contains R1s MAC address, and the destination address field containsH1s MAC address.6.3 WIFI: 802.11 WIRELESS LANS 539InternetRouterAPH1R1BSS 1BSS 2APFigure 6.14 The use of address fields in 802.11 frames: Sendingframes between H1 and R1 When the Ethernet frame arrives at the AP, the AP converts the 802.3 Ethernetframe to an 802.11 frame before transmitting the frame into the wireless chan-nel. The AP fills in address 1 and address 2 with H1s MAC address and itsown MAC address, respectively, as described above. For address 3, the APinserts the MAC address of R1. In this manner, H1 can determine (fromaddress 3) the MAC address of the router interface that sent the datagram intothe subnet.Now consider what happens when the wireless station H1 responds by moving adatagram from H1 to R1. H1 creates an 802.11 frame, filling the fields for address 1 and address 2 with theAPs MAC address and H1s MAC address, respectively, as described above. Foraddress 3, H1 inserts R1s MAC address. When the AP receives the 802.11 frame, it converts the frame to an Ethernetframe. The source address field for this frame is H1s MAC address, and the des-tination address field is R1s MAC address. Thus, address 3 allows the AP todetermine the appropriate destination MAC address when constructing the Eth-ernet frame.In summary, address 3 plays a crucial role for internetworking the BSS with a wiredLAN.Sequence Number, Duration, and Frame Control FieldsRecall that in 802.11, whenever a station correctly receives a frame from anotherstation, it sends back an acknowledgment. Because acknowledgments can get lost,the sending station may send multiple copies of a given frame. As we saw in our dis-cussion of the rdt2.1 protocol (Section 3.4.1), the use of sequence numbers allowsthe receiver to distinguish between a newly transmitted frame and the retransmis-sion of a previous frame. The sequence number field in the 802.11 frame thus servesexactly the same purpose here at the link layer as it did in the transport layer inChapter 3.Recall that the 802.11 protocol allows a transmitting station to reserve the chan-nel for a period of time that includes the time to transmit its data frame and the timeto transmit an acknowledgment. This duration value is included in the frames dura-tion field (both for data frames and for the RTS and CTS frames).As shown in Figure 6.13, the frame control field includes many subfields. Wellsay just a few words about some of the more important subfields; for a more com-plete discussion, you are encouraged to consult the 802.11 specification [Held 2001;Crow 1997; IEEE 802.11 1999]. The type and subtype fields are used to distinguishthe association, RTS, CTS, ACK, and data frames. The to and from fields are usedto define the meanings of the different address fields. (These meanings change540 CHAPTER 6 WIRELESS AND MOBILE NETWORKSdepending on whether ad hoc or infrastructure modes are used and, in the case ofinfrastructure mode, whether a wireless station or an AP is sending the frame.)Finally the WEP field indicates whether encryption is being used or not. (WEP isdiscussed in Chapter 8.)6.3.4 Mobility in the Same IP SubnetIn order to increase the physical range of a wireless LAN, companies and universitieswill often deploy multiple BSSs within the same IP subnet. This naturally raises theissue of mobility among the BSSshow do wireless stations seamlessly move fromone BSS to another while maintaining ongoing TCP sessions? As well see in this sub-section, mobility can be handled in a relatively straightforward manner when the BSSsare part of the subnet. When stations move between subnets, more sophisticated mobil-ity management protocols will be needed, such as those well study in Sections 6.5and 6.6.Lets now look at a specific example of mobility between BSSs in the samesubnet. Figure 6.15 shows two interconnected BSSs with a host, H1, movingfrom BSS1 to BSS2. Because in this example the interconnection device thatconnects the two BSSs is not a router, all of the stations in the two BSSs, includ-ing the APs, belong to the same IP subnet. Thus, when H1 moves from BSS1 toBSS2, it may keep its IP address and all of its ongoing TCP connections. If theinterconnection device were a router, then H1 would have to obtain a new IPaddress in the subnet in which it was moving. This address change would disrupt(and eventually terminate) any on-going TCP connections at H1. In Section 6.6,well see how a network-layer mobility protocol, such as mobile IP, can be usedto avoid this problem.6.3 WIFI: 802.11 WIRELESS LANS 541BSS 1 BSS 2 H1SwitchAP 1 AP 2 Figure 6.15 Mobility in the same subnetBut what specifically happens when H1 moves from BSS1 to BSS2? As H1wanders away from AP1, H1 detects a weakening signal from AP1 and starts to scanfor a stronger signal. H1 receives beacon frames from AP2 (which in many corpo-rate and university settings will have the same SSID as AP1). H1 then disassociateswith AP1 and associates with AP2, while keeping its IP address and maintaining itsongoing TCP sessions.This addresses the handoff problem from the host and AP viewpoint. But whatabout the switch in Figure 6.15? How does it know that the host has moved fromone AP to another? As you may recall from Chapter 5, switches are self-learningand automatically build their forwarding tables. This self-learning feature nicelyhandles occasional moves (for example, when an employee gets transferred fromone department to another); however, switches were not designed to support highlymobile users who want to maintain TCP connections while moving between BSSs.To appreciate the problem here, recall that before the move, the switch has an entryin its forwarding table that pairs H1s MAC address with the outgoing switch inter-face through which H1 can be reached. If H1 is initially in BSS1, then a datagramdestined to H1 will be directed to H1 via AP1. Once H1 associates with BSS2, how-ever, its frames should be directed to AP2. One solution (a bit of a hack, really) isfor AP2 to send a broadcast Ethernet frame with H1s source address to the switchjust after the new association. When the switch receives the frame, it updates its for-warding table, allowing H1 to be reached via AP2. The 802.11f standards group isdeveloping an inter-AP protocol to handle these and related issues.6.3.5 Advanced Features in 802.11Well wrap up our coverage of 802.11 with a short discussion of two advanced capa-bilities found in 802.11 networks. As well see, these capabilities are not completelyspecified in the 802.11 standard, but rather are made possible by mechanisms speci-fied in the standard. This allows different vendors to implement these capabilitiesusing their own (proprietary) approaches, presumably giving them an edge over thecompetition.802.11 Rate AdaptationWe saw earlier in Figure 6.3 that different modulation techniques (with the differ-ent transmission rates that they provide) are appropriate for different SNR scenar-ios. Consider for example a mobile 802.11 user who is initially 20 meters awayfrom the base station, with a high signal-to-noise ratio. Given the high SNR, theuser can communicate with the base station using a physical-layer modulation tech-nique that provides high transmission rates while maintaining a low BER. This isone happy user! Suppose now that the user becomes mobile, walking away from542 CHAPTER 6 WIRELESS AND MOBILE NETWORKSthe base station, with the SNR falling as the distance from the base stationincreases. In this case, if the modulation technique used in the 802.11 protocoloperating between the base station and the user does not change, the BER willbecome unacceptably high as the SNR decreases, and eventually no transmittedframes will be received correctly.For this reason, some 802.11 implementations have a rate adaptation capabilitythat adaptively selects the underlying physical-layer modulation technique to usebased on current or recent channel characteristics. If a node sends two frames in arow without receiving an acknowledgment (an implicit indication of bit errors onthe channel), the transmission rate falls back to the next lower rate. If 10 frames in arow are acknowledged, or if a timer that tracks the time since the last fallbackexpires, the transmission rate increases to the next higher rate. This rate adaptationmechanism shares the same probing philosophy as TCPs congestion-controlmechanismwhen conditions are good (reflected by ACK receipts), the transmis-sion rate is increased until something bad happens (the lack of ACK receipts);when something bad happens, the transmission rate is reduced. 802.11 rate adap-tation and TCP congestion control are thus similar to the young child who is con-stantly pushing his/her parents for more and more (say candy for a young child, latercurfew hours for the teenager) until the parents finally say Enough! and the childbacks off (only to try again later after conditions have hopefully improved!). A num-ber of other schemes have also been proposed to improve on this basic automaticrate-adjustment scheme [Kamerman 1997; Holland 2001; Lacage 2004].Power ManagementPower is a precious resource in mobile devices, and thus the 802.11 standard pro-vides power-management capabilities that allow 802.11 nodes to minimize theamount of time that their sense, transmit, and receive functions and other circuitryneed to be on. 802.11 power management operates as follows. A node is able toexplicitly alternate between sleep and wake states (not unlike a sleepy student in aclassroom!). A node indicates to the access point that it will be going to sleep by set-ting the power-management bit in the header of an 802.11 frame to 1. A timer in thenode is then set to wake up the node just before the AP is scheduled to send its bea-con frame (recall that an AP typically sends a beacon frame every 100 msec). Sincethe AP knows from the set power-transmission bit that the node is going to sleep, it(the AP) knows that it should not send any frames to that node, and will buffer anyframes destined for the sleeping host for later transmission.A node will wake up just before the AP sends a beacon frame, and quickly enterthe fully active state (unlike the sleepy student, this wakeup requires only 250microseconds [Kamerman 1997]!). The beacon frames sent out by the AP contain alist of nodes whose frames have been buffered at the AP. If there are no bufferedframes for the node, it can go back to sleep. Otherwise, the node can explicitly6.3 WIFI: 802.11 WIRELESS LANS 543request that the buffered frames be sent by sending a polling message to the AP.With an inter-beacon time of 100 msec, a wakeup time of 250 microseconds, and asimilarly small time to receive a beacon frame and check to ensure that there are nobuffered frames, a node that has no frames to send or receive can be asleep 99% ofthe time, resulting in a significant energy savings.6.3.6 Personal Area Networks: Bluetooth and ZigbeeAs illustrated in Figure 6.2, the IEEE 802.11 WiFi standard is aimed at communi-cation among devices separated by up to 100 meters (except when 802.11 is used ina point-to-point configuration with a directional antenna). Two other IEEE 802protocolsBluetooth and Zigbee (defined in the IEEE 802.15.1 and IEEE 802.15.4standards [IEEE 802.15 2012]) and WiMAX (defined in the IEEE 802.16 standard[IEEE 802.16d 2004; IEEE 802.16e 2005])are standards for communicatingover shorter and longer distances, respectively. We will touch on WiMAX brieflywhen we discuss cellular data networks in Section 6.4, and so here, we will focuson networks for shorter distances.BluetoothAn IEEE 802.15.1 network operates over a short range, at low power, and at lowcost. It is essentially a low-power, short-range, low-rate cable replacement technol-ogy for interconnecting notebooks, peripheral devices, cellular phones, and smart-phones, whereas 802.11 is a higher-power, medium-range, higher-rate accesstechnology. For this reason, 802.15.1 networks are sometimes referred to as wirelesspersonal area networks (WPANs). The link and physical layers of 802.15.1 are basedon the earlier Bluetooth specification for personal area networks [Held 2001, Bis-dikian 2001]. 802.15.1 networks operate in the 2.4 GHz unlicensed radio band in aTDM manner, with time slots of 625 microseconds. During each time slot, a sendertransmits on one of 79 channels, with the channel changing in a known but pseudo-random manner from slot to slot. This form of channel hopping, known asfrequency-hopping spread spectrum (FHSS), spreads transmissions in time overthe frequency spectrum. 802.15.1 can provide data rates up to 4 Mbps.802.15.1 networks are ad hoc networks: No network infrastructure (e.g., anaccess point) is needed to interconnect 802.15.1 devices. Thus, 802.15.1 devicesmust organize themselves. 802.15.1 devices are first organized into a piconet of upto eight active devices, as shown in Figure 6.16. One of these devices is designatedas the master, with the remaining devices acting as slaves. The master node trulyrules the piconetits clock determines time in the piconet, it can transmit in eachodd-numbered slot, and a slave can transmit only after the master has communicatedwith it in the previous slot and even then the slave can only transmit to the master.In addition to the slave devices, there can also be up to 255 parked devices in the544 CHAPTER 6 WIRELESS AND MOBILE NETWORKSnetwork. These devices cannot communicate until their status has been changedfrom parked to active by the master node.For more information about 802.15.1 WPANs, the interested reader should con-sult the Bluetooth references [Held 2001, Bisdikian 2001] or the official IEEE802.15 Web site [IEEE 802.15 2012].ZigbeeA second personal area network standardized by the IEEE is the 802.14.5 standard[IEEE 802.15 2012] known as Zigbee. While Bluetooth networks provide a cablereplacement data rate of over a Megabit per second, Zigbee is targeted at lower-powered, lower-data-rate, lower-duty-cycle applications than Bluetooth. While wemay tend to think that bigger and faster is better, not all network applications needhigh bandwidth and the consequent higher costs (both economic and power costs).For example, home temperature and light sensors, security devices, and wall-mounted switches are all very simple, low-power, low-duty-cycle, low-cost devices.Zigbee is thus well-suited for these devices. Zigbee defines channel rates of 20, 40,100, and 250 Kbps, depending on the channel frequency.Nodes in a Zigbee network come in two flavors. So-called reduced-functiondevices operate as slave devices under the control of a single full-functiondevice, much as Bluetooth slave devices. A full-function device can operate as amaster device as in Bluetooth by controlling multiple slave devices, and multiplefull-function devices can additionally be configured into a mesh network inwhich full-function devices route frames amongst themselves. Zigbee shares6.3 WIFI: 802.11 WIRELESS LANS 545Radius ofcoverageMaster deviceSlave deviceParked deviceKey:MMSSS SPPPPPFigure 6.16 A Bluetooth piconetmany protocol mechanisms that weve already encountered in other link-layerprotocols: beacon frames and link-layer acknowledgments (similar to 802.11),carrier-sense random access protocols with binary exponential backoff (similarto 802.11 and Ethernet), and fixed, guaranteed allocation of time slots (similar toDOCSIS).Zigbee networks can be configured in many different ways. Lets consider thesimple case of a single full-function device controlling multiple reduced-functiondevices in a time-slotted manner using beacon frames. Figure 6.17 shows the casewhere the Zigbee network divides time into recurring super frames, each of whichbegins with a beacon frame. Each beacon frame divides the super frame into anactive period (during which devices may transmit) and an inactive period (duringwhich all devices, including the controller, can sleep and thus conserve power). Theactive period consists of 16 time slots, some of which are used by devices in aCSMA/CA random access manner, and some of which are allocated by the con-troller to specific devices, thus providing guaranteed channel access for thosedevices. More details about Zigbee networks can be found at [Baronti 2007, IEEE802.15.4 2012].6.4 Cellular Internet AccessIn the previous section we examined how an Internet host can access the Internetwhen inside a WiFi hotspotthat is, when it is within the vicinity of an 802.11access point. But most WiFi hotspots have a small coverage area of between 10 and100 meters in diameter. What do we do then when we have a desperate need forwireless Internet access and we cannot access a WiFi hotspot?Given that cellular telephony is now ubiquitous in many areas throughout theworld, a natural strategy is to extend cellular networks so that they support notonly voice telephony but wireless Internet access as well. Ideally, this Internetaccess would be at a reasonably high speed and would provide for seamless546 CHAPTER 6 WIRELESS AND MOBILE NETWORKSFigure 6.17 Zigbee 802.14.4 super-frame structureBeaconGuaranteed slotsContention slots Inactive periodSuper framemobility, allowing users to maintain their TCP sessions while traveling, forexample, on a bus or a train. With sufficiently high upstream and downstream bitrates, the user could even maintain video-conferencing sessions while roamingabout. This scenario is not that far-fetched. As of 2012, many cellular telephonyproviders in the U.S. offer their subscribers a cellular Internet access service forunder $50 per month with typical downstream and upstream bit rates in the hun-dreds of kilobits per second. Data rates of several megabits per second arebecoming available as broadband data services such as those we will cover herebecome more widely deployed.In this section, we provide a brief overview of current and emerging cellularInternet access technologies. Our focus here will be on both the wireless first hop aswell as the network that connects the wireless first hop into the larger telephone net-work and/or the Internet; in Section 6.7 well consider how calls are routed to a usermoving between base stations. Our brief discussion will necessarily provide only asimplified and high-level description of cellular technologies. Modern cellular com-munications, of course, has great breadth and depth, with many universities offeringseveral courses on the topic. Readers seeking a deeper understanding are encouragedto see [Goodman 1997; Kaaranen 2001; Lin 2001; Korhonen 2003; Schiller 2003;Scourias 2012; Turner 2012; Akyildiz 2010], as well as the particularly excellent andexhaustive reference [Mouly 1992].6.4.1 An Overview of Cellular Network ArchitectureIn our description of cellular network architecture in this section, well adopt the ter-minology of the Global System for Mobile Communications (GSM) standards. (Forhistory buffs, the GSM acronym was originally derived from Groupe Spcial Mobile,until the more anglicized name was adopted, preserving the original acronym letters.)In the 1980s, Europeans recognized the need for a pan-European digital cellulartelephony system that would replace the numerous incompatible analog cellulartelephony systems, leading to the GSM standard [Mouly 1992]. Europeans deployedGSM technology with great success in the early 1990s, and since then GSM hasgrown to be the 800-pound gorilla of the cellular telephone world, with more than80% of all cellular subscribers worldwide using GSM.When people talk about cellular technology, they often classify the technologyas belonging to one of several generations. The earliest generations were designedprimarily for voice traffic. First generation (1G) systems were analog FDMA sys-tems designed exclusively for voice-only communication. These 1G systems arealmost extinct now, having been replaced by digital 2G systems. The original 2Gsystems were also designed for voice, but later extended (2.5G) to support data (i.e.,Internet) as well as voice service. The 3G systems that currently are being deployedalso support voice and data, but with an ever increasing emphasis on data capabili-ties and higher-speed radio access links.6.4 CELLULAR INTERNET ACCESS 547Cellular Network Architecture, 2G: Voice Connections to the Telephone NetworkThe term cellular refers to the fact that the region covered by a cellular networkis partitioned into a number of geographic coverage areas, known as cells, shownas hexagons on the left side of Figure 6.18. As with the 802.11WiFi standard westudied in Section 6.3.1, GSM has its own particular nomenclature. Each cellcontains a base transceiver station (BTS) that transmits signals to and receivessignals from the mobile stations in its cell. The coverage area of a cell depends548 CHAPTER 6 WIRELESS AND MOBILE NETWORKS3G CELLULAR MOBILE VERSUS WIRELESS LANSMany cellular mobile phone operators are deploying 3G cellular mobile systems withindoor data rates of 2 Mbps and outdoor data rates of 384 kbps and higher. These 3Gsystems are being deployed in licensed radio-frequency bands, with some operatorspaying considerable sums to governments for spectrum-use licenses. 3G systems allowusers to access the Internet from remote outdoor locations while on the move, in amanner similar to todays cellular phone access. For example, 3G technology permits auser to access road map information while driving a car, or movie theater informationwhile sunbathing on a beach. Nevertheless, one may question the extent to which 3Gsystems will be used, given their cost and the fact that users may often have simultane-ous access to both wireless LANs and 3G: The emerging wireless LAN infrastructure may become nearly ubiquitous. IEEE 802.11wireless LANs, operating at 54 Mbps, are enjoying widespread deployment. Almostall portable computers and smartphones are factory-equipped with 802.11 LANcapabilities. Furthermore, emerging Internet appliancessuch as wireless camerasand picture frameswill also have small and low-powered wireless LAN capabilities. Wireless LAN base stations can also handle mobile phone appliances. Manyphones are already capable of connecting to the cellular phone network or to anIP network either natively or using a Skype-like Voice-over-IP service, thus bypass-ing the operators cellular voice and 3G data services.Of course, many other experts believe that 3G not only will be a major success,but will also dramatically revolutionize the way we work and live. Most likely, both WiFi and 3G will both become prevalent wireless technologies, with roaming wireless devices automatically selecting the access technology that provides thebest service at their current physical location.CASE HISTORY6.4 CELLULAR INTERNET ACCESS 549on many factors, including the transmitting power of the BTS, the transmittingpower of the user devices, obstructing buildings in the cell, and the height of basestation antennas. Although Figure 6.18 shows each cell containing one basetransceiver station residing in the middle of the cell, many systems today placethe BTS at corners where three cells intersect, so that a single BTS with direc-tional antennas can service three cells.The GSM standard for 2G cellular systems uses combined FDM/TDM (radio)for the air interface. Recall from Chapter 1 that, with pure FDM, the channel is par-titioned into a number of frequency bands with each band devoted to a call. Alsorecall from Chapter 1 that, with pure TDM, time is partitioned into frames with eachframe further partitioned into slots and each call being assigned the use of a particu-lar slot in the revolving frame. In combined FDM/TDM systems, the channel is par-titioned into a number of frequency sub-bands; within each sub-band, time ispartitioned into frames and slots. Thus, for a combined FDM/TDM system, if the channel is partitioned into F sub-bands and time is partitioned into T slots, thenBSCBSCMSCKey: Base transceiver station(BTS)Base station controller(BSC)Mobile switching center(MSC)Mobile subscribersGatewayMSCBase Station System(BSS)Base Station System (BSS)Public telephonenetworkGFigure 6.18 Components of the GSM 2G cellular network architecturethe channel will be able to support F.T simultaneous calls. Recall that we saw in Section 5.3.4 that cable access networks also use a combined FDM/TDM approach.GSM systems consist of 200-kHz frequency bands with each band supporting eightTDM calls. GSM encodes speech at 13 kbps and 12.2 kbps.A GSM networks base station controller (BSC) will typically service severaltens of base transceiver stations. The role of the BSC is to allocate BTS radio chan-nels to mobile subscribers, perform paging (finding the cell in which a mobile useris resident), and perform handoff of mobile usersa topic well cover shortly inSection 6.7.2. The base station controller and its controlled base transceiver stationscollectively constitute a GSM base station system (BSS).As well see in Section 6.7, the mobile switching center (MSC) plays the cen-tral role in user authorization and accounting (e.g., determining whether a mobiledevice is allowed to connect to the cellular network), call establishment and tear-down, and handoff. A single MSC will typically contain up to five BSCs, resultingin approximately 200K subscribers per MSC. A cellular providers network willhave a number of MSCs, with special MSCs known as gateway MSCs connectingthe providers cellular network to the larger public telephone network.6.4.2 3G Cellular Data Networks: Extending the Internet toCellular SubscribersOur discussion in Section 6.4.1 focused on connecting cellular voice users to thepublic telephone network. But, of course, when were on the go, wed also like toread email, access the Web, get location-dependent services (e.g., maps and restau-rant recommendations) and perhaps even watch streaming video. To do this, oursmartphone will need to run a full TCP/IP protocol stack (including the physicallink, network, transport, and application layers) and connect into the Internet viathe cellular data network. The topic of cellular data networks is a rather bewilder-ing collection of competing and ever-evolving standards as one generation (andhalf-generation) succeeds the former and introduces new technologies and serviceswith new acronyms. To make matters worse, theres no single official body thatsets requirements for 2.5G, 3G, 3.5G, or 4G technologies, making it hard to sortout the differences among competing standards. In our discussion below, wellfocus on the UMTS (Universal Mobile Telecommunications Service) 3G standardsdeveloped by the 3rd Generation Partnership project (3GPP) [3GPP 2012], awidely deployed 3G technology.Lets take a top-down look at 3G cellular data network architecture shown inFigure 6.19.3G Core NetworkThe 3G core cellular data network connects radio access networks to the public Inter-net. The core network interoperates with components of the existing cellular voice550 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.4 CELLULAR INTERNET ACCESS 551GatewayMSCGKey:Serving GPRSSupport Node(SGSN)Gateway GPRSSupport Node(GGSN)Radio NetworkController (RNC) GGSNSGSNGGMSCPublic telephonenetworkRadio Interface(WCDMA, HSPA)Radio Access NetworkUniversal Terrestrial RadioAccess Network (UTRAN)Core NetworkGeneral Packet Radio Service(GPRS) Core NetworkPublicInternetPublicInternetFigure 6.19 3G system architecturenetwork (in particular, the MSC) that we previously encountered in Figure 6.18.Given the considerable amount of existing infrastructure (and profitable services!) inthe existing cellular voice network, the approach taken by the designers of 3G dataservices is clear: leave the existing core GSM cellular voice network untouched,adding additional cellular data functionality in parallel to the existing cellular voicenetwork. The alternativeintegrating new data services directly into the core of theexisting cellular voice networkwould have raised the same challenges encounteredin Section 4.4.4, where we discussed integrating new (IPv6) and legacy (IPv4) tech-nologies in the Internet.There are two types of nodes in the 3G core network: Serving GPRS Sup-port Nodes (SGSNs) and Gateway GPRS Support Nodes (GGSNs). (GPRSstands for Generalized Packet Radio Service, an early cellular data service in 2Gnetworks; here we discuss the evolved version of GPRS in 3G networks). AnSGSN is responsible for delivering datagrams to/from the mobile nodes in theradio access network to which the SGSN is attached. The SGSN interacts withthe cellular voice networks MSC for that area, providing user authorization andhandoff, maintaining location (cell) information about active mobile nodes, andperforming datagram forwarding between mobile nodes in the radio access net-work and a GGSN. The GGSN acts as a gateway, connecting multiple SGSNsinto the larger Internet. A GGSN is thus the last piece of 3G infrastructure that adatagram originating at a mobile node encounters before entering the larger Inter-net. To the outside world, the GGSN looks like any other gateway router; themobility of the 3G nodes within the GGSNs network is hidden from the outsideworld behind the GGSN.3G Radio Access Network: The Wireless EdgeThe 3G radio access network is the wireless first-hop network that we see as a3G user. The Radio Network Controller (RNC) typically controls several cellbase transceiver stations similar to the base stations that we encountered in 2Gsystems (but officially known in 3G UMTS parlance as a Node Bsa rathernon-descriptive name!). Each cells wireless link operates between the mobilenodes and a base transceiver station, just as in 2G networks. The RNC connectsto both the circuit-switched cellular voice network via an MSC, and to thepacket-switched Internet via an SGSN. Thus, while 3G cellular voice and cellu-lar data services use different core networks, they share a common first/last-hopradio access network.A significant change in 3G UMTS over 2G networks is that rather than usingGSMs FDMA/TDMA scheme, UMTS uses a CDMA technique known as DirectSequence Wideband CDMA (DS-WCDMA) [Dahlman 1998] within TDMAslots; TDMA slots, in turn, are available on multiple frequenciesan interestinguse of all three dedicated channel-sharing approaches that we earlier identified inChapter 5 and similar to the approach taken in wired cable access networks (seeSection 5.3.4). This change requires a new 3G cellular wireless-access networkoperating in parallel with the 2G BSS radio network shown in Figure 6.19. Thedata service associated with the WCDMA specification is known as HSP (HighSpeed Packet Access) and promises downlink data rates of up to 14 Mbps.Details regarding 3G networks can be found at the 3rd Generation PartnershipProject (3GPP) Web site [3GPP 2012].552 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.4.3 On to 4G: LTEWith 3G systems now being deployed worldwide, can 4G systems be far behind?Certainly not! Indeed, the design, early testing, and initial deployment of 4G sys-tems are already underway. The 4G Long-Term Evolution (LTE) standard put for-ward by the 3GPP has two important innovations over 3G systems: Evolved Packet Core (EPC) [3GPP Network Architecture 2012]. The EPC isa simplified all-IP core network that unifies the separate circuit-switched cel-lular voice network and the packet-switched cellular data network shown inFigure 6.19. It is an all-IP network in that both voice and data will be car-ried in IP datagrams. As weve seen in Chapter 4 and will study in more detailin Chapter 7, IPs best effort service model is not inherently well-suited tothe stringent performance requirements of Voice-over-IP (VoIP) traffic unlessnetwork resources are carefully managed to avoid (rather than react to) con-gestion. Thus, a key task of the EPC is to manage network resources to pro-vide this high quality of service. The EPC also makes a clear separationbetween the network control and user data planes, with many of the mobilitysupport features that we will study in Section 6.7 being implemented in thecontrol plane. The EPC allows multiple types of radio access networks,including legacy 2G and 3G radio access networks, to attach to the corenetwork. Two very readable introductions to the EPC are [Motorola 2007;Alcatel-Lucent 2009]. LTE Radio Access Network. LTE uses a combination of frequency divisionmultiplexing and time division multiplexing on the downstream channel,known as orthogonal frequency division multiplexing (OFDM) [Rohde 2008;Ericsson 2011]. (The term orthogonal comes from the fact the signals beingsent on different frequency channels are created so that they interfere very lit-tle with each other, even when channel frequencies are tightly spaced). In LTE,each active mobile node is allocated one or more 0.5 ms time slots in one ormore of the channel frequencies. Figure 6.20 shows an allocation of eight timeslots over four frequencies. By being allocated increasingly more time slots(whether on the same frequency or on different frequencies), a mobile node isable to achieve increasingly higher transmission rates. Slot (re)allocationamong mobile nodes can be performed as often as once every millisecond. Dif-ferent modulation schemes can also be used to change the transmission rate;see our earlier discussion of Figure 6.3 and dynamic selection of modulationschemes in WiFi networks. Another innovation in the LTE radio network is theuse of sophisticated multiple-input, multiple output (MIMO) antennas. Themaximum data rate for an LTE user is 100 Mbps in the downstream directionand 50 Mbps in the upstream direction, when using 20 MHz worth of wirelessspectrum.6.4 CELLULAR INTERNET ACCESS 553The particular allocation of time slots to mobile nodes is not mandated by theLTE standard. Instead, the decision of which mobile nodes will be allowed totransmit in a given time slot on a given frequency is determined by the schedul-ing algorithms provided by the LTE equipment vendor and/or the network opera-tor. With opportunistic scheduling [Bender 2000; Kolding 2003; Kulkarni 2005],matching the physical-layer protocol to the channel conditions between thesender and receiver and choosing the receivers to which packets will be sentbased on channel conditions allow the radio network controller to make best useof the wireless medium. In addition, user priorities and contracted levels of serv-ice (e.g., silver, gold, or platinum) can be used in scheduling downstream packettransmissions. In addition to the LTE capabilities described above, LTE-Advancedallows for downstream bandwidths of hundreds of Mbps by allocating aggregatedchannels to a mobile node [Akyildiz 2010].An additional 4G wireless technologyWiMAX (World Interoperability forMicrowave Access)is a family of IEEE 802.16 standards that differ significantlyfrom LTE. Whether LTE or WiMAX becomes the 4G technology of choice is still tobe seen, but at the time of this writing (spring 2012), LTE appears to have signifi-cantly more momentum. A detailed discussion of WiMAX can be found on thisbooks Web site.554 CHAPTER 6 WIRELESS AND MOBILE NETWORKSf1f2f3f4f5f60 0.5 1.0 1.5 2.0 2.5 9.0 9.5 10.0Figure 6.20 Twenty 0.5 ms slots organized into 10 ms frames at eachfrequency. An eight-slot allocation is shown shaded.6.5 Mobility Management: PrinciplesHaving covered the wireless nature of the communication links in a wireless net-work, its now time to turn our attention to the mobility that these wireless linksenable. In the broadest sense, a mobile node is one that changes its point of attach-ment into the network over time. Because the term mobility has taken on manymeanings in both the computer and telephony worlds, it will serve us well first toconsider several dimensions of mobility in some detail. From the network layers standpoint, how mobile is a user? A physically mobileuser will present a very different set of challenges to the network layer, depend-ing on how he or she moves between points of attachment to the network. At oneend of the spectrum in Figure 6.21, a user may carry a laptop with a wireless net-work interface card around in a building. As we saw in Section 6.3.4, this user isnot mobile from a network-layer perspective. Moreover, if the user associateswith the same access point regardless of location, the user is not even mobilefrom the perspective of the link layer.At the other end of the spectrum, consider the user zooming along the autobahnin a BMW at 150 kilometers per hour, passing through multiple wireless accessnetworks and wanting to maintain an uninterrupted TCP connection to a remoteapplication throughout the trip. This user is definitely mobile! In between theseextremes is a user who takes a laptop from one location (e.g., office or dormi-tory) into another (e.g., coffeeshop, classroom) and wants to connect into thenetwork in the new location. This user is also mobile (although less so than theBMW driver!) but does not need to maintain an ongoing connection while mov-ing between points of attachment to the network. Figure 6.21 illustrates this spec-trum of user mobility from the network layers perspective.6.5 MOBILITY MANAGEMENT: PRINCIPLES 555User moves onlywithin same wirelessaccess networkNo mobility High mobilityUser moves betweenaccess networks,shutting down whilemoving betweennetworksUser moves betweenaccess networks,while maintainingongoing connectionsFigure 6.21 Various degrees of mobility, from the network layers pointof view How important is it for the mobile nodes address to always remain the same?With mobile telephony, your phone numberessentially the network-layeraddress of your phoneremains the same as you travel from one providersmobile phone network to another. Must a laptop similarly maintain the same IPaddress while moving between IP networks?The answer to this question will depend strongly on the applications being run.For the BMW driver who wants to maintain an uninterrupted TCP connection toa remote application while zipping along the autobahn, it would be convenient tomaintain the same IP address. Recall from Chapter 3 that an Internet applicationneeds to know the IP address and port number of the remote entity with which itis communicating. If a mobile entity is able to maintain its IP address as itmoves, mobility becomes invisible from the application standpoint. There isgreat value to this transparencyan application need not be concerned with apotentially changing IP address, and the same application code serves mobileand nonmobile connections alike. Well see in the following section that mobileIP provides this transparency, allowing a mobile node to maintain its permanentIP address while moving among networks.On the other hand, a less glamorous mobile user might simply want to turn off anoffice laptop, bring that laptop home, power up, and work from home. If the laptopfunctions primarily as a client in client-server applications (e.g., send/read e-mail,browse the Web, Telnet to a remote host) from home, the particular IP address usedby the laptop is not that important. In particular, one could get by fine with anaddress that is temporarily allocated to the laptop by the ISP serving the home. Wesaw in Section 4.4 that DHCP already provides this functionality. What supporting wired infrastructure is available? In all of our scenarios above,weve implicitly assumed that there is a fixed infrastructure to which the mobileuser can connectfor example, the homes ISP network, the wireless access net-work in the office, or the wireless access networks lining the autobahn. What ifno such infrastructure exists? If two users are within communication proximityof each other, can they establish a network connection in the absence of any othernetwork-layer infrastructure? Ad hoc networking provides precisely these capa-bilities. This rapidly developing area is at the cutting edge of mobile networkingresearch and is beyond the scope of this book. [Perkins 2000] and the IETFMobile Ad Hoc Network (manet) working group Web pages [manet 2012] pro-vide thorough treatments of the subject.In order to illustrate the issues involved in allowing a mobile user to maintainongoing connections while moving between networks, lets consider a humananalogy. A twenty-something adult moving out of the family home becomesmobile, living in a series of dormitories and/or apartments, and often changingaddresses. If an old friend wants to get in touch, how can that friend find theaddress of her mobile friend? One common way is to contact the family, since a556 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.5 MOBILITY MANAGEMENT: PRINCIPLES 557mobile adult will often register his or her current address with the family (if for noother reason than so that the parents can send money to help pay the rent!). Thefamily home, with its permanent address, becomes that one place that others cango as a first step in communicating with the mobile adult. Later communicationfrom the friend may be either indirect (for example, with mail being sent first tothe parents home and then forwarded to the mobile adult) or direct (for example,with the friend using the address obtained from the parents to send mail directly toher mobile friend).In a network setting, the permanent home of a mobile node (such as a laptopor smartphone) is known as the home network, and the entity within the home network that performs the mobility management functions discussed below onbehalf of the mobile node is known as the home agent. The network in which the mobile node is currently residing is known as the foreign (or visited) network,and the entity within the foreign network that helps the mobile node with the mobility management functions discussed below is known as a foreign agent.For mobile professionals, their home network might likely be their company net-work, while the visited network might be the network of a colleague they are visit-ing. A correspondent is the entity wishing to communicate with the mobile node.Figure 6.22 illustrates these concepts, as well as addressing concepts consideredbelow. In Figure 6.22, note that agents are shown as being collocated with routers(e.g., as processes running on routers), but alternatively they could be executing onother hosts or servers in the network.6.5.1 AddressingWe noted above that in order for user mobility to be transparent to network applica-tions, it is desirable for a mobile node to keep its address as it moves from one net-work to another. When a mobile node is resident in a foreign network, all trafficaddressed to the nodes permanent address now needs to be routed to the foreignnetwork. How can this be done? One option is for the foreign network to advertiseto all other networks that the mobile node is resident in its network. This could bevia the usual exchange of intradomain and interdomain routing information andwould require few changes to the existing routing infrastructure. The foreign net-work could simply advertise to its neighbors that it has a highly specific route to themobile nodes permanent address (that is, essentially inform other networks that ithas the correct path for routing datagrams to the mobile nodes permanent address;see Section 4.4). These neighbors would then propagate this routing informationthroughout the network as part of the normal procedure of updating routing infor-mation and forwarding tables. When the mobile node leaves one foreign networkand joins another, the new foreign network would advertise a new, highly specificroute to the mobile node, and the old foreign network would withdraw its routinginformation regarding the mobile node.558 CHAPTER 6 WIRELESS AND MOBILE NETWORKSThis solves two problems at once, and it does so without making significantchanges to the network-layer infrastructure. Other networks know the location ofthe mobile node, and it is easy to route datagrams to the mobile node, since the for-warding tables will direct datagrams to the foreign network. A significant drawback,however, is that of scalability. If mobility management were to be the responsibilityof network routers, the routers would have to maintain forwarding table entries forpotentially millions of mobile nodes, and update these entries as nodes move. Someadditional drawbacks are explored in the problems at the end of this chapter.An alternative approach (and one that has been adopted in practice) is to pushmobility functionality from the network core to the network edgea recurringtheme in our study of Internet architecture. A natural way to do this is via the mobilenodes home network. In much the same way that parents of the mobile twenty-something track their childs location, the home agent in the mobile nodes homenetwork can track the foreign network in which the mobile node resides. A protocolHome agentHome network:128.119.40/24Visited network:79.129.13/24Mobile nodePermanent address:128.119.40.186 Permanent address:128.119.40.186Foreign agentCare-of address:79.129.13.2CorrespondentWide areanetworkFigure 6.22 Initial elements of a mobile network architecture6.5 MOBILITY MANAGEMENT: PRINCIPLES 559between the mobile node (or a foreign agent representing the mobile node) and thehome agent will certainly be needed to update the mobile nodes location.Lets now consider the foreign agent in more detail. The conceptually simplestapproach, shown in Figure 6.22, is to locate foreign agents at the edge routers in theforeign network. One role of the foreign agent is to create a so-called care-of address(COA) for the mobile node, with the network portion of the COA matching that ofthe foreign network. There are thus two addresses associated with a mobile node, itspermanent address (analogous to our mobile youths familys home address) and itsCOA, sometimes known as a foreign address (analogous to the address of the housein which our mobile youth is currently residing). In the example in Figure 6.22, thepermanent address of the mobile node is 128.119.40.186. When visiting network79.129.13/24, the mobile node has a COA of 79.129.13.2. A second role of the for-eign agent is to inform the home agent that the mobile node is resident in its (the for-eign agents) network and has the given COA. Well see shortly that the COA will beused to reroute datagrams to the mobile node via its foreign agent.Although we have separated the functionality of the mobile node and the for-eign agent, it is worth noting that the mobile node can also assume the responsibili-ties of the foreign agent. For example, the mobile node could obtain a COA in theforeign network (for example, using a protocol such as DHCP) and itself inform thehome agent of its COA.6.5.2 Routing to a Mobile NodeWe have now seen how a mobile node obtains a COA and how the home agent canbe informed of that address. But having the home agent know the COA solves onlypart of the problem. How should datagrams be addressed and forwarded to themobile node? Since only the home agent (and not network-wide routers) knows thelocation of the mobile node, it will no longer suffice to simply address a datagram tothe mobile nodes permanent address and send it into the network-layer infrastruc-ture. Something more must be done. Two approaches can be identified, which wewill refer to as indirect and direct routing.Indirect Routing to a Mobile NodeLets first consider a correspondent that wants to send a datagram to a mobile node.In the indirect routing approach, the correspondent simply addresses the datagramto the mobile nodes permanent address and sends the datagram into the network,blissfully unaware of whether the mobile node is resident in its home network or isvisiting a foreign network; mobility is thus completely transparent to the correspon-dent. Such datagrams are first routed, as usual, to the mobile nodes home network.This is illustrated in step 1 in Figure 6.23.Lets now turn our attention to the home agent. In addition to being responsiblefor interacting with a foreign agent to track the mobile nodes COA, the home agent560 CHAPTER 6 WIRELESS AND MOBILE NETWORKShas another very important function. Its second job is to be on the lookout for arriv-ing datagrams addressed to nodes whose home network is that of the home agent butthat are currently resident in a foreign network. The home agent intercepts thesedatagrams and then forwards them to a mobile node in a two-step process. The data-gram is first forwarded to the foreign agent, using the mobile nodes COA (step 2 inFigure 6.23), and then forwarded from the foreign agent to the mobile node (step 3in Figure 6.23).It is instructive to consider this rerouting in more detail. The home agent willneed to address the datagram using the mobile nodes COA, so that the networklayer will route the datagram to the foreign network. On the other hand, it isdesirable to leave the correspondents datagram intact, since the application receiv-ing the datagram should be unaware that the datagram was forwarded via the homeagent. Both goals can be satisfied by having the home agent encapsulate the corre-spondents original complete datagram within a new (larger) datagram. This largerHomeagentHome network:128.119.40/24Visited network:79.129.13/24Mobile nodePermanent address:128.119.40.186 Permanent address:128.119.40.186ForeignagentCare-ofaddress:79.129.13.2Wide areanetworkCorrespondent1243Figure 6.23 Indirect routing to a mobile node6.5 MOBILITY MANAGEMENT: PRINCIPLES 561datagram is addressed and delivered to the mobile nodes COA. The foreign agent,who owns the COA, will receive and decapsulate the datagramthat is, removethe correspondents original datagram from within the larger encapsulating data-gram and forward (step 3 in Figure 6.23) the original datagram to the mobile node.Figure 6.24 shows a correspondents original datagram being sent to the home net-work, an encapsulated datagram being sent to the foreign agent, and the originaldatagram being delivered to the mobile node. The sharp reader will note that theencapsulation/decapsulation described here is identical to the notion of tunneling,discussed in Chapter 4 in the context of IP multicast and IPv6.Lets next consider how a mobile node sends datagrams to a correspondent.This is quite simple, as the mobile node can address its datagram directly to the cor-respondent (using its own permanent address as the source address, and the corre-spondents address as the destination address). Since the mobile node knows thecorrespondents address, there is no need to route the datagram back through thehome agent. This is shown as step 4 in Figure 6.23.Lets summarize our discussion of indirect routing by listing the new network-layer functionality required to support mobility.HomeagentPermanent address:128.119.40.186 Permanent address:128.119.40.186ForeignagentCorrespondentdest: 128.119.40.186dest: 79.129.13.2 dest: 128.119.40.186dest: 128.119.40.186Care-of address:79.129.13.2Figure 6.24 Encapsulation and decapsulation A mobile-nodetoforeign-agent protocol. The mobile node will register withthe foreign agent when attaching to the foreign network. Similarly, a mobilenode will deregister with the foreign agent when it leaves the foreign network. A foreign-agenttohome-agent registration protocol. The foreign agent willregister the mobile nodes COA with the home agent. A foreign agent need notexplicitly deregister a COA when a mobile node leaves its network, because thesubsequent registration of a new COA, when the mobile node moves to a newnetwork, will take care of this. A home-agent datagram encapsulation protocol. Encapsulation and forward-ing of the correspondents original datagram within a datagram addressed tothe COA. A foreign-agent decapsulation protocol. Extraction of the correspondents origi-nal datagram from the encapsulating datagram, and the forwarding of the origi-nal datagram to the mobile node.The previous discussion provides all the piecesforeign agents, the homeagent, and indirect forwardingneeded for a mobile node to maintain an ongo-ing connection while moving among networks. As an example of how thesepieces fit together, assume the mobile node is attached to foreign network A, hasregistered a COA in network A with its home agent, and is receiving datagramsthat are being indirectly routed through its home agent. The mobile node nowmoves to foreign network B and registers with the foreign agent in network B,which informs the home agent of the mobile nodes new COA. From this pointon, the home agent will reroute datagrams to foreign network B. As far as a cor-respondent is concerned, mobility is transparentdatagrams are routed via thesame home agent both before and after the move. As far as the home agent is con-cerned, there is no disruption in the flow of datagramsarriving datagrams arefirst forwarded to foreign network A; after the change in COA, datagrams are for-warded to foreign network B. But will the mobile node see an interrupted flow ofdatagrams as it moves between networks? As long as the time between themobile nodes disconnection from network A (at which point it can no longerreceive datagrams via A) and its attachment to network B (at which point it willregister a new COA with its home agent) is small, few datagrams will be lost.Recall from Chapter 3 that end-to-end connections can suffer datagram loss dueto network congestion. Hence occasional datagram loss within a connectionwhen a node moves between networks is by no means a catastrophic problem. Ifloss-free communication is required, upper-layer mechanisms will recover fromdatagram loss, whether such loss results from network congestion or from usermobility.An indirect routing approach is used in the mobile IP standard [RFC 5944], asdiscussed in Section 6.6.562 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.5 MOBILITY MANAGEMENT: PRINCIPLES 563Direct Routing to a Mobile NodeThe indirect routing approach illustrated in Figure 6.23 suffers from an inefficiencyknown as the triangle routing problemdatagrams addressed to the mobile nodemust be routed first to the home agent and then to the foreign network, even when amuch more efficient route exists between the correspondent and the mobile node. Inthe worst case, imagine a mobile user who is visiting the foreign network of a col-league. The two are sitting side by side and exchanging data over the network. Data-grams from the correspondent (in this case the colleague of the visitor) are routed tothe mobile users home agent and then back again to the foreign network!Direct routing overcomes the inefficiency of triangle routing, but does so at thecost of additional complexity. In the direct routing approach, a correspondent agent inthe correspondents network first learns the COA of the mobile node. This can be doneby having the correspondent agent query the home agent, assuming that (as in the caseof indirect routing) the mobile node has an up-to-date value for its COA registered withits home agent. It is also possible for the correspondent itself to perform the function ofthe correspondent agent, just as a mobile node could perform the function of the for-eign agent. This is shown as steps 1 and 2 in Figure 6.25. The correspondent agent thentunnels datagrams directly to the mobile nodes COA, in a manner analogous to the tun-neling performed by the home agent, steps 3 and 4 in Figure 6.25.While direct routing overcomes the triangle routing problem, it introduces twoimportant additional challenges: A mobile-user location protocol is needed for the correspondent agent to querythe home agent to obtain the mobile nodes COA (steps 1 and 2 in Figure 6.25). When the mobile node moves from one foreign network to another, how will datanow be forwarded to the new foreign network? In the case of indirect routing, thisproblem was easily solved by updating the COA maintained by the homeagent. However, with direct routing, the home agent is queried for the COA bythe correspondent agent only once, at the beginning of the session. Thus, updat-ing the COA at the home agent, while necessary, will not be enough to solve theproblem of routing data to the mobile nodes new foreign network.One solution would be to create a new protocol to notify the correspondent ofthe changing COA. An alternate solution, and one that well see adopted in practice inGSM networks, works as follows. Suppose data is currently being forwarded to themobile node in the foreign network where the mobile node was located when the ses-sion first started (step 1 in Figure 6.26). Well identify the foreign agent in that for-eign network where the mobile node was first found as the anchor foreign agent.When the mobile node moves to a new foreign network (step 2 in Figure 6.26), themobile node registers with the new foreign agent (step 3), and the new foreign agentprovides the anchor foreign agent with the mobile nodes new COA (step 4). When564 CHAPTER 6 WIRELESS AND MOBILE NETWORKSHomeagentHome network:128.119.40/24Visited network:79.129.13/24Mobile nodePermanent address:128.119.40.186Key:Permanent address:128.119.40.186ForeignagentCare-of address:79.129.13.2Wide areanetworkCorrespondentControl messages Correspondentagent12 3Data flow 4Figure 6.25 Direct routing to a mobile userthe anchor foreign agent receives an encapsulated datagram for a departed mobilenode, it can then re-encapsulate the datagram and forward it to the mobile node (step 5) using the new COA. If the mobile node later moves yet again to a new for-eign network, the foreign agent in that new visited network would then contact theanchor foreign agent in order to set up forwarding to this new foreign network.6.6 Mobile IPThe Internet architecture and protocols for supporting mobility, collectively knownas mobile IP, are defined primarily in RFC 5944 for IPv4. Mobile IP is a flexiblestandard, supporting many different modes of operation (for example, operation6.6 MOBILE IP 565HomeagentHome network: Foreign networkbeing visited atsession start:New foreignnetwork:AnchorforeignagentNew foreign agentWide areanetworkCorrespondentCorrespondentagent14235Figure 6.26 Mobile transfer between networks with direct routingwith or without a foreign agent), multiple ways for agents and mobile nodes to dis-cover each other, use of single or multiple COAs, and multiple forms of encapsula-tion. As such, mobile IP is a complex standard, and would require an entire book todescribe in detail; indeed one such book is [Perkins 1998b]. Our modest goal here isto provide an overview of the most important aspects of mobile IP and to illustrateits use in a few common-case scenarios.The mobile IP architecture contains many of the elements we have consideredabove, including the concepts of home agents, foreign agents, care-of addresses, andencapsulation/decapsulation. The current standard [RFC 5944] specifies the use ofindirect routing to the mobile node.The mobile IP standard consists of three main pieces: Agent discovery. Mobile IP defines the protocols used by a home or foreign agentto advertise its services to mobile nodes, and protocols for mobile nodes to solicitthe services of a foreign or home agent.566 CHAPTER 6 WIRELESS AND MOBILE NETWORKS Registration with the home agent. Mobile IP defines the protocols used by themobile node and/or foreign agent to register and deregister COAs with a mobilenodes home agent. Indirect routing of datagrams. The standard also defines the manner in whichdatagrams are forwarded to mobile nodes by a home agent, including rules forforwarding datagrams, rules for handling error conditions, and several forms ofencapsulation [RFC 2003, RFC 2004].Security considerations are prominent throughout the mobile IP standard. Forexample, authentication of a mobile node is clearly needed to ensure that a mali-cious user does not register a bogus care-of address with a home agent, whichcould cause all datagrams addressed to an IP address to be redirected to the mali-cious user. Mobile IP achieves security using many of the mechanisms that we willexamine in Chapter 8, so we will not address security considerations in our discus-sion below.Agent DiscoveryA mobile IP node arriving to a new network, whether attaching to a foreign networkor returning to its home network, must learn the identity of the corresponding for-eign or home agent. Indeed it is the discovery of a new foreign agent, with a newnetwork address, that allows the network layer in a mobile node to learn that it hasmoved into a new foreign network. This process is known as agent discovery.Agent discovery can be accomplished in one of two ways: via agent advertisementor via agent solicitation.With agent advertisement, a foreign or home agent advertises its servicesusing an extension to the existing router discovery protocol [RFC 1256]. Theagent periodically broadcasts an ICMP message with a type field of 9 (router dis-covery) on all links to which it is connected. The router discovery message con-tains the IP address of the router (that is, the agent), thus allowing a mobile nodeto learn the agents IP address. The router discovery message also contains amobility agent advertisement extension that contains additional informationneeded by the mobile node. Among the more important fields in the extension arethe following: Home agent bit (H). Indicates that the agent is a home agent for the network inwhich it resides. Foreign agent bit (F). Indicates that the agent is a foreign agent for the networkin which it resides. Registration required bit (R). Indicates that a mobile user in this network mustregister with a foreign agent. In particular, a mobile user cannot obtain a care-of address in the foreign network (for example, using DHCP) and assume the6.6 MOBILE IP 567functionality of the foreign agent for itself, without registering with the for-eign agent. M, G encapsulation bits. Indicate whether a form of encapsulation other than IP-in-IP encapsulation will be used. Care-of address (COA) fields. A list of one or more care-of addresses providedby the foreign agent. In our example below, the COA will be associated with theforeign agent, who will receive datagrams sent to the COA and then forwardthem to the appropriate mobile node. The mobile user will select one of theseaddresses as its COA when registering with its home agent.Figure 6.27 illustrates some of the key fields in the agent advertisement message.With agent solicitation, a mobile node wanting to learn about agents withoutwaiting to receive an agent advertisement can broadcast an agent solicitation mes-sage, which is simply an ICMP message with type value 10. An agent receiving thesolicitation will unicast an agent advertisement directly to the mobile node, whichcan then proceed as if it had received an unsolicited advertisement.Type = 9 Code = 0Type = 16 Length Sequence numberRegistration lifetime ReservedRBHFMGrTbitsChecksumStandardICMP fields0 8 16 24Router address0 or more care-of addressesMobility agentadvertisementextensionFigure 6.27 ICMP router discovery message with mobility agent advertisement extension568 CHAPTER 6 WIRELESS AND MOBILE NETWORKSRegistration with the Home AgentOnce a mobile IP node has received a COA, that address must be registered with thehome agent. This can be done either via the foreign agent (who then registersthe COA with the home agent) or directly by the mobile IP node itself. We considerthe former case below. Four steps are involved.1. Following the receipt of a foreign agent advertisement, a mobile node sends amobile IP registration message to the foreign agent. The registration message iscarried within a UDP datagram and sent to port 434. The registration messagecarries a COA advertised by the foreign agent, the address of the home agent(HA), the permanent address of the mobile node (MA), the requested lifetimeof the registration, and a 64-bit registration identification. The requested regis-tration lifetime is the number of seconds that the registration is to be valid. Ifthe registration is not renewed at the home agent within the specified lifetime,the registration will become invalid. The registration identifier acts like asequence number and serves to match a received registration reply with a reg-istration request, as discussed below.2. The foreign agent receives the registration message and records the mobile nodespermanent IP address. The foreign agent now knows that it should be looking fordatagrams containing an encapsulated datagram whose destination addressmatches the permanent address of the mobile node. The foreign agent then sends amobile IP registration message (again, within a UDP datagram) to port 434 of thehome agent. The message contains the COA, HA, MA, encapsulation formatrequested, requested registration lifetime, and registration identification.3. The home agent receives the registration request and checks for authenticityand correctness. The home agent binds the mobile nodes permanent IP addresswith the COA; in the future, datagrams arriving at the home agent andaddressed to the mobile node will now be encapsulated and tunneled to theCOA. The home agent sends a mobile IP registration reply containing the HA,MA, actual registration lifetime, and the registration identification of therequest that is being satisfied with this reply.4. The foreign agent receives the registration reply and then forwards it to themobile node.At this point, registration is complete, and the mobile node can receive data-grams sent to its permanent address. Figure 6.28 illustrates these steps. Note that thehome agent specifies a lifetime that is smaller than the lifetime requested by themobile node.A foreign agent need not explicitly deregister a COA when a mobile nodeleaves its network. This will occur automatically, when the mobile node moves to anew network (whether another foreign network or its home network) and registers anew COA.6.6 MOBILE IP 569The mobile IP standard allows many additional scenarios and capabilities inaddition to those described previously. The interested reader should consult [Perkins1998b; RFC 5944].Home agentHA: 128.119.40.7Mobile agentMA: 128.119.40.186Visited network:79.129.13/24ICMP agent adv.COA: 79.129.13.2 . . . COA: 79.129.13.2 HA:128.119.40.7MA: 128.119.40.186 Lifetime: 9999 identification: 714 . . . Registration req.COA: 79.129.13.2 HA:128.119.40.7MA: 128.119.40.186 Lifetime: 9999 identification: 714 encapsulation format . . . Registration req.Time Time TimeHA: 128.119.40.7 MA: 128.119.40.186 Lifetime: 4999 identification: 714 encapsulation format . . . Registration replyHA: 128.119.40.7 MA: 128.119.40.186 Lifetime: 4999 identification: 714 . . . Registration replyForeign agentCOA: 79.129.13.2Figure 6.28 Agent advertisement and mobile IP registration570 CHAPTER 6 WIRELESS AND MOBILE NETWORKS6.7 Managing Mobility in Cellular NetworksHaving examined how mobility is managed in IP networks, lets now turn our atten-tion to networks with an even longer history of supporting mobilitycellulartelephony networks. Whereas we focused on the first-hop wireless link in cellularnetworks in Section 6.4, well focus here on mobility, using the GSM cellular net-work architecture [Goodman 1997; Mouly 1992; Scourias 2012; Kaaranen 2001;Korhonen 2003; Turner 2012] as our case study, since it is a mature and widelydeployed technology. As in the case of mobile IP, well see that a number of the fun-damental principles we identified in Section 6.5 are embodied in GSMs networkarchitecture.Like mobile IP, GSM adopts an indirect routing approach (see Section 6.5.2),first routing the correspondents call to the mobile users home network and fromthere to the visited network. In GSM terminology, the mobile userss home networkis referred to as the mobile users home public land mobile network (homePLMN). Since the PLMN acronym is a bit of a mouthful, and mindful of our questto avoid an alphabet soup of acronyms, well refer to the GSM home PLMN simplyas the home network. The home network is the cellular provider with which themobile user has a subscription (i.e., the provider that bills the user for monthly cel-lular service). The visited PLMN, which well refer to simply as the visited net-work, is the network in which the mobile user is currently residing.As in the case of mobile IP, the responsibilities of the home and visited net-works are quite different. The home network maintains a database known as the home location register(HLR), which contains the permanent cell phone number and subscriber pro-file information for each of its subscribers. Importantly, the HLR also con-tains information about the current locations of these subscribers. That is, if amobile user is currently roaming in another providers cellular network, theHLR contains enough information to obtain (via a process well describeshortly) an address in the visited network to which a call to the mobile usershould be routed. As well see, a special switch in the home network, knownas the Gateway Mobile services Switching Center (GMSC) is contacted bya correspondent when a call is placed to a mobile user. Again, in our quest toavoid an alphabet soup of acronyms, well refer to the GMSC here by a moredescriptive term, home MSC. The visited network maintains a database known as the visitor location register(VLR). The VLR contains an entry for each mobile user that is currently in theportion of the network served by the VLR. VLR entries thus come and go asmobile users enter and leave the network. A VLR is usually co-located with themobile switching center (MSC) that coordinates the setup of a call to and fromthe visited network.6.7 MANAGING MOBILITY IN CELLULAR NETWORKS 571In practice, a providers cellular network will serve as a home network for its sub-scribers and as a visited network for mobile users whose subscription is with a dif-ferent cellular provider.6.7.1 Routing Calls to a Mobile UserWere now in a position to describe how a call is placed to a mobile GSM user in avisited network. Well consider a simple example below; more complex scenarios aredescribed in [Mouly 1992]. The steps, as illustrated in Figure 6.29, are as follows:1. The correspondent dials the mobile users phone number. This number itselfdoes not refer to a particular telephone line or location (after all, the phonenumber is fixed and the user is mobile!). The leading digits in the number aresufficient to globally identify the mobiles home network. The call is routedfrom the correspondent through the PSTN to the home MSC in the mobileshome network. This is the first leg of the call.2. The home MSC receives the call and interrogates the HLR to determine thelocation of the mobile user. In the simplest case, the HLR returns the mobileMobileuserVisitednetworkHomenetworkPublic switched telephonenetwork13CorrespondentVLRHLR2Figure 6.29 Placing a call to a mobile user: indirect routing572 CHAPTER 6 WIRELESS AND MOBILE NETWORKSstation roaming number (MSRN), which we will refer to as the roamingnumber. Note that this number is different from the mobiles permanent phonenumber, which is associated with the mobiles home network. The roamingnumber is ephemeral: It is temporarily assigned to a mobile when it enters a vis-ited network. The roaming number serves a role similar to that of the care-ofaddress in mobile IP and, like the COA, is invisible to the correspondent and themobile. If HLR does not have the roaming number, it returns the address of theVLR in the visited network. In this case (not shown in Figure 6.29), the homeMSC will need to query the VLR to obtain the roaming number of the mobilenode. But how does the HLR get the roaming number or the VLR address in thefirst place? What happens to these values when the mobile user moves toanother visited network? Well consider these important questions shortly.3. Given the roaming number, the home MSC sets up the second leg of the callthrough the network to the MSC in the visited network. The call is completed,being routed from the correspondent to the home MSC, and from there to thevisited MSC, and from there to the base station serving the mobile user.An unresolved question in step 2 is how the HLR obtains information aboutthe location of the mobile user. When a mobile telephone is switched on or enters apart of a visited network that is covered by a new VLR, the mobile must registerwith the visited network. This is done through the exchange of signaling messagesbetween the mobile and the VLR. The visited VLR, in turn, sends a location updaterequest message to the mobiles HLR. This message informs the HLR of either theroaming number at which the mobile can be contacted, or the address of the VLR(which can then later be queried to obtain the mobile number). As part of thisexchange, the VLR also obtains subscriber information from the HLR about themobile and determines what services (if any) should be accorded the mobile userby the visited network.6.7.2 Handoffs in GSMA handoff occurs when a mobile station changes its association from one base sta-tion to another during a call. As shown in Figure 6.30, a mobiles call is initially(before handoff) routed to the mobile through one base station (which well refer toas the old base station), and after handoff is routed to the mobile through anotherbase station (which well refer to as the new base station). Note that a handoffbetween base stations results not only in the mobile transmitting/receiving to/from anew base station, but also in the rerouting of the ongoing call from a switching pointwithin the network to the new base station. Lets initially assume that the old andnew base stations share the same MSC, and that the rerouting occurs at this MSC.There may be several reasons for handoff to occur, including (1) the signalbetween the current base station and the mobile may have deteriorated to such anextent that the call is in danger of being dropped, and (2) a cell may have become6.7 MANAGING MOBILITY IN CELLULAR NETWORKS 573overloaded, handling a large number of calls. This congestion may be alleviated byhanding off mobiles to less congested nearby cells.While it is associated with a base station, a mobile periodically measures thestrength of a beacon signal from its current base station as well as beacon signals fromnearby base stations that it can hear. These measurements are reported once or twicea second to the mobiles current base station. Handoff in GSM is initiated by the oldbase station based on these measurements, the current loads of mobiles in nearby cells,and other factors [Mouly 1992]. The GSM standard does not specify the specific algo-rithm to be used by a base station to determine whether or not to perform handoff.Figure 6.31 illustrates the steps involved when a base station does decide tohand off a mobile user:1. The old base station (BS) informs the visited MSC that a handoff is to be per-formed and the BS (or possible set of BSs) to which the mobile is to be handed off.2. The visited MSC initiates path setup to the new BS, allocating the resourcesneeded to carry the rerouted call, and signaling the new BS that a handoff isabout to occur.3. The new BS allocates and activates a radio channel for use by the mobile.4. The new BS signals back to the visited MSC and the old BS that the visited-MSC-to-new-BS path has been established and that the mobile should beinformed of the impending handoff. The new BS provides all of the informa-tion that the mobile will need to associate with the new BS.5. The mobile is informed that it should perform a handoff. Note that up until thispoint, the mobile has been blissfully unaware that the network has been layingthe groundwork (e.g., allocating a channel in the new BS and allocating a pathfrom the visited MSC to the new BS) for a handoff.6. The mobile and the new BS exchange one or more messages to fully activatethe new channel in the new BS.Old BS New BSOldroutingNewroutingVLRFigure 6.30 Handoff scenario between base stations with a commonMSC574 CHAPTER 6 WIRELESS AND MOBILE NETWORKS7. The mobile sends a handoff complete message to the new BS, which is for-warded up to the visited MSC. The visited MSC then reroutes the ongoing callto the mobile via the new BS.8. The resources allocated along the path to the old BS are then released.Lets conclude our discussion of handoff by considering what happens when themobile moves to a BS that is associated with a different MSC than the old BS, and whathappens when this inter-MSC handoff occurs more than once. As shown in Figure 6.32,GSM defines the notion of an anchor MSC. The anchor MSC is the MSC visited bythe mobile when a call first begins; the anchor MSC thus remains unchanged duringthe call. Throughout the calls duration and regardless of the number of inter-MSCtransfers performed by the mobile, the call is routed from the home MSC to the anchorMSC, and then from the anchor MSC to the visited MSC where the mobile is currentlylocated. When a mobile moves from the coverage area of one MSC to another, theongoing call is rerouted from the anchor MSC to the new visited MSC containing thenew base station. Thus, at all times there are at most three MSCs (the home MSC, theanchor MSC, and the visited MSC) between the correspondent and the mobile. Figure6.32 illustrates the routing of a call among the MSCs visited by a mobile user.Rather than maintaining a single MSC hop from the anchor MSC to the currentMSC, an alternative approach would have been to simply chain the MSCs visited bythe mobile, having an old MSC forward the ongoing call to the new MSC each timethe mobile moves to a new MSC. Such MSC chaining can in fact occur in IS-41 cel-lular networks, with an optional path minimization step to remove MSCs between theanchor MSC and the current visited MSC [Lin 2001].Lets wrap up our discussion of GSM mobility management with a comparisonof mobility management in GSM and Mobile IP. The comparison in Table 6.2 indi-cates that although IP and cellular networks are fundamentally different in manyways, they share a surprising number of common functional elements and overallapproaches in handling mobility.OldBSNewBS15782364VLRFigure 6.31 Steps in accomplishing a handoff between base stationswith a common MSC6.8 WIRELESS AND MOBILITY: IMPACT ON HIGHER-LAYER PROTOCOLS 5756.8 Wireless and Mobility: Impact on Higher-Layer ProtocolsIn this chapter, weve seen that wireless networks differ significantly from their wiredcounterparts at both the link layer (as a result of wireless channel characteristics such asfading, multipath, and hidden terminals) and at the network layer (as a result of mobileusers who change their points of attachment to the network). But are there important dif-ferences at the transport and application layers? Its tempting to think that these differ-ences will be minor, since the network layer provides the same best-effort deliveryservice model to upper layers in both wired and wireless networks. Similarly, if proto-cols such as TCP or UDP are used to provide transport-layer services to applications inboth wired and wireless networks, then the application layer should remain unchanged aswell. In one sense our intuition is rightTCP and UDP can (and do) operate in networkswith wireless links. On the other hand, transport protocols in general, and TCP in partic-ular, can sometimes have very different performance in wired and wireless networks,and it is here, in terms of performance, that differences are manifested. Lets see why.Recall that TCP retransmits a segment that is either lost or corrupted on the pathbetween sender and receiver. In the case of mobile users, loss can result from eitherHome network Correspondenta.Before handoffAnchorMSCPSTNb.After handoffCorrespondentAnchorMSCPSTNHome networkFigure 6.32 Rerouting via the anchor MSC576 CHAPTER 6 WIRELESS AND MOBILE NETWORKSnetwork congestion (router buffer overflow) or from handoff (e.g., from delays inrerouting segments to a mobiles new point of attachment to the network). In allcases, TCPs receiver-to-sender ACK indicates only that a segment was not receivedintact; the sender is unaware of whether the segment was lost due to congestion,during handoff, or due to detected bit errors. In all cases, the senders response isthe sameto retransmit the segment. TCPs congestion-control response is also thesame in all casesTCP decreases its congestion window, as discussed in Section3.7. By unconditionally decreasing its congestion window, TCP implicitly assumesthat segment loss results from congestion rather than corruption or handoff. We sawin Section 6.2 that bit errors are much more common in wireless networks than inwired networks. When such bit errors occur or when handoff loss occurs, theresreally no reason for the TCP sender to decrease its congestion window (and thusdecrease its sending rate). Indeed, it may well be the case that router buffers areempty and packets are flowing along the end-to-end path unimpeded by congestion.Researchers realized in the early to mid 1990s that given high bit error rates onwireless links and the possibility of handoff loss, TCPs congestion-control responsecould be problematic in a wireless setting. Three broad classes of approaches arepossible for dealing with this problem: Local recovery. Local recovery protocols recover from bit errors when and where(e.g., at the wireless link) they occur, e.g., the 802.11 ARQ protocol we studiedGSM element Comment on GSM element Mobile IP elementHome system Network to which the mobile users permanent phone Home networknumber belongs.Gateway mobile switching center or Home MSC: point of contact to obtain routable address of Home agentsimply home MSC, Home mobile user. HLR: database in home system containing permanent location register (HLR) phone number, profile information, current location of mobile user,subscription information.Visited system Network other than home system where mobile user is currently residing. Visited network.Visited mobile services switching center, Visited MSC: responsible for setting up calls to/from mobile nodes Foreign agentVisitor location register (VLR) in cells associated with MSC. VLR: temporary database entry in visited system, containing subscription information for each visiting mobile user.Mobile station roaming number Routable address for telephone call segment between home MSC Care-of address(MSRN) or simply roaming number and visited MSC, visible to neither the mobile nor the correspondent.Table 6.2 Commonalities between mobile IP and GSM mobility6.8 WIRELESS AND MOBILITY: IMPACT ON HIGHER-LAYER PROTOCOLS 577in Section 6.3, or more sophisticated approaches that use both ARQ and FEC[Ayanoglu 1995]. TCP sender awareness of wireless links. In the local recovery approaches, theTCP sender is blissfully unaware that its segments are traversing a wireless link.An alternative approach is for the TCP sender and receiver to be aware of theexistence of a wireless link, to distinguish between congestive losses occurring inthe wired network and corruption/loss occurring at the wireless link, and toinvoke congestion control only in response to congestive wired-network losses.[Balakrishnan 1997] investigates various types of TCP, assuming that end systemscan make this distinction. [Liu 2003] investigates techniques for distinguishingbetween losses on the wired and wireless segments of an end-to-end path. Split-connection approaches. In a split-connection approach [Bakre 1995], theend-to-end connection between the mobile user and the other end point is brokeninto two transport-layer connections: one from the mobile host to the wirelessaccess point, and one from the wireless access point to the other communicationend point (which well assume here is a wired host). The end-to-end connectionis thus formed by the concatenation of a wireless part and a wired part. The trans-port layer over the wireless segment can be a standard TCP connection [Bakre1995], or a specially tailored error recovery protocol on top of UDP. [Yavatkar1994] investigates the use of a transport-layer selective repeat protocol over thewireless connection. Measurements reported in [Wei 2006] indicate that splitTCP connections are widely used in cellular data networks, and that significantimprovements can indeed be made through the use of split TCP connections.Our treatment of TCP over wireless links has been necessarily brief here. In-depth surveys of TCP challenges and solutions in wireless networks can be found in[Hanabali 2005; Leung 2006]. We encourage you to consult the references fordetails of this ongoing area of research.Having considered transport-layer protocols, let us next consider the effect ofwireless and mobility on application-layer protocols. Here, an important considera-tion is that wireless links often have relatively low bandwidths, as we saw in Figure6.2. As a result, applications that operate over wireless links, particularly over cellu-lar wireless links, must treat bandwidth as a scarce commodity. For example, a Webserver serving content to a Web browser executing on a 3G phone will likely not beable to provide the same image-rich content that it gives to a browser operating overa wired connection. Although wireless links do provide challenges at the applicationlayer, the mobility they enable also makes possible a rich set of location-aware andcontext-aware applications [Chen 2000; Baldauf 2007]. More generally, wirelessand mobile networks will play a key role in realizing the ubiquitous computing envi-ronments of the future [Weiser 1991]. Its fair to say that weve only seen the tip ofthe iceberg when it comes to the impact of wireless and mobile networks on net-worked applications and their protocols!6.9 SummaryWireless and mobile networks have revolutionized telephony and are having anincreasingly profound impact in the world of computer networks as well. With theiranytime, anywhere, untethered access into the global network infrastructure, theyare not only making network access more ubiquitous, they are also enabling anexciting new set of location-dependent services. Given the growing importance ofwireless and mobile networks, this chapter has focused on the principles, commonlink technologies, and network architectures for supporting wireless and mobilecommunication.We began this chapter with an introduction to wireless and mobile networks,drawing an important distinction between the challenges posed by the wirelessnature of the communication links in such networks, and by the mobility that thesewireless links enable. This allowed us to better isolate, identify, and master the keyconcepts in each area. We focused first on wireless communication, considering thecharacteristics of a wireless link in Section 6.2. In Sections 6.3 and 6.4, we exam-ined the link-level aspects of the IEEE 802.11 (WiFi) wireless LAN standard, twoIEEE 802.15 personal area networks (Bluetooth and Zigbee), and 3G and 4G cellu-lar Internet access. We then turned our attention to the issue of mobility. In Section6.5, we identified several forms of mobility, with points along this spectrum posingdifferent challenges and admitting different solutions. We considered the problemsof locating and routing to a mobile user, as well as approaches for handing off themobile user who dynamically moves from one point of attachment to the network toanother. We examined how these issues were addressed in the mobile IP standardand in GSM, in Sections 6.6 and 6.7, respectively. Finally, we considered the impactof wireless links and mobility on transport-layer protocols and networked applica-tions in Section 6.8.Although we have devoted an entire chapter to the study of wireless and mobilenetworks, an entire book (or more) would be required to fully explore this excitingand rapidly expanding field. We encourage you to delve more deeply into this fieldby consulting the many references provided in this chapter.Homework Problems and QuestionsChapter 6 Review QuestionsSECTION 6.1R1. What does it mean for a wireless network to be operating in infrastructuremode? If the network is not in infrastructure mode, what mode of operationis it in, and what is the difference between that mode of operation and infra-structure mode?578 CHAPTER 6 WIRELESS AND MOBILE NETWORKSR2. What are the four types of wireless networks identified in our taxonomy inSection 6.1? Which of these types of wireless networks have you used?SECTION 6.2R3. What are the differences between the following types of wireless channelimpairments: path loss, multipath propagation, interference from othersources?R4. As a mobile node gets farther and farther away from a base station, what aretwo actions that a base station could take to ensure that the loss probability ofa transmitted frame does not increase?SECTIONS 6.3 AND 6.4R5. Describe the role of the beacon frames in 802.11.R6. True or false: Before an 802.11 station transmits a data frame, it must firstsend an RTS frame and receive a corresponding CTS frame.R7. Why are acknowledgments used in 802.11 but not in wired Ethernet?R8. True or false: Ethernet and 802.11 use the same frame structure.R9. Describe how the RTS threshold works.R10. Suppose the IEEE 802.11 RTS and CTS frames were as long as the standardDATA and ACK frames. Would there be any advantage to using the CTS andRTS frames? Why or why not?R11. Section 6.3.4 discusses 802.11 mobility, in which a wireless station movesfrom one BSS to another within the same subnet. When the APs are intercon-nected with a switch, an AP may need to send a frame with a spoofed MACaddress to get the switch to forward the frame properly. Why?R12. What are the differences between a master device in a Bluetooth network anda base station in an 802.11 network?R13. What is meant by a super frame in the 802.15.4 Zigbee standard?R14. What is the role of the core network in the 3G cellular data architecture?R15. What is the role of the RNC in the 3G cellular data network architecture?What role does the RNC play in the cellular voice network?SECTIONS 6.5 AND 6.6R16. If a node has a wireless connection to the Internet, does that node have to bemobile? Explain. Suppose that a user with a laptop walks around her housewith her laptop, and always accesses the Internet through the same accesspoint. Is this user mobile from a network standpoint? Explain.R17. What is the difference between a permanent address and a care-of address?Who assigns a care-of address?HOMEWORK PROBLEMS AND QUESTIONS 579R18. Consider a TCP connection going over Mobile IP. True or false: The TCPconnection phase between the correspondent and the mobile host goesthrough the mobiles home network, but the data transfer phase is directlybetween the correspondent and the mobile host, bypassing the home network.SECTION 6.7R19. What are the purposes of the HLR and VLR in GSM networks? What ele-ments of mobile IP are similar to the HLR and VLR?R20. What is the role of the anchor MSC in GSM networks?SECTION 6.8R21. What are three approaches that can be taken to avoid having a single wirelesslink degrade the performance of an end-to-end transport-layer TCP connection?ProblemsP1. Consider the single-sender CDMA example in Figure 6.5. What would be thesenders output (for the 2 data bits shown) if the senders CDMA code were(1, 1, 1, 1, 1, 1, 1, 1)?P2. Consider sender 2 in Figure 6.6. What is the senders output to the channel(before it is added to the signal from sender 1), Z2i,m?P3. Suppose that the receiver in Figure 6.6 wanted to receive the data being sentby sender 2. Show (by calculation) that the receiver is indeed able to recoversender 2s data from the aggregate channel signal by using sender 2s code.P4. For the two-sender, two-receiver example, give an example of two CDMAcodes containing 1 and 1 values that do not allow the two receivers toextract the original transmitted bits from the two CDMA senders.P5. Suppose there are two ISPs providing WiFi access in a particular caf, witheach ISP operating its own AP and having its own IP address block.a. Further suppose that by accident, each ISP has configured its AP to oper-ate over channel 11. Will the 802.11 protocol completely break down inthis situation? Discuss what happens when two stations, each associatedwith a different ISP, attempt to transmit at the same time.b. Now suppose that one AP operates over channel 1 and the other overchannel 11. How do your answers change?P6. In step 4 of the CSMA/CA protocol, a station that successfully transmits aframe begins the CSMA/CA protocol for a second frame at step 2, rather thanat step 1. What rationale might the designers of CSMA/CA have had in mindby having such a station not transmit the second frame immediately (if thechannel is sensed idle)?580 CHAPTER 6 WIRELESS AND MOBILE NETWORKSP7. Suppose an 802.11b station is configured to always reserve the channel with theRTS/CTS sequence. Suppose this station suddenly wants to transmit 1,000 bytesof data, and all other stations are idle at this time. As a function of SIFS andDIFS, and ignoring propagation delay and assuming no bit errors, calculate thetime required to transmit the frame and receive the acknowledgment.P8. Consider the scenario shown in Figure 6.33, in which there are four wirelessnodes, A, B, C, and D. The radio coverage of the four nodes is shown via theshaded ovals; all nodes share the same frequency. When A transmits, it canonly be heard/received by B; when B transmits, both A and C canhear/receive from B; when C transmits, both B and D can hear/receive fromC; when D transmits, only C can hear/receive from D.Suppose now that each node has an infinite supply of messages that it wantsto send to each of the other nodes. If a messages destination is not an imme-diate neighbor, then the message must be relayed. For example, if A wants tosend to D, a message from A must first be sent to B, which then sends themessage to C, which then sends the message to D. Time is slotted, with amessage transmission time taking exactly one time slot, e.g., as in slottedAloha. During a slot, a node can do one of the following: (i) send a message;(ii) receive a message (if exactly one message is being sent to it), (iii) remainsilent. As always, if a node hears two or more simultaneous transmissions, acollision occurs and none of the transmitted messages are received success-fully. You can assume here that there are no bit-level errors, and thus ifexactly one message is sent, it will be received correctly by those within thetransmission radius of the sender.a. Suppose now that an omniscient controller (i.e., a controller that knowsthe state of every node in the network) can command each node to dowhatever it (the omniscient controller) wishes, i.e., to send a message, toreceive a message, or to remain silent. Given this omniscient controller,what is the maximum rate at which a data message can be transferredfrom C to A, given that there are no other messages between any othersource/destination pairs?PROBLEMS 581A B C DFigure 6.33 Scenario for problem P8b. Suppose now that A sends messages to B, and D sends messages to C.What is the combined maximum rate at which data messages can flowfrom A to B and from D to C?c. Suppose now that A sends messages to B, and C sends messages to D.What is the combined maximum rate at which data messages can flowfrom A to B and from C to D?d. Suppose now that the wireless links are replaced by wired links. Repeatquestions (a) through (c) again in this wired scenario.e. Now suppose we are again in the wireless scenario, and that for every datamessage sent from source to destination, the destination will send an ACKmessage back to the source (e.g., as in TCP). Also suppose that each ACKmessage takes up one slot. Repeat questions (a) (c) above for this scenario.P9. Describe the format of the 802.15.1 Bluetooth frame. You will have to dosome reading outside of the text to find this information. Is there anything inthe frame format that inherently limits the number of active nodes in an802.15.1 network to eight active nodes? Explain.P10. Consider the following idealized LTE scenario. The downstream channel(see Figure 6.20) is slotted in time, across F frequencies. There are fournodes, A, B, C, and D, reachable from the base station at rates of 10 Mbps,5 Mbps, 2.5 Mbps, and 1 Mbps, respectively, on the downstream channel.These rates assume that the base station utilizes all time slots available onall F frequencies to send to just one station. The base station has an infiniteamount of data to send to each of the nodes, and can send to any one ofthese four nodes using any of the F frequencies during any time slot in thedownstream sub-frame.a. What is the maximum rate at which the base station can send to the nodes,assuming it can send to any node it chooses during each time slot? Is yoursolution fair? Explain and define what you mean by fair.b. If there is a fairness requirement that each node must receive an equalamount of data during each one second interval, what is the average transmission rate by the base station (to all nodes) during the downstreamsub-frame? Explain how you arrived at your answer.c. Suppose that the fairness criterion is that any node can receive at mosttwice as much data as any other node during the sub-frame. What is theaverage transmission rate by the base station (to all nodes) during the sub-frame? Explain how you arrived at your answer.P11. In Section 6.5, one proposed solution that allowed mobile users to maintaintheir IP addresses as they moved among foreign networks was to have a for-eign network advertise a highly specific route to the mobile user and use theexisting routing infrastructure to propagate this information throughout the582 CHAPTER 6 WIRELESS AND MOBILE NETWORKSnetwork. We identified scalability as one concern. Suppose that when amobile user moves from one network to another, the new foreign networkadvertises a specific route to the mobile user, and the old foreign networkwithdraws its route. Consider how routing information propagates in adistance-vector algorithm (particularly for the case of interdomain routingamong networks that span the globe).a. Will other routers be able to route datagrams immediately to the new for-eign network as soon as the foreign network begins advertising its route?b. Is it possible for different routers to believe that different foreign networkscontain the mobile user?c. Discuss the timescale over which other routers in the network will eventu-ally learn the path to the mobile users.P12. Suppose the correspondent in Figure 6.22 were mobile. Sketch the additionalnetwork-layer infrastructure that would be needed to route the datagram fromthe original mobile user to the (now mobile) correspondent. Show the struc-ture of the datagram(s) between the original mobile user and the (nowmobile) correspondent, as in Figure 6.23.P13. In mobile IP, what effect will mobility have on end-to-end delays of data-grams between the source and destination?P14. Consider the chaining example discussed at the end of Section 6.7.2. Supposea mobile user visits foreign networks A, B, and C, and that a correspondentbegins a connection to the mobile user when it is resident in foreign network A.List the sequence of messages between foreign agents, and between foreignagents and the home agent as the mobile user moves from network A to net-work B to network C. Next, suppose chaining is not performed, and the cor-respondent (as well as the home agent) must be explicitly notified of thechanges in the mobile users care-of address. List the sequence of messagesthat would need to be exchanged in this second scenario.P15. Consider two mobile nodes in a foreign network having a foreign agent. Is itpossible for the two mobile nodes to use the same care-of address in mobileIP? Explain your answer.P16. In our discussion of how the VLR updated the HLR with information aboutthe mobiles current location, what are the advantages and disadvantages ofproviding the MSRN as opposed to the address of the VLR to the HLR?Wireshark LabAt the companion Web site for this textbook, http://www.awl.com/kurose-ross,youll find a Wireshark lab for this chapter that captures and studies the 802.11frames exchanged between a wireless laptop and an access point.WIRESHARK LAB 583http://www.awl.com/kurose-rossPlease describe a few of the most exciting projects you have worked on during yourcareer. What were the biggest challenges?In the mid-90s at USC and ISI, I had the great fortune to work with the likes of SteveDeering, Mark Handley, and Van Jacobson on the design of multicast routing protocols (inparticular, PIM). I tried to carry many of the architectural design lessons from multicast intothe design of ecological monitoring arrays, where for the first time I really began to takeapplications and multidisciplinary research seriously. That interest in jointly innovating inthe social and technological space is what interests me so much about my latest area ofresearch, mobile health. The challenges in these projects were as diverse as the problemdomains, but what they all had in common was the need to keep our eyes open to whetherwe had the problem definition right as we iterated between design and deployment, proto-type and pilot. None of them were problems that could be solved analytically, with simula-tion or even in constructed laboratory experiments. They all challenged our ability to retain584Deborah EstrinDeborah Estrin is Professor of Computer Science at UCLA, the Jon Postel Chair in Computer Networks, Director of the Center forEmbedded Networked Sensing (CENS), and co-founder of the non-profit openmhealth.org. She received her Ph.D. (1985) in ComputerScience from M.I.T., and her B.S. (1980) from UC Berkeley. Estrinsearly research focused on the design of network protocols, includingmulticast and inter-domain routing. In 2002 Estrin founded the NSF-funded Science and Technology Center, CENS (http://cens.ucla.edu), to develop and explore environmental monitoring technologies and applications.Currently Estrin and collaborators are developing participatory sensing systems, leveragingthe programmability, proximity, and pervasiveness of mobile phones; the primary deploymentcontexts are mobile health (http://openmhealth.org), community data gathering, and STEMeducation (http://mobilizingcs.org). Professor Estrin is an elected member of the AmericanAcademy of Arts and Sciences (2007) and the National Academy of Engineering (2009).She is a fellow of the IEEE, ACM, and AAAS. She was selected as the first ACM-W AthenaLecturer (2006), awarded the Anita Borg Institutes Women of Vision Award for Innovation(2007), inducted into the WITI hall of fame (2008) and awarded Doctor Honoris Causa fromEPFL (2008) and Uppsala University (2011).AN INTERVIEW WITH…http://cens.ucla.eduhttp://cens.ucla.eduhttp://openmhealth.orghttp://mobilizingcs.orgclean architectures in the presence of messy problems and contexts, and they all called forextensive collaboration.What changes and innovations do you see happening in wireless networks and mobility in the future?I have never put much faith into predicting the future, but I would say we might see the endof feature phones (i.e., those that are not programmable and are used only for voice and textmessaging) as smart phones become more and more powerful and the primary point ofInternet access for many. I also think that we will see the continued proliferation of embed-ded SIMs by which all sorts of devices have the ability to communicate via the cellular net-work at low data rates.Where do you see the future of networking and the Internet?The efforts in named data and software-defined networking will emerge to create a moremanageable, evolvable, and richer infrastructure and more generally represent moving therole of architecture higher up in the stack. In the beginnings of the Internet, architecture waslayer 4 and below, with applications being more siloed/monolithic, sitting on top. Now dataand analytics dominate transport.What people inspired you professionally?There are three people who come to mind. First, Dave Clark, the secret sauce and unsunghero of the Internet community. I was lucky to be around in the early days to see him act asthe organizing principle of the IAB and Internet governance; the priest of rough consen-sus and running code. Second, Scott Shenker, for his intellectual brilliance, integrity, andpersistence. I strive for, but rarely attain, his clarity in defining problems and solutions. Heis always the first person I email for advice on matters large and small. Third, my sisterJudy Estrin, who had the creativity and courage to spend her career bringing ideas and con-cepts to market. Without the Judys of the world the Internet technologies would never havetransformed our lives.What are your recommendations for students who want careers in computer science and networking?First, build a strong foundation in your academic work, balanced with any and every real-world work experience you can get. As you look for a working environment, seek opportu-nities in problem areas you really care about and with smart teams that you can learn from.585This page intentionally left blank CHAPTER 7MultimediaNetworking587People in all corners of the world are currently using the Internet to watch moviesand television shows on demand. Internet movie and television distribution compa-nies such as Netflix and Hulu in North America and Youku and Kankan in Chinahave practically become household names. But people are not only watching Internet videos, they are using sites like YouTube to upload and distribute their ownuser-generated content, becoming Internet video producers as well as consumers.Moreover, network applications such as Skype, Google Talk, and QQ (enormouslypopular in China) allow people to not only make telephone calls over the Internet,but to also enhance those calls with video and multi-person conferencing. In fact, we can safely predict that by the end of the current decade almost all video dis-tribution and voice conversations will take place end-to-end over the Internet, oftento wireless devices connected to the Internet via 4G and WiFi access networks.We begin this chapter with a taxonomy of multimedia applications in Section 7.1.Well see that a multimedia application can be classified as either streaming storedaudio/video, conversational voice/video-over-IP, or streaming live audio/video.Well see that each of these classes of applications has its own unique servicerequirements that differ significantly from those of traditional elastic applicationssuch as e-mail, Web browsing, and remote login. In Section 7.2, well examinevideo streaming in some detail. Well explore many of the underlying principlesbehind video streaming, including client buffering, prefetching, and adapting videoquality to available bandwidth. We will also investigate Content Distribution Net-works (CDNs), which are used extensively today by the leading video streamingsystems. We then examine the YouTube, Netflix, and Kankan systems as case studies for streaming video. In Section 7.3, we investigate conversational voice andvideo, which, unlike elastic applications, are highly sensitive to end-to-end delaybut can tolerate occasional loss of data. Here well examine how techniques such asadaptive playout, forward error correction, and error concealment can mitigateagainst network-induced packet loss and delay. Well also examine Skype as a casestudy. In Section 7.4, well study RTP and SIP, two popular protocols for real-timeconversational voice and video applications. In Section 7.5, well investigate mech-anisms within the network that can be used to distinguish one class of traffic (e.g.,delay-sensitive applications such as conversational voice) from another (e.g., elasticapplications such as browsing Web pages), and provide differentiated service amongmultiple classes of traffic.7.1 Multimedia Networking ApplicationsWe define a multimedia network application as any network application thatemploys audio or video. In this section, we provide a taxonomy of multimedia appli-cations. Well see that each class of applications in the taxonomy has its own uniqueset of service requirements and design issues. But before diving into an in-depth dis-cussion of Internet multimedia applications, it is useful to consider the intrinsiccharacteristics of the audio and video media themselves.7.1.1 Properties of VideoPerhaps the most salient characteristic of video is its high bit rate. Video distrib-uted over the Internet typically ranges from 100 kbps for low-quality video confer-encing to over 3 Mbps for streaming high-definition movies. To get a sense of howvideo bandwidth demands compare with those of other Internet applications, letsbriefly consider three different users, each using a different Internet application. Ourfirst user, Frank, is going quickly through photos posted on his friends Facebookpages. Lets assume that Frank is looking at a new photo every 10 seconds, and thatphotos are on average 200 Kbytes in size. (As usual, throughout this discussion wemake the simplifying assumption that 1 Kbyte = 8,000 bits.) Our second user,Martha, is streaming music from the Internet (the cloud) to her smartphone. Letsassume Martha is listening to many MP3 songs, one after the other, each encoded ata rate of 128 kbps. Our third user, Victor, is watching a video that has been encodedat 2 Mbps. Finally, lets suppose that the session length for all three users is 4,000seconds (approximately 67 minutes). Table 7.1 compares the bit rates and the totalbytes transferred for these three users. We see that video streaming consumes by far588 CHAPTER 7 MULTIMEDIA NETWORKINGthe most bandwidth, having a bit rate of more than ten times greater than that of theFacebook and music-streaming applications. Therefore, when designing networkedvideo applications, the first thing we must keep in mind is the high bit-rate require-ments of video. Given the popularity of video and its high bit rate, it is perhaps notsurprising that Cisco predicts [Cisco 2011] that streaming and stored video will beapproximately 90 percent of global consumer Internet traffic by 2015.Another important characteristic of video is that it can be compressed, therebytrading off video quality with bit rate. A video is a sequence of images, typicallybeing displayed at a constant rate, for example, at 24 or 30 images per second. Anuncompressed, digitally encoded image consists of an array of pixels, with eachpixel encoded into a number of bits to represent luminance and color. There are twotypes of redundancy in video, both of which can be exploited by video compres-sion. Spatial redundancy is the redundancy within a given image. Intuitively, animage that consists of mostly white space has a high degree of redundancy and canbe efficiently compressed without significantly sacrificing image quality. Temporalredundancy reflects repetition from image to subsequent image. If, for example, animage and the subsequent image are exactly the same, there is no reason to re-encode the subsequent image; it is instead more efficient simply to indicate duringencoding that the subsequent image is exactly the same. Todays off-the-shelf com-pression algorithms can compress a video to essentially any bit rate desired. Ofcourse, the higher the bit rate, the better the image quality and the better the overalluser viewing experience.We can also use compression to create multiple versions of the same video,each at a different quality level. For example, we can use compression to create, say,three versions of the same video, at rates of 300 kbps, 1 Mbps, and 3 Mbps. Userscan then decide which version they want to watch as a function of their currentavailable bandwidth. Users with high-speed Internet connections might choose the3 Mbps version; users watching the video over 3G with a smartphone might choosethe 300 kbps version. Similarly, the video in a video conference application can becompressed on-the-fly to provide the best video quality given the available end-to-end bandwidth between conversing users.7.1 MULTIMEDIA NETWORKING APPLICATIONS 589Bit rate Bytes transferred in 67 minFacebook Frank 160 kbps 80 MbytesMartha Music 128 kbps 64 MbytesVictor Video 2 Mbps 1 GbyteTable 7.1 Comparison of bit-rate requirements of three Internet applications7.1.2 Properties of AudioDigital audio (including digitized speech and music) has significantly lower band-width requirements than video. Digital audio, however, has its own unique proper-ties that must be considered when designing multimedia network applications. Tounderstand these properties, lets first consider how analog audio (which humansand musical instruments generate) is converted to a digital signal: The analog audio signal is sampled at some fixed rate, for example, at 8,000samples per second. The value of each sample is an arbitrary real number. Each of the samples is then rounded to one of a finite number of values. Thisoperation is referred to as quantization. The number of such finite valuescalled quantization valuesis typically a power of two, for example, 256 quan-tization values. Each of the quantization values is represented by a fixed number of bits. Forexample, if there are 256 quantization values, then each valueand hence eachaudio sampleis represented by one byte. The bit representations of all the sam-ples are then concatenated together to form the digital representation of the sig-nal. As an example, if an analog audio signal is sampled at 8,000 samples persecond and each sample is quantized and represented by 8 bits, then the resultingdigital signal will have a rate of 64,000 bits per second. For playback throughaudio speakers, the digital signal can then be converted backthat is, decodedto an analog signal. However, the decoded analog signal is only an approxima-tion of the original signal, and the sound quality may be noticeably degraded (forexample, high-frequency sounds may be missing in the decoded signal). Byincreasing the sampling rate and the number of quantization values, the decodedsignal can better approximate the original analog signal. Thus (as with video),there is a trade-off between the quality of the decoded signal and the bit-rate andstorage requirements of the digital signal.The basic encoding technique that we just described is called pulse code modulation(PCM). Speech encoding often uses PCM, with a sampling rate of 8,000 samples persecond and 8 bits per sample, resulting in a rate of 64 kbps. The audio compact disk(CD) also uses PCM, with a sampling rate of 44,100 samples per second with 16 bitsper sample; this gives a rate of 705.6 kbps for mono and 1.411 Mbps for stereo.PCM-encoded speech and music, however, are rarely used in the Internet.Instead, as with video, compression techniques are used to reduce the bit rates of thestream. Human speech can be compressed to less than 10 kbps and still be intelligi-ble. A popular compression technique for near CD-quality stereo music is MPEG 1layer 3, more commonly known as MP3. MP3 encoders can compress to many dif-ferent rates; 128 kbps is the most common encoding rate and produces very littlesound degradation. A related standard is Advanced Audio Coding (AAC), whichhas been popularized by Apple. As with video, multiple versions of a prerecordedaudio stream can be created, each at a different bit rate.590 CHAPTER 7 MULTIMEDIA NETWORKINGAlthough audio bit rates are generally much less than those of video, users aregenerally much more sensitive to audio glitches than video glitches. Consider, forexample, a video conference taking place over the Internet. If, from time to time, thevideo signal is lost for a few seconds, the video conference can likely proceed with-out too much user frustration. If, however, the audio signal is frequently lost, theusers may have to terminate the session.7.1.3 Types of Multimedia Network ApplicationsThe Internet supports a large variety of useful and entertaining multimedia applica-tions. In this subsection, we classify multimedia applications into three broad cate-gories: (i) streaming stored audio/video, (ii) conversational voice/video-over-IP,and (iii) streaming live audio/video. As we will soon see, each of these applicationcategories has its own set of service requirements and design issues.Streaming Stored Audio and VideoTo keep the discussion concrete, we focus here on streaming stored video, whichtypically combines video and audio components. Streaming stored audio (such asstreaming music) is very similar to streaming stored video, although the bit rates aretypically much lower.In this class of applications, the underlying medium is prerecorded video, suchas a movie, a television show, a prerecorded sporting event, or a prerecorded user-generated video (such as those commonly seen on YouTube). These prerecordedvideos are placed on servers, and users send requests to the servers to view the videoson demand. Many Internet companies today provide streaming video, includingYouTube (Google), Netflix, and Hulu. By some estimates, streaming stored videomakes up over 50 percent of the downstream traffic in the Internet access networkstoday [Cisco 2011]. Streaming stored video has three key distinguishing features. Streaming. In a streaming stored video application, the client typically beginsvideo playout within a few seconds after it begins receiving the video from theserver. This means that the client will be playing out from one location in the video while at the same time receiving later parts of the video from theserver. This technique, known as streaming, avoids having to download the entire video file (and incurring a potentially long delay) before playout begins. Interactivity. Because the media is prerecorded, the user may pause, repositionforward, reposition backward, fast-forward, and so on through the video content.The time from when the user makes such a request until the action manifests itselfat the client should be less than a few seconds for acceptable responsiveness. Continuous playout. Once playout of the video begins, it should proceedaccording to the original timing of the recording. Therefore, data must bereceived from the server in time for its playout at the client; otherwise, users7.1 MULTIMEDIA NETWORKING APPLICATIONS 591experience video frame freezing (when the client waits for the delayed frames)or frame skipping (when the client skips over delayed frames).By far, the most important performance measure for streaming video is averagethroughput. In order to provide continuous playout, the network must provide anaverage throughput to the streaming application that is at least as large the bit rate ofthe video itself. As we will see in Section 7.2, by using buffering and prefetching, itis possible to provide continuous playout even when the throughput fluctuates, aslong as the average throughput (averaged over 510 seconds) remains above thevideo rate [Wang 2008].For many streaming video applications, prerecorded video is stored on, andstreamed from, a CDN rather than from a single data center. There are also manyP2P video streaming applications for which the video is stored on users hosts(peers), with different chunks of video arriving from different peers that may spreadaround the globe. Given the prominence of Internet video streaming, we willexplore video streaming in some depth in Section 7.2, paying particular attention toclient buffering, prefetching, adapting quality to bandwidth availability, and CDNdistribution.Conversational Voice- and Video-over-IPReal-time conversational voice over the Internet is often referred to as Internettelephony, since, from the users perspective, it is similar to the traditional circuit-switched telephone service. It is also commonly called Voice-over-IP (VoIP). Con-versational video is similar, except that it includes the video of the participants aswell as their voices. Most of todays voice and video conversational systems allowusers to create conferences with three or more participants. Conversational voiceand video are widely used in the Internet today, with the Internet companies Skype,QQ, and Google Talk boasting hundreds of millions of daily users.In our discussion of application service requirements in Chapter 2 (Figure 2.4),we identified a number of axes along which application requirements can be classi-fied. Two of these axestiming considerations and tolerance of data lossare par-ticularly important for conversational voice and video applications. Timingconsiderations are important because audio and video conversational applicationsare highly delay-sensitive. For a conversation with two or more interacting speak-ers, the delay from when a user speaks or moves until the action is manifested at theother end should be less than a few hundred milliseconds. For voice, delays smallerthan 150 milliseconds are not perceived by a human listener, delays between 150and 400 milliseconds can be acceptable, and delays exceeding 400 milliseconds canresult in frustrating, if not completely unintelligible, voice conversations.On the other hand, conversational multimedia applications are loss-tolerantoccasional loss only causes occasional glitches in audio/video playback, and theselosses can often be partially or fully concealed. These delay-sensitive but loss-tolerant592 CHAPTER 7 MULTIMEDIA NETWORKINGcharacteristics are clearly different from those of elastic data applications such as Webbrowsing, e-mail, social networks, and remote login. For elastic applications, longdelays are annoying but not particularly harmful; the completeness and integrity of thetransferred data, however, are of paramount importance. We will explore conversa-tional voice and video in more depth in Section 7.3, paying particular attention to howadaptive playout, forward error correction, and error concealment can mitigate againstnetwork-induced packet loss and delay.Streaming Live Audio and VideoThis third class of applications is similar to traditional broadcast radio and televi-sion, except that transmission takes place over the Internet. These applications allowa user to receive a live radio or television transmissionsuch as a live sportingevent or an ongoing news eventtransmitted from any corner of the world. Today,thousands of radio and television stations around the world are broadcasting contentover the Internet.Live, broadcast-like applications often have many users who receive the sameaudio/video program at the same time. Although the distribution of live audio/videoto many receivers can be efficiently accomplished using the IP multicasting tech-niques described in Section 4.7, multicast distribution is more often accomplishedtoday via application-layer multicast (using P2P networks or CDNs) or through mul-tiple separate unicast streams. As with streaming stored multimedia, the networkmust provide each live multimedia flow with an average throughput that is largerthan the video consumption rate. Because the event is live, delay can also be an issue,although the timing constraints are much less stringent than those for conversationalvoice. Delays of up to ten seconds or so from when the user chooses to view a livetransmission to when playout begins can be tolerated. We will not cover streaminglive media in this book because many of the techniques used for streaming livemediainitial buffering delay, adaptive bandwidth use, and CDN distributionaresimilar to those for streaming stored media.7.2 Streaming Stored Video For streaming video applications, prerecorded videos are placed on servers, andusers send requests to these servers to view the videos on demand. The user maywatch the video from beginning to end without interruption, may stop watching thevideo well before it ends, or interact with the video by pausing or repositioning to afuture or past scene. Streaming video systems can be classified into three categories:UDP streaming, HTTP streaming, and adaptive HTTP streaming. Although allthree types of systems are used in practice, the majority of todays systems employHTTP streaming and adaptive HTTP streaming.7.2 STREAMING STORED VIDEO 593A common characteristic of all three forms of video streaming is the extensiveuse of client-side application buffering to mitigate the effects of varying end-to-enddelays and varying amounts of available bandwidth between server and client. Forstreaming video (both stored and live), users generally can tolerate a small several-second initial delay between when the client requests a video and when video play-out begins at the client. Consequently, when the video starts to arrive at the client,the client need not immediately begin playout, but can instead build up a reserve ofvideo in an application buffer. Once the client has built up a reserve of several sec-onds of buffered-but-not-yet-played video, the client can then begin video playout.There are two important advantages provided by such client buffering. First, client-side buffering can absorb variations in server-to-client delay. If a particular piece ofvideo data is delayed, as long as it arrives before the reserve of received-but-not-yet-played video is exhausted, this long delay will not be noticed. Second, if theserver-to-client bandwidth briefly drops below the video consumption rate, a usercan continue to enjoy continuous playback, again as long as the client applicationbuffer does not become completely drained.Figure 7.1 illustrates client-side buffering. In this simple example, suppose thatvideo is encoded at a fixed bit rate, and thus each video block contains video framesthat are to be played out over the same fixed amount of time, . The server trans-mits the first video block at , the second block at , the third block at, and so on. Once the client begins playout, each block should be played outtime units after the previous block in order to reproduce the timing of the originalrecorded video. Because of the variable end-to-end network delays, different videoblocks experience different delays. The first video block arrives at the client at t1and the second block arrives at . The network delay for the ith block is the hori-zontal distance between the time the block was transmitted by the server and thet2t0 + 2t0 + t0594 CHAPTER 7 MULTIMEDIA NETWORKINGVariablenetworkdelayClientplayoutdelayConstant bitrate videotransmissionby server123456789101112Constant bitrate videoplayoutby clientTimeVideo block numbert0 t1 t2 t3t0+2t0+ t1+ t3+Videoreceptionat clientFigure 7.1 Client playout delay in video streamingtime it is received at the client; note that the network delay varies from one videoblock to another. In this example, if the client were to begin playout as soon as thefirst block arrived at , then the second block would not have arrived in time to beplayed out at out at . In this case, video playout would either have to stall(waiting for block 1 to arrive) or block 1 could be skippedboth resulting in unde-sirable playout impairments. Instead, if the client were to delay the start of playoutuntil , when blocks 1 through 6 have all arrived, periodic playout can proceed withall blocks having been received before their playout time.7.2.1 UDP Streaming We only briefly discuss UDP streaming here, referring the reader to more in-depth dis-cussions of the protocols behind these systems where appropriate. With UDP stream-ing, the server transmits video at a rate that matches the clients video consumption rateby clocking out the video chunks over UDP at a steady rate. For example, if the videoconsumption rate is 2 Mbps and each UDP packet carries 8,000 bits of video, then theserver would transmit one UDP packet into its socket every (8000 bits)/(2 Mbps) = 4 msec. As we learned in Chapter 3, because UDP does not employ a congestion-controlmechanism, the server can push packets into the network at the consumption rate of thevideo without the rate-control restrictions of TCP. UDP streaming typically uses a smallclient-side buffer, big enough to hold less than a second of video.Before passing the video chunks to UDP, the server will encapsulate the videochunks within transport packets specially designed for transporting audio and video,using the Real-Time Transport Protocol (RTP) [RFC 3550] or a similar (possiblyproprietary) scheme. We delay our coverage of RTP until Section 7.3, where we dis-cuss RTP in the context of conversational voice and video systems.Another distinguishing property of UDP streaming is that in addition to the server-to-client video stream, the client and server also maintain, in parallel, a separate controlconnection over which the client sends commands regarding session state changes(such as pause, resume, reposition, and so on). This control connection is in many waysanalogous to the FTP control connection we studied in Chapter 2. The Real-TimeStreaming Protocol (RTSP) [RFC 2326], explained in some detail in the companionWeb site for this textbook, is a popular open protocol for such a control connection.Although UDP streaming has been employed in many open-source systems andproprietary products, it suffers from three significant drawbacks. First, due to theunpredictable and varying amount of available bandwidth between server and client,constant-rate UDP streaming can fail to provide continuous playout. For example,consider the scenario where the video consumption rate is 1 Mbps and the server-to-client available bandwidth is usually more than 1 Mbps, but every few minutesthe available bandwidth drops below 1 Mbps for several seconds. In such a scenario,a UDP streaming system that transmits video at a constant rate of 1 Mbps overRTP/UDP would likely provide a poor user experience, with freezing or skippedframes soon after the available bandwidth falls below 1 Mbps. The second drawbackt3t1 + t17.2 STREAMING STORED VIDEO 595of UDP streaming is that it requires a media control server, such as an RTSP server,to process client-to-server interactivity requests and to track client state (e.g., theclients playout point in the video, whether the video is being paused or played, andso on) for each ongoing client session. This increases the overall cost and complex-ity of deploying a large-scale video-on-demand system. The third drawback is thatmany firewalls are configured to block UDP traffic, preventing the users behindthese firewalls from receiving UDP video.7.2.2 HTTP StreamingIn HTTP streaming, the video is simply stored in an HTTP server as an ordinary filewith a specific URL. When a user wants to see the video, the client establishes aTCP connection with the server and issues an HTTP GET request for that URL. Theserver then sends the video file, within an HTTP response message, as quickly aspossible, that is, as quickly as TCP congestion control and flow control will allow.On the client side, the bytes are collected in a client application buffer. Once thenumber of bytes in this buffer exceeds a predetermined threshold, the client applica-tion begins playbackspecifically, it periodically grabs video frames from the client application buffer, decompresses the frames, and displays them on theusers screen.We learned in Chapter 3 that when transferring a file over TCP, the server-to-client transmission rate can vary significantly due to TCPs congestion control mecha-nism. In particular, it is not uncommon for the transmission rate to vary in asaw-tooth manner (for example, Figure 3.53) associated with TCP congestion con-trol. Furthermore, packets can also be significantly delayed due to TCPs retransmis-sion mechanism. Because of these characteristics of TCP, the conventional wisdom inthe 1990s was that video streaming would never work well over TCP. Over time, how-ever, designers of streaming video systems learned that TCPs congestion control andreliable-data transfer mechanisms do not necessarily preclude continuous playoutwhen client buffering and prefetching (discussed in the next section) are used.The use of HTTP over TCP also allows the video to traverse firewalls and NATsmore easily (which are often configured to block most UDP traffic but to allow mostHTTP traffic). Streaming over HTTP also obviates the need for a media controlserver, such as an RTSP server, reducing the cost of a large-scale deployment overthe Internet. Due to all of these advantages, most video streaming applicationstodayincluding YouTube and Netflixuse HTTP streaming (over TCP) as itsunderlying streaming protocol.Prefetching VideoWe just learned, client-side buffering can be used to mitigate the effects of vary-ing end-to-end delays and varying available bandwidth. In our earlier example inFigure 7.1, the server transmits video at the rate at which the video is to be played596 CHAPTER 7 MULTIMEDIA NETWORKING7.2 STREAMING STORED VIDEO 597out. However, for streaming stored video, the client can attempt to download thevideo at a rate higher than the consumption rate, thereby prefetching videoframes that are to be consumed in the future. This prefetched video is naturallystored in the client application buffer. Such prefetching occurs naturally with TCPstreaming, since TCPs congestion avoidance mechanism will attempt to use all ofthe available bandwidth between server and client.To gain some insight into prefetching, lets take a look at a simple example.Suppose the video consumption rate is 1 Mbps but the network is capable of deliv-ering the video from server to client at a constant rate of 1.5 Mbps. Then the clientwill not only be able to play out the video with a very small playout delay, but willalso be able to increase the amount of buffered video data by 500 Kbits every second. In this manner, if in the future the client receives data at a rate of less than 1Mbps for a brief period of time, the client will be able to continue to provide contin-uous playback due to the reserve in its buffer. [Wang 2008] shows that when theaverage TCP throughput is roughly twice the media bit rate, streaming over TCPresults in minimal starvation and low buffering delays.Client Application Buffer and TCP BuffersFigure 7.2 illustrates the interaction between client and server for HTTP streaming.At the server side, the portion of the video file in white has already been sent intothe servers socket, while the darkened portion is what remains to be sent. Afterpassing through the socket door, the bytes are placed in the TCP send bufferbefore being transmitted into the Internet, as described in Chapter 3. In Figure 7.2,Video fileWeb serverClientTCP sendbufferTCP receivebufferTCP applicationbufferFrames readout periodicallyfrom buffer,decompressed,and displayedon screenFigure 7.2 Streaming stored video over HTTP/TCPbecause the TCP send buffer is shown to be full, the server is momentarily preventedfrom sending more bytes from the video file into the socket. On the client side, theclient application (media player) reads bytes from the TCP receive buffer (throughits client socket) and places the bytes into the client application buffer. At the sametime, the client application periodically grabs video frames from the client applicationbuffer, decompresses the frames, and displays them on the users screen. Note thatif the client application buffer is larger than the video file, then the whole process ofmoving bytes from the servers storage to the clients application buffer is equiva-lent to an ordinary file download over HTTPthe client simply pulls the video offthe server as fast as TCP will allow!Consider now what happens when the user pauses the video during the streaming process. During the pause period, bits are not removed from the clientapplication buffer, even though bits continue to enter the buffer from the server. Ifthe client application buffer is finite, it may eventually become full, which willcause back pressure all the way back to the server. Specifically, once the clientapplication buffer becomes full, bytes can no longer be removed from the client TCP receive buffer, so it too becomes full. Once the client receive TCP bufferbecomes full, bytes can no longer be removed from the client TCP send buffer, soit also becomes full. Once the TCP send buffer becomes full, the server cannot sendany more bytes into the socket. Thus, if the user pauses the video, the server maybe forced to stop transmitting, in which case the server will be blocked until theuser resumes the video.In fact, even during regular playback (that is, without pausing), if the clientapplication buffer becomes full, back pressure will cause the TCP buffers tobecome full, which will force the server to reduce its rate. To determine theresulting rate, note that when the client application removes f bits, it creates roomfor f bits in the client application buffer, which in turn allows the server to send fadditional bits. Thus, the server send rate can be no higher than the video con-sumption rate at the client. Therefore, a full client application buffer indirectlyimposes a limit on the rate that video can be sent from server to client whenstreaming over HTTP.Analysis of Video StreamingSome simple modeling will provide more insight into initial playout delay andfreezing due to application buffer depletion. As shown in Figure 7.3, let B denotethe size (in bits) of the clients application buffer, and let Q denote the number ofbits that must be buffered before the client application begins playout. (Of course, Q < B.) Let r denote the video consumption ratethe rate at which the client drawsbits out of the client application buffer during playback. So, for example, if thevideos frame rate is 30 frames/sec, and each (compressed) frame is 100,000 bits,then r = 3 Mbps. To see the forest through the trees, well ignore TCPs send andreceive buffers.598 CHAPTER 7 MULTIMEDIA NETWORKINGLets assume that the server sends bits at a constant rate x whenever the clientbuffer is not full. (This is a gross simplification, since TCPs send rate varies due tocongestion control; well examine more realistic time-dependent rates x(t) in theproblems at the end of this chapter.) Suppose at time t = 0, the application buffer isempty and video begins arriving to the client application buffer. We now ask at whattime does playout begin? And while we are at it, at what time does theclient application buffer become full?First, lets determine , the time when Q bits have entered the applicationbuffer and playout begins. Recall that bits arrive to the client application buffer atrate x and no bits are removed from this buffer before playout begins. Thus, theamount of time required to build up Q bits (the initial buffering delay) is Q/x.Now lets determine , the point in time when the client application bufferbecomes full. We first observe that if x < r (that is, if the server send rate is less thanthe video consumption rate), then the client buffer will never become full! Indeed,starting at time , the buffer will be depleted at rate r and will only be filled at rate x < r. Eventually the client buffer will empty out entirely, at which time the videowill freeze on the screen while the client buffer waits another seconds to build upQ bits of video. Thus, when the available rate in the network is less than the videorate, playout will alternate between periods of continuous playout and periods offreezing. In a homework problem, you will be asked to determine the length of eachcontinuous playout and freezing period as a function of Q, r, and x. Now lets deter-mine for when x > r. In this case, starting at time , the buffer increases from Q to
B at rate x r since bits are being depleted at rate r but are arriving at rate x, as
shown in Figure 7.3. Given these hints, you will be asked in a homework problem
to determine , the time the client buffer becomes full. Note that when the available
rate in the network is more than the video rate, after the initial buffering delay, the
user will enjoy continuous playout until the video ends.
tf
tptf
tp
tp
tf
tp =
tp
t = tft = tp
7.2 STREAMING STORED VIDEO 599
Fill rate = x Depletion rate = r
Video
server
Internet
Q
B
Client application buffer
Figure 7.3 Analysis of client-side buffering for video streaming
Early Termination and Repositioning the Video
HTTP streaming systems often make use of the HTTP byte-range header in the
HTTP GET request message, which specifies the specific range of bytes the client
currently wants to retrieve from the desired video. This is particularly useful when
the user wants to reposition (that is, jump) to a future point in time in the video.
When the user repositions to a new position, the client sends a new HTTP request,
indicating with the byte-range header from which byte in the file should the server
send data. When the server receives the new HTTP request, it can forget about any
earlier request and instead send bytes beginning with the byte indicated in the byte-
range request.
While we are on the subject of repositioning, we briefly mention that when a user
repositions to a future point in the video or terminates the video early, some
prefetched-but-not-yet-viewed data transmitted by the server will go unwatcheda
waste of network bandwidth and server resources. For example, suppose that the client
buffer is full with B bits at some time t0 into the video, and at this time the user reposi-
tions to some instant t > t0 + B/r into the video, and then watches the video to comple-
tion from that point on. In this case, all B bits in the buffer will be unwatched and the
bandwidth and server resources that were used to transmit those B bits have been com-
pletely wasted. There is significant wasted bandwidth in the Internet due to early ter-
mination, which can be quite costly, particularly for wireless links [Ihm 2011]. For this
reason, many streaming systems use only a moderate-size client application buffer, or
will limit the amount of prefetched video using the byte-range header in HTTP
requests [Rao 2011].
Repositioning and early termination are analogous to cooking a large meal, eat-
ing only a portion of it, and throwing the rest away, thereby wasting food. So the
next time your parents criticize you for wasting food by not eating all your dinner,
you can quickly retort by saying they are wasting bandwidth and server resources
when they reposition while watching movies over the Internet! But, of course, two
wrongs do not make a rightboth food and bandwidth are not to be wasted!
7.2.3 Adaptive Streaming and DASH
Although HTTP streaming, as described in the previous subsection, has been exten-
sively deployed in practice (for example, by YouTube since its inception), it has a
major shortcoming: All clients receive the same encoding of the video, despite the
large variations in the amount of bandwidth available to a client, both across differ-
ent clients and also over time for the same client. This has led to the development of
a new type of HTTP-based streaming, often referred to as Dynamic Adaptive
Streaming over HTTP (DASH). In DASH, the video is encoded into several dif-
ferent versions, with each version having a different bit rate and, correspondingly, a
different quality level. The client dynamically requests chunks of video segments of
a few seconds in length from the different versions. When the amount of available
600 CHAPTER 7 MULTIMEDIA NETWORKING
bandwidth is high, the client naturally selects chunks from a high-rate version; and
when the available bandwidth is low, it naturally selects from a low-rate version.
The client selects different chunks one at a time with HTTP GET request messages
[Akhshabi 2011].
On one hand, DASH allows clients with different Internet access rates to stream
in video at different encoding rates. Clients with low-speed 3G connections can
receive a low bit-rate (and low-quality) version, and clients with fiber connections
can receive a high-quality version. On the other hand, DASH allows a client to adapt
to the available bandwidth if the end-to-end bandwidth changes during the session.
This feature is particularly important for mobile users, who typically see their band-
width availability fluctuate as they move with respect to the base stations. Comcast,
for example, has deployed an adaptive streaming system in which each video source
file is encoded into 8 to 10 different MPEG-4 formats, allowing the highest quality
video format to be streamed to the client, with adaptation being performed in
response to changing network and device conditions.
With DASH, each video version is stored in the HTTP server, each with a differ-
ent URL. The HTTP server also has a manifest file, which provides a URL for each
version along with its bit rate. The client first requests the manifest file and learns
about the various versions. The client then selects one chunk at a time by specifying a
URL and a byte range in an HTTP GET request message for each chunk. While down-
loading chunks, the client also measures the received bandwidth and runs a rate deter-
mination algorithm to select the chunk to request next. Naturally, if the client has a lot
of video buffered and if the measured receive bandwidth is high, it will choose a
chunk from a high-rate version. And naturally if the client has little video buffered and
the measured received bandwidth is low, it will choose a chunk from a low-rate ver-
sion. DASH therefore allows the client to freely switch among different quality levels.
Since a sudden drop in bit rate by changing versions may result in noticeable visual
quality degradation, the bit-rate reduction may be achieved using multiple intermedi-
ate versions to smoothly transition to a rate where the clients consumption rate drops
below its available receive bandwidth. When the network conditions improve, the
client can then later choose chunks from higher bit-rate versions.
By dynamically monitoring the available bandwidth and client buffer level, and
adjusting the transmission rate with version switching, DASH can often achieve
continuous playout at the best possible quality level without frame freezing or skip-
ping. Furthermore, since the client (rather than the server) maintains the intelligence
to determine which chunk to send next, the scheme also improves server-side scala-
bility. Another benefit of this approach is that the client can use the HTTP byte-range
request to precisely control the amount of prefetched video that it buffers locally.
We conclude our brief discussion of DASH by mentioning that for many imple-
mentations, the server not only stores many versions of the video but also separately
stores many versions of the audio. Each audio version has its own quality level and bit
rate and has its own URL. In these implementations, the client dynamically selects
both video and audio chunks, and locally synchronizes audio and video playout.
7.2 STREAMING STORED VIDEO 601
7.2.4 Content Distribution Networks
Today, many Internet video companies are distributing on-demand multi-Mbps
streams to millions of users on a daily basis. YouTube, for example, with a library
of hundreds of millions of videos, distributes hundreds of millions of video streams
to users around the world every day [Ding 2011]. Streaming all this traffic to loca-
tions all over the world while providing continuous playout and high interactivity is
clearly a challenging task.
For an Internet video company, perhaps the most straightforward approach to
providing streaming video service is to build a single massive data center, store all
of its videos in the data center, and stream the videos directly from the data center to
clients worldwide. But there are three major problems with this approach. First, if
the client is far from the data center, server-to-client packets will cross many com-
munication links and likely pass through many ISPs, with some of the ISPs possibly
located on different continents. If one of these links provides a throughput that is
less than the video consumption rate, the end-to-end throughput will also be below
the consumption rate, resulting in annoying freezing delays for the user. (Recall
from Chapter 1 that the end-to-end throughput of a stream is governed by the
throughput in the bottleneck link.) The likelihood of this happening increases as the
number of links in the end-to-end path increases. A second drawback is that a popu-
lar video will likely be sent many times over the same communication links. Not
only does this waste network bandwidth, but the Internet video company itself will
be paying its provider ISP (connected to the data center) for sending the same bytes
into the Internet over and over again. A third problem with this solution is that a sin-
gle data center represents a single point of failureif the data center or its links to
the Internet goes down, it would not be able to distribute any video streams.
In order to meet the challenge of distributing massive amounts of video data to
users distributed around the world, almost all major video-streaming companies
make use of Content Distribution Networks (CDNs). A CDN manages servers in
multiple geographically distributed locations, stores copies of the videos (and other
types of Web content, including documents, images, and audio) in its servers, and
attempts to direct each user request to a CDN location that will provide the best user
experience. The CDN may be a private CDN, that is, owned by the content provider
itself; for example, Googles CDN distributes YouTube videos and other types of
content. The CDN may alternatively be a third-party CDN that distributes content
on behalf of multiple content providers; Akamais CDN, for example, is a third-
party CDN that distributes Netflix and Hulu content, among others. A very readable
overview of modern CDNs is [Leighton 2009].
CDNs typically adopt one of two different server placement philosophies
[Huang 2008]:
Enter Deep. One philosophy, pioneered by Akamai, is to enter deep into the
access networks of Internet Service Providers, by deploying server clusters in
access ISPs all over the world. (Access networks are described in Section 1.3.)
602 CHAPTER 7 MULTIMEDIA NETWORKING
Akamai takes this approach with clusters in approximately 1,700 locations. The
goal is to get close to end users, thereby improving user-perceived delay and
throughput by decreasing the number of links and routers between the end user and
the CDN cluster from which it receives content. Because of this highly distributed
design, the task of maintaining and managing the clusters becomes challenging.
GOOGLES NETWORK INFRASTRUCTURE
To support its vast array of cloud servicesincluding search, gmail, calendar,
YouTube video, maps, documents, and social networksGoogle has deployed an
extensive private network and CDN infrastructure. Googles CDN infrastructure has
three tiers of server clusters:
Eight mega data centers, with six located in the United States and two locat-
ed in Europe [Google Locations 2012], with each data center having on the
order of 100,000 servers. These mega data centers are responsible for serving
dynamic (and often personalized) content, including search results and gmail
messages.
About 30 bring-home clusters (see discussion in 7.2.4), with each cluster con-
sisting on the order of 100500 servers [Adhikari 2011a]. The cluster loca-
tions are distributed around the world, with each location typically near multi-
ple tier-1 ISP PoPs. These clusters are responsible for serving static content,
including YouTube videos [Adhikari 2011a].
Many hundreds of enter-deep clusters (see discussion in 7.2.4), with each
cluster located within an access ISP. Here a cluster typically consists of tens of
servers within a single rack. These enter-deep servers perform TCP splitting (see
Section 3.7) and serve static content [Chen 2011], including the static portions
of Web pages that embody search results.
All of these data centers and cluster locations are networked together with Googles
own private network, as part of one enormous AS (AS 15169). When a user makes a
search query, often the query is first sent over the local ISP to a nearby enter-deep
cache, from where the static content is retrieved; while providing the static content to
the client, the nearby cache also forwards the query over Googles private network to
one of the mega data centers, from where the personalized search results are retrieved.
For a YouTube video, the video itself may come from one of the bring-home caches,
whereas portions of the Web page surrounding the video may come from the nearby
enter-deep cache, and the advertisements surrounding the video come from the data
centers. In summary, except for the local ISPs, the Google cloud services are largely
provided by a network infrastructure that is independent of the public Internet.
CASE STUDY
7.2 STREAMING STORED VIDEO 603
Bring Home. A second design philosophy, taken by Limelight and many other
CDN companies, is to bring the ISPs home by building large clusters at a smaller
number (for example, tens) of key locations and connecting these clusters using
a private high-speed network. Instead of getting inside the access ISPs, these
CDNs typically place each cluster at a location that is simultaneously near the
PoPs (see Section 1.3) of many tier-1 ISPs, for example, within a few miles of
both AT&T and Verizon PoPs in a major city. Compared with the enter-deep
design philosophy, the bring-home design typically results in lower maintenance
and management overhead, possibly at the expense of higher delay and lower
throughput to end users.
Once its clusters are in place, the CDN replicates content across its clusters. The
CDN may not want to place a copy of every video in each cluster, since some videos
are rarely viewed or are only popular in some countries. In fact, many CDNs do not
push videos to their clusters but instead use a simple pull strategy: If a client
requests a video from a cluster that is not storing the video, then the cluster retrieves
the video (from a central repository or from another cluster) and stores a copy
locally while streaming the video to the client at the same time. Similar to Internet
caches (see Chapter 2), when a clusters storage becomes full, it removes videos that
are not frequently requested.
CDN Operation
Having identified the two major approaches toward deploying a CDN, lets now
dive down into the nuts and bolts of how a CDN operates. When a browser in
a users host is instructed to retrieve a specific video (identified by a URL), the
CDN must intercept the request so that it can (1) determine a suitable CDN
server cluster for that client at that time, and (2) redirect the clients request to
a server in that cluster. Well shortly discuss how a CDN can determine a suitable
cluster. But first lets examine the mechanics behind intercepting and redirecting
a request.
Most CDNs take advantage of DNS to intercept and redirect requests; an inter-
esting discussion of such a use of the DNS is [Vixie 2009]. Lets consider a simple
example to illustrate how DNS is typically involved. Suppose a content provider,
NetCinema, employs the third-party CDN company, KingCDN, to distribute its
videos to its customers. On the NetCinema Web pages, each of its videos is assigned
a URL that includes the string video and a unique identifier for the video itself; for
example, Transformers 7 might be assigned http://video.netcinema.com/6Y7B23V.
Six steps then occur, as shown in Figure 7.4:
1. The user visits the Web page at NetCinema.
2. When the user clicks on the link http://video.netcinema.com/6Y7B23V, the
users host sends a DNS query for video.netcinema.com.
604 CHAPTER 7 MULTIMEDIA NETWORKING
http://video.netcinema.com/6Y7B23V
http://video.netcinema.com/6Y7B23V
Figure 7.4 DNS redirects a users request to a CDN server
3. The users Local DNS Server (LDNS) relays the DNS query to an authorita-
tive DNS server for NetCinema, which observes the string video in the
hostname video.netcinema.com. To hand over the DNS query to KingCDN,
instead of returning an IP address, the NetCinema authoritative DNS server
returns to the LDNS a hostname in the KingCDNs domain, for example,
a1105.kingcdn.com.
4. From this point on, the DNS query enters into KingCDNs private DNS
infrastructure. The users LDNS then sends a second query, now for
a1105.kingcdn.com, and KingCDNs DNS system eventually returns the
IP addresses of a KingCDN content server to the LDNS. It is thus here,
within the KingCDNs DNS system, that the CDN server from which the
client will receive its content is specified.
5. The LDNS forwards the IP address of the content-serving CDN node to the
users host.
6. Once the client receives the IP address for a KingCDN content server, it
establishes a direct TCP connection with the server at that IP address and
issues an HTTP GET request for the video. If DASH is used, the server will
first send to the client a manifest file with a list of URLs, one for each
version of the video, and the client will dynamically select chunks from the
different versions.
Local
DNS server
NetCinema authoritative
DNS server
www.NetCinema.com
KingCDN authoritative
server
KingCDN content
distribution server
2
5
6
3
1
4
7.2 STREAMING STORED VIDEO 605
www.NetCinema.com
Cluster Selection Strategies
At the core of any CDN deployment is a cluster selection strategy, that is, a mech-
anism for dynamically directing clients to a server cluster or a data center within the
CDN. As we just saw, the CDN learns the IP address of the clients LDNS server via
the clients DNS lookup. After learning this IP address, the CDN needs to select an
appropriate cluster based on this IP address. CDNs generally employ proprietary
cluster selection strategies. We now briefly survey a number of natural approaches,
each of which has its own advantages and disadvantages.
One simple strategy is to assign the client to the cluster that is geographically
closest. Using commercial geo-location databases (such as Quova [Quova 2012]
and Max-Mind [MaxMind 2012]), each LDNS IP address is mapped to a geographic
location. When a DNS request is received from a particular LDNS, the CDN
chooses the geographically closest cluster, that is, the cluster that is the fewest kilo-
meters from the LDNS as the bird flies. Such a solution can work reasonably well
for a large fraction of the clients [Agarwal 2009]. However, for some clients, the
solution may perform poorly, since the geographically closest cluster may not be the
closest cluster along the network path. Furthermore, a problem inherent with all
DNS-based approaches is that some end-users are configured to use remotely
located LDNSs [Shaikh 2001; Mao 2002], in which case the LDNS location may be
far from the clients location. Moreover, this simple strategy ignores the variation in
delay and available bandwidth over time of Internet paths, always assigning the
same cluster to a particular client.
In order to determine the best cluster for a client based on the current traffic
conditions, CDNs can instead perform periodic real-time measurements of delay
and loss performance between their clusters and clients. For instance, a CDN can
have each of its clusters periodically send probes (for example, ping messages or
DNS queries) to all of the LDNSs around the world. One drawback of this approach
is that many LDNSs are configured to not respond to such probes.
An alternative to sending extraneous traffic for measuring path properties is to
use the characteristics of recent and ongoing traffic between the clients and CDN
servers. For instance, the delay between a client and a cluster can be estimated by
examining the gap between server-to-client SYNACK and client-to-server ACK
during the TCP three-way handshake. Such solutions, however, require redirecting
clients to (possibly) suboptimal clusters from time to time in order to measure the
properties of paths to these clusters. Although only a small number of requests need
to serve as probes, the selected clients can suffer significant performance degradation
when receiving content (video or otherwise) [Andrews 2002; Krishnan 2009].
Another alternative for cluster-to-client path probing is to use DNS query traffic to
measure the delay between clients and clusters. Specifically, during the DNS phase
(within Step 4 in Figure 7.4), the clients LDNS can be occasionally directed to dif-
ferent DNS authoritative servers installed at the various cluster locations, yielding
DNS traffic that can then be measured between the LDNS and these cluster locations.
606 CHAPTER 7 MULTIMEDIA NETWORKING
In this scheme, the DNS servers continue to return the optimal cluster for the client,
so that delivery of videos and other Web objects does not suffer [Huang 2010].
A very different approach to matching clients with CDN servers is to use IP
anycast [RFC 1546]. The idea behind IP anycast is to have the routers in the Inter-
net route the clients packets to the closest cluster, as determined by BGP. Specifi-
cally, as shown in Figure 7.5, during the IP-anycast configuration stage, the CDN
company assigns the same IP address to each of its clusters, and uses standard BGP
to advertise this IP address from each of the different cluster locations. When a BGP
router receives multiple route advertisements for this same IP address, it treats these
advertisements as providing different paths to the same physical location (when, in
fact, the advertisements are for different paths to different physical locations).
Following standard operating procedures, the BGP router will then pick the best
(for example, closest, as determined by AS-hop counts) route to the IP address
according to its local route selection mechanism. For example, if one BGP route
AS1
AS3
3b
3c
3a
1a
1c
1b
1d
AS2
AS4
2a
2c
4a 4c
4b
Advertise
212.21.21.21
CDN Server B
CDN Server A
Advertise
212.21.21.21
Receive BGP
advertisements for
212.21.21.21 from
AS1 and from AS4.
Forward towards
Server B since it is
closer.
2b
Figure 7.5 Using IP anycast to route clients to closest CDN cluster
7.2 STREAMING STORED VIDEO 607
(corresponding to one location) is only one AS hop away from the router, and all
other BGP routes (corresponding to other locations) are two or more AS hops away,
then the BGP router would typically choose to route packets to the location that
needs to traverse only one AS (see Section 4.6). After this initial configuration phase,
the CDN can do its main job of distributing content. When any client wants to see
any video, the CDNs DNS returns the anycast address, no matter where the client is
located. When the client sends a packet to that IP address, the packet is routed to the
closest cluster as determined by the preconfigured forwarding tables, which were
configured with BGP as just described. This approach has the advantage of finding
the cluster that is closest to the client rather than the cluster that is closest to the
clients LDNS. However, the IP anycast strategy again does not take into account the
dynamic nature of the Internet over short time scales [Ballani 2006].
Besides network-related considerations such as delay, loss, and bandwidth per-
formance, there are many additional important factors that go into designing a clus-
ter selection strategy. Load on the clusters is one such factorclients should not be
directed to overloaded clusters. ISP delivery cost is another factorthe clusters may
be chosen so that specific ISPs are used to carry CDN-to-client traffic, taking into
account the different cost structures in the contractual relationships between ISPs
and cluster operators.
7.2.5 Case Studies: Netflix, YouTube, and Kankan
We conclude our discussion of streaming stored video by taking a look at three
highly successful large-scale deployments: Netflix, YouTube, and Kankan. Well see
that all these systems take very different approaches, yet employ many of the under-
lying principles discussed in this section.
Netflix
Generating almost 30 percent of the downstream U.S. Internet traffic in 2011, Netflix
has become the leading service provider for online movies and TV shows in the United
States [Sandvine 2011]. In order to rapidly deploy its large-scale service, Netflix has
made extensive use of third-party cloud services and CDNs. Indeed, Netflix is an inter-
esting example of a company deploying a large-scale online service by renting servers,
bandwidth, storage, and database services from third parties while using hardly any
infrastructure of its own. The following discussion is adapted from a very readable
measurement study of the Netflix architecture [Adhikari 2012]. As well see, Netflix
employs many of the techniques covered earlier in this section, including video distri-
bution using a CDN (actually multiple CDNs) and adaptive streaming over HTTP.
Figure 7.6 shows the basic architecture of the Netflix video-streaming platform.
It has four major components: the registration and payment servers, the Amazon
cloud, multiple CDN providers, and clients. In its own hardware infrastructure, Net-
flix maintains registration and payment servers, which handle registration of new
608 CHAPTER 7 MULTIMEDIA NETWORKING
Figure 7.6 Netflix video streaming platform
accounts and capture credit-card payment information. Except for these basic func-
tions, Netflix runs its online service by employing machines (or virtual machines) in
the Amazon cloud. Some of the functions taking place in the Amazon cloud include:
Content ingestion. Before Netflix can distribute a movie to its customers, it
must first ingest and process the movie. Netflix receives studio master versions
of movies and uploads them to hosts in the Amazon cloud.
Content processing. The machines in the Amazon cloud create many different
formats for each movie, suitable for a diverse array of client video players run-
ning on desktop computers, smartphones, and game consoles connected to tele-
visions. A different version is created for each of these formats and at multiple
bit rates, allowing for adaptive streaming over HTTP using DASH.
Uploading versions to the CDNs. Once all of the versions of a movie have
been created, the hosts in the Amazon cloud upload the versions to the CDNs.
To deliver the movies to its customers on demand, Netflix makes extensive use of
CDN technology. In fact, as of this writing in 2012, Netflix employs not one but three
third-party CDN companies at the same timeAkamai, Limelight, and Level-3.
Having described the components of the Netflix architecture, lets take a closer
look at the interaction between the client and the various servers that are involved in
Amazon Cloud
CDN server
CDN server
Upload
versions
to CDNs
Netflix
registration and
payment servers
CDN server
Client
Manifest
fileRegistration
and payment
Video
chunks
(DASH)
7.2 STREAMING STORED VIDEO 609
movie delivery. The Web pages for browsing the Netflix video library are served
from servers in the Amazon cloud. When the user selects a movie to Play Now,
the users client obtains a manifest file, also from servers in the Amazon cloud. The
manifest file includes a variety of information, including a ranked list of CDNs and
the URLs for the different versions of the movie, which are used for DASH play-
back. The ranking of the CDNs is determined by Netflix, and may change from one
streaming session to the next. Typically the client will select the CDN that is ranked
highest in the manifest file. After the client selects a CDN, the CDN leverages DNS
to redirect the client to a specific CDN server, as described in Section 7.2.4. The
client and that CDN server then interact using DASH. Specifically, as described in
Section 7.2.3, the client uses the byte-range header in HTTP GET request messages,
to request chunks from the different versions of the movie. Netflix uses chunks that
are approximately four-seconds long [Adhikari 2012]. While the chunks are being
downloaded, the client measures the received throughput and runs a rate-determination
algorithm to determine the quality of the next chunk to request.
Netflix embodies many of the key principles discussed earlier in this section,
including adaptive streaming and CDN distribution. Netflix also nicely illustrates
how a major Internet service, generating almost 30 percent of Internet traffic, can
run almost entirely on a third-party cloud and third-party CDN infrastructures, using
very little infrastructure of its own!
YouTube
With approximately half a billion videos in its library and half a billion video views
per day [Ding 2011], YouTube is indisputably the worlds largest video-sharing site.
YouTube began its service in April 2005 and was acquired by Google in November
2006. Although the Google/YouTube design and protocols are proprietary, through
several independent measurement efforts we can gain a basic understanding about
how YouTube operates [Zink 2009; Torres 2011; Adhikari 2011a].
As with Netflix, YouTube makes extensive use of CDN technology to dis-
tribute its videos [Torres 2011]. Unlike Netflix, however, Google does not
employ third-party CDNs but instead uses its own private CDN to distribute
YouTube videos. Google has installed server clusters in many hundreds of differ-
ent locations. From a subset of about 50 of these locations, Google distributes
YouTube videos [Adhikari 2011a]. Google uses DNS to redirect a customer
request to a specific cluster, as described in Section 7.2.4. Most of the time,
Googles cluster selection strategy directs the client to the cluster for which the
RTT between client and cluster is the lowest; however, in order to balance the
load across clusters, sometimes the client is directed (via DNS) to a more distant
cluster [Torres 2011]. Furthermore, if a cluster does not have the requested video,
instead of fetching it from somewhere else and relaying it to the client, the clus-
ter may return an HTTP redirect message, thereby redirecting the client to
another cluster [Torres 2011].
610 CHAPTER 7 MULTIMEDIA NETWORKING
YouTube employs HTTP streaming, as discussed in Section 7.2.2. YouTube
often makes a small number of different versions available for a video, each with a
different bit rate and corresponding quality level. As of 2011, YouTube does not
employ adaptive streaming (such as DASH), but instead requires the user to manu-
ally select a version. In order to save bandwidth and server resources that would be
wasted by repositioning or early termination, YouTube uses the HTTP byte range
request to limit the flow of transmitted data after a target amount of video is
prefetched.
A few million videos are uploaded to YouTube every day. Not only are
YouTube videos streamed from server to client over HTTP, but YouTube uploaders
also upload their videos from client to server over HTTP. YouTube processes each
video it receives, converting it to a YouTube video format and creating multiple ver-
sions at different bit rates. This processing takes place entirely within Google data
centers. Thus, in stark contrast to Netflix, which runs its service almost entirely on
third-party infrastructures, Google runs the entire YouTube service within its own
vast infrastructure of data centers, private CDN, and private global network
interconnecting its data centers and CDN clusters. (See the case study on Googles
network infrastructure in Section 7.2.4.)
Kankan
We just saw that for both the Netflix and YouTube services, servers operated by
CDNs (either third-party or private CDNs) stream videos to clients. Netflix and
YouTube not only have to pay for the server hardware (either directly through own-
ership or indirectly through rent), but also for the bandwidth the servers use to dis-
tribute the videos. Given the scale of these services and the amount of bandwidth
they are consuming, such a client-server deployment is extremely costly.
We conclude this section by describing an entirely different approach for provid-
ing video on demand over the Internet at a large scaleone that allows the service
provider to significantly reduce its infrastructure and bandwidth costs. As you might
suspect, this approach uses P2P delivery instead of client-server (via CDNs) delivery.
P2P video delivery is used with great success by several companies in China, includ-
ing Kankan (owned and operated by Xunlei), PPTV (formerly PPLive), and PPs (for-
merly PPstream). Kankan, currently the leading P2P-based video-on-demand provider
in China, has over 20 million unique users viewing its videos every month.
At a high level, P2P video streaming is very similar to BitTorrent file down-
loading (discussed in Chapter 2). When a peer wants to see a video, it contacts a
tracker (which may be centralized or peer-based using a DHT) to discover other
peers in the system that have a copy of that video. This peer then requests chunks
of the video file in parallel from these other peers that have the file. Different from
downloading with BitTorrent, however, requests are preferentially made for
chunks that are to be played back in the near future in order to ensure continuous
playback.
7.2 STREAMING STORED VIDEO 611
The Kankan design employs a tracker and its own DHT for tracking content.
Swarm sizes for the most popular content involve tens of thousands of peers, typi-
cally larger than the largest swarms in BitTorrent [Dhungel 2012]. The Kankan pro-
tocolsfor communication between peer and tracker, between peer and DHT, and
among peersare all proprietary. Interestingly, for distributing video chunks among
peers, Kankan uses UDP whenever possible, leading to massive amounts of UDP
traffic within Chinas Internet [Zhang M 2010].
7.3 Voice-over-IP
Real-time conversational voice over the Internet is often referred to as Internet
telephony, since, from the users perspective, it is similar to the traditional
circuit-switched telephone service. It is also commonly called Voice-over-IP
(VoIP). In this section we describe the principles and protocols underlying VoIP.
Conversational video is similar in many respects to VoIP, except that it includes
the video of the participants as well as their voices. To keep the discussion focused
and concrete, we focus here only on voice in this section rather than combined
voice and video.
7.3.1 Limitations of the Best-Effort IP Service
The Internets network-layer protocol, IP, provides best-effort service. That is to say
the service makes its best effort to move each datagram from source to destination
as quickly as possible but makes no promises whatsoever about getting the packet
to the destination within some delay bound or about a limit on the percentage of
packets lost. The lack of such guarantees poses significant challenges to the design
of real-time conversational applications, which are acutely sensitive to packet delay,
jitter, and loss.
In this section, well cover several ways in which the performance of
VoIP over a best-effort network can be enhanced. Our focus will be on applica-
tion-layer techniques, that is, approaches that do not require any changes in the
network core or even in the transport layer at the end hosts. To keep the discus-
sion concrete, well discuss the limitations of best-effort IP service in the context
of a specific VoIP example. The sender generates bytes at a rate of 8,000 bytes
per second; every 20 msecs the sender gathers these bytes into a chunk. A chunk
and a special header (discussed below) are encapsulated in a UDP segment, via a
call to the socket interface. Thus, the number of bytes in a chunk is (20 msecs)
(8,000 bytes/sec) = 160 bytes, and a UDP segment is sent every 20 msecs.
If each packet makes it to the receiver with a constant end-to-end delay, then
packets arrive at the receiver periodically every 20 msecs. In these ideal conditions,
612 CHAPTER 7 MULTIMEDIA NETWORKING
the receiver can simply play back each chunk as soon as it arrives. But unfortu-
nately, some packets can be lost and most packets will not have the same end-to-end
delay, even in a lightly congested Internet. For this reason, the receiver must take
more care in determining (1) when to play back a chunk, and (2) what to do with a
missing chunk.
Packet Loss
Consider one of the UDP segments generated by our VoIP application. The UDP
segment is encapsulated in an IP datagram. As the datagram wanders through the
network, it passes through router buffers (that is, queues) while waiting for trans-
mission on outbound links. It is possible that one or more of the buffers in the path
from sender to receiver is full, in which case the arriving IP datagram may be dis-
carded, never to arrive at the receiving application.
Loss could be eliminated by sending the packets over TCP (which provides
for reliable data transfer) rather than over UDP. However, retransmission mecha-
nisms are often considered unacceptable for conversational real-time audio appli-
cations such as VoIP, because they increase end-to-end delay [Bolot 1996].
Furthermore, due to TCP congestion control, packet loss may result in a reduc-
tion of the TCP senders transmission rate to a rate that is lower than the
receivers drain rate, possibly leading to buffer starvation. This can have a severe
impact on voice intelligibility at the receiver. For these reasons, most existing
VoIP applications run over UDP by default. [Baset 2006] reports that UDP is
used by Skype unless a user is behind a NAT or firewall that blocks UDP
segments (in which case TCP is used).
But losing packets is not necessarily as disastrous as one might think.
Indeed, packet loss rates between 1 and 20 percent can be tolerated, depending
on how voice is encoded and transmitted, and on how the loss is concealed at the
receiver. For example, forward error correction (FEC) can help conceal packet
loss. Well see below that with FEC, redundant information is transmitted along
with the original information so that some of the lost original data can be recov-
ered from the redundant information. Nevertheless, if one or more of the links
between sender and receiver is severely congested, and packet loss exceeds 10 to
20 percent (for example, on a wireless link), then there is really nothing that can
be done to achieve acceptable audio quality. Clearly, best-effort service has its
limitations.
End-to-End Delay
End-to-end delay is the accumulation of transmission, processing, and queuing
delays in routers; propagation delays in links; and end-system processing delays.
For real-time conversational applications, such as VoIP, end-to-end delays smaller
than 150 msecs are not perceived by a human listener; delays between 150 and 400
7.3 VOICE-OVER-IP 613
msecs can be acceptable but are not ideal; and delays exceeding 400 msecs can seri-
ously hinder the interactivity in voice conversations. The receiving side of a VoIP
application will typically disregard any packets that are delayed more than a certain
threshold, for example, more than 400 msecs. Thus, packets that are delayed by
more than the threshold are effectively lost.
Packet Jitter
A crucial component of end-to-end delay is the varying queuing delays that a packet
experiences in the networks routers. Because of these varying delays, the time from
when a packet is generated at the source until it is received at the receiver can fluctu-
ate from packet to packet, as shown in Figure 7.1. This phenomenon is called jitter.
As an example, consider two consecutive packets in our VoIP application. The sender
sends the second packet 20 msecs after sending the first packet. But at the receiver,
the spacing between these packets can become greater than 20 msecs. To see this,
suppose the first packet arrives at a nearly empty queue at a router, but just before the
second packet arrives at the queue a large number of packets from other sources
arrive at the same queue. Because the first packet experiences a small queuing delay
and the second packet suffers a large queuing delay at this router, the first and second
packets become spaced by more than 20 msecs. The spacing between consecutive
packets can also become less than 20 msecs. To see this, again consider two consecu-
tive packets. Suppose the first packet joins the end of a queue with a large number of
packets, and the second packet arrives at the queue before this first packet is trans-
mitted and before any packets from other sources arrive at the queue. In this case, our
two packets find themselves one right after the other in the queue. If the time it takes
to transmit a packet on the routers outbound link is less than 20 msecs, then the spac-
ing between first and second packets becomes less than 20 msecs.
The situation is analogous to driving cars on roads. Suppose you and your
friend are each driving in your own cars from San Diego to Phoenix. Suppose
you and your friend have similar driving styles, and that you both drive at
100 km/hour, traffic permitting. If your friend starts out one hour before you,
depending on intervening traffic, you may arrive at Phoenix more or less than one
hour after your friend.
If the receiver ignores the presence of jitter and plays out chunks as soon as
they arrive, then the resulting audio quality can easily become unintelligible at the
receiver. Fortunately, jitter can often be removed by using sequence numbers,
timestamps, and a playout delay, as discussed below.
7.3.2 Removing Jitter at the Receiver for Audio
For our VoIP application, where packets are being generated periodically, the
receiver should attempt to provide periodic playout of voice chunks in the presence
614 CHAPTER 7 MULTIMEDIA NETWORKING
of random network jitter. This is typically done by combining the following two
mechanisms:
Prepending each chunk with a timestamp. The sender stamps each chunk with
the time at which the chunk was generated.
Delaying playout of chunks at the receiver. As we saw in our earlier discussion
of Figure 7.1, the playout delay of the received audio chunks must be long
enough so that most of the packets are received before their scheduled playout
times. This playout delay can either be fixed throughout the duration of the audio
session or vary adaptively during the audio session lifetime.
We now discuss how these three mechanisms, when combined, can alleviate or
even eliminate the effects of jitter. We examine two playback strategies: fixed play-
out delay and adaptive playout delay.
Fixed Playout Delay
With the fixed-delay strategy, the receiver attempts to play out each chunk exactly q
msecs after the chunk is generated. So if a chunk is timestamped at the sender at
time t, the receiver plays out the chunk at time t + q, assuming the chunk has arrived
by that time. Packets that arrive after their scheduled playout times are discarded
and considered lost.
What is a good choice for q? VoIP can support delays up to about 400 msecs,
although a more satisfying conversational experience is achieved with smaller values
of q. On the other hand, if q is made much smaller than 400 msecs, then many packets
may miss their scheduled playback times due to the network-induced packet jitter.
Roughly speaking, if large variations in end-to-end delay are typical, it is preferable to
use a large q; on the other hand, if delay is small and variations in delay are also small,
it is preferable to use a small q, perhaps less than 150 msecs.
The trade-off between the playback delay and packet loss is illustrated in
Figure 7.7. The figure shows the times at which packets are generated and played
out for a single talk spurt. Two distinct initial playout delays are considered. As
shown by the leftmost staircase, the sender generates packets at regular inter-
valssay, every 20 msecs. The first packet in this talk spurt is received at time r.
As shown in the figure, the arrivals of subsequent packets are not evenly spaced
due to the network jitter.
For the first playout schedule, the fixed initial playout delay is set to p r. With
this schedule, the fourth packet does not arrive by its scheduled playout time, and
the receiver considers it lost. For the second playout schedule, the fixed initial play-
out delay is set to p r. For this schedule, all packets arrive before their scheduled
playout times, and there is therefore no loss.
7.3 VOICE-OVER-IP 615
616 CHAPTER 7 MULTIMEDIA NETWORKING
Packets
generated
Time
P
a
ck
e
ts
r p p
Playout
schedule
pr
Playout
schedule
pr
Packets
received
Missed
playout
Figure 7.7 Packet loss for different fixed playout delays
Adaptive Playout Delay
The previous example demonstrates an important delay-loss trade-off that arises
when designing a playout strategy with fixed playout delays. By making the initial
playout delay large, most packets will make their deadlines and there will therefore
be negligible loss; however, for conversational services such as VoIP, long delays
can become bothersome if not intolerable. Ideally, we would like the playout delay
to be minimized subject to the constraint that the loss be below a few percent.
The natural way to deal with this trade-off is to estimate the network delay and
the variance of the network delay, and to adjust the playout delay accordingly at the
beginning of each talk spurt. This adaptive adjustment of playout delays at the
beginning of the talk spurts will cause the senders silent periods to be compressed
and elongated; however, compression and elongation of silence by a small amount
is not noticeable in speech.
Following [Ramjee 1994], we now describe a generic algorithm that the
receiver can use to adaptively adjust its playout delays. To this end, let
ti = the timestamp of the ith packet = the time the packet was generated by the
sender
ri = the time packet i is received by receiver
pi = the time packet i is played at receiver
The end-to-end network delay of the ith packet is ri ti. Due to network jitter,
this delay will vary from packet to packet. Let di denote an estimate of the average
network delay upon reception of the ith packet. This estimate is constructed from
the timestamps as follows:
di = (1 u) di1 + u (ri ti)
where u is a fixed constant (for example, u = 0.01). Thus di is a smoothed average
of the observed network delays r1 t1, . . . , ri ti. The estimate places more weight
on the recently observed network delays than on the observed network delays of the
distant past. This form of estimate should not be completely unfamiliar; a similar
idea is used to estimate round-trip times in TCP, as discussed in Chapter 3. Let vi
denote an estimate of the average deviation of the delay from the estimated average
delay. This estimate is also constructed from the timestamps:
vi = (1 u) vi1 + u | ri ti di |
The estimates di and vi are calculated for every packet received, although they are
used only to determine the playout point for the first packet in any talk spurt.
Once having calculated these estimates, the receiver employs the following
algorithm for the playout of packets. If packet i is the first packet of a talk spurt, its
playout time, pi, is computed as:
pi = ti + di + Kvi
where K is a positive constant (for example, K = 4). The purpose of the Kvi term is to
set the playout time far enough into the future so that only a small fraction of the arriv-
ing packets in the talk spurt will be lost due to late arrivals. The playout point for any
subsequent packet in a talk spurt is computed as an offset from the point in time when
the first packet in the talk spurt was played out. In particular, let
qi = pi ti
be the length of time from when the first packet in the talk spurt is generated until it
is played out. If packet j also belongs to this talk spurt, it is played out at time
pj = tj + qi
The algorithm just described makes perfect sense assuming that the receiver can
tell whether a packet is the first packet in the talk spurt. This can be done by exam-
ining the signal energy in each received packet.
7.3.3 Recovering from Packet Loss
We have discussed in some detail how a VoIP application can deal with packet jitter.
We now briefly describe several schemes that attempt to preserve acceptable audio
7.3 VOICE-OVER-IP 617
quality in the presence of packet loss. Such schemes are called loss recovery schemes.
Here we define packet loss in a broad sense: A packet is lost either if it never arrives at
the receiver or if it arrives after its scheduled playout time. Our VoIP example will
again serve as a context for describing loss recovery schemes.
As mentioned at the beginning of this section, retransmitting lost packets may not
be feasible in a real-time conversational application such as VoIP. Indeed, retransmit-
ting a packet that has missed its playout deadline serves absolutely no purpose. And
retransmitting a packet that overflowed a router queue cannot normally be accom-
plished quickly enough. Because of these considerations, VoIP applications often use
some type of loss anticipation scheme. Two types of loss anticipation schemes are
forward error correction (FEC) and interleaving.
Forward Error Correction (FEC)
The basic idea of FEC is to add redundant information to the original packet stream.
For the cost of marginally increasing the transmission rate, the redundant information
can be used to reconstruct approximations or exact versions of some of the lost packets.
Following [Bolot 1996] and [Perkins 1998], we now outline two simple FEC mecha-
nisms. The first mechanism sends a redundant encoded chunk after every n chunks. The
redundant chunk is obtained by exclusive OR-ing the n original chunks [Shacham
1990]. In this manner if any one packet of the group of n + 1 packets is lost, the receiver
can fully reconstruct the lost packet. But if two or more packets in a group are lost, the
receiver cannot reconstruct the lost packets. By keeping n + 1, the group size, small, a
large fraction of the lost packets can be recovered when loss is not excessive. However,
the smaller the group size, the greater the relative increase of the transmission rate. In
particular, the transmission rate increases by a factor of 1/n, so that, if n = 3, then the
transmission rate increases by 33 percent. Furthermore, this simple scheme increases
the playout delay, as the receiver must wait to receive the entire group of packets before
it can begin playout. For more practical details about how FEC works for multimedia
transport see [RFC 5109].
The second FEC mechanism is to send a lower-resolution audio stream as the
redundant information. For example, the sender might create a nominal audio
stream and a corresponding low-resolution, low-bit rate audio stream. (The nominal
stream could be a PCM encoding at 64 kbps, and the lower-quality stream could be
a GSM encoding at 13 kbps.) The low-bit rate stream is referred to as the redundant
stream. As shown in Figure 7.8, the sender constructs the nth packet by taking the
nth chunk from the nominal stream and appending to it the (n 1)st chunk from the
redundant stream. In this manner, whenever there is nonconsecutive packet loss, the
receiver can conceal the loss by playing out the low-bit rate encoded chunk that
arrives with the subsequent packet. Of course, low-bit rate chunks give lower qual-
ity than the nominal chunks. However, a stream of mostly high-quality chunks,
occasional low-quality chunks, and no missing chunks gives good overall audio
quality. Note that in this scheme, the receiver only has to receive two packets before
playback, so that the increased playout delay is small. Furthermore, if the low-bit
618 CHAPTER 7 MULTIMEDIA NETWORKING
rate encoding is much less than the nominal encoding, then the marginal increase in
the transmission rate will be small.
In order to cope with consecutive loss, we can use a simple variation. Instead of
appending just the (n 1)st low-bit rate chunk to the nth nominal chunk, the sender
can append the (n 1)st and (n 2)nd low-bit rate chunk, or append the (n 1)st
and (n 3)rd low-bit rate chunk, and so on. By appending more low-bit rate chunks
to each nominal chunk, the audio quality at the receiver becomes acceptable for a
wider variety of harsh best-effort environments. On the other hand, the additional
chunks increase the transmission bandwidth and the playout delay.
Interleaving
As an alternative to redundant transmission, a VoIP application can send interleaved
audio. As shown in Figure 7.9, the sender resequences units of audio data before trans-
mission, so that originally adjacent units are separated by a certain distance in the trans-
mitted stream. Interleaving can mitigate the effect of packet losses. If, for example,
units are 5 msecs in length and chunks are 20 msecs (that is, four units per chunk), then
the first chunk could contain units 1, 5, 9, and 13; the second chunk could contain units
2, 6, 10, and 14; and so on. Figure 7.9 shows that the loss of a single packet from an
interleaved stream results in multiple small gaps in the reconstructed stream, as
opposed to the single large gap that would occur in a noninterleaved stream.
Interleaving can significantly improve the perceived quality of an audio stream
[Perkins 1998]. It also has low overhead. The obvious disadvantage of interleaving is
that it increases latency. This limits its use for conversational applications such as VoIP,
although it can perform well for streaming stored audio. A major advantage of inter-
leaving is that it does not increase the bandwidth requirements of a stream.
1
1 1
1
1
2
2
2 2
3
3
loss
3 4
3 41 2
3 4
4
Redundancy
Received
stream
Original
stream
Reconstructed
stream
Figure 7.8 Piggybacking lower-quality redundant information
7.3 VOICE-OVER-IP 619
Error Concealment
Error concealment schemes attempt to produce a replacement for a lost packet that
is similar to the original. As discussed in [Perkins 1998], this is possible since audio
signals, and in particular speech, exhibit large amounts of short-term self-similarity.
As such, these techniques work for relatively small loss rates (less than 15 percent),
and for small packets (440 msecs). When the loss length approaches the length of a
phoneme (5100 msecs) these techniques break down, since whole phonemes may
be missed by the listener.
Perhaps the simplest form of receiver-based recovery is packet repetition.
Packet repetition replaces lost packets with copies of the packets that arrived
immediately before the loss. It has low computational complexity and performs
reasonably well. Another form of receiver-based recovery is interpolation, which
uses audio before and after the loss to interpolate a suitable packet to cover the
loss. Interpolation performs somewhat better than packet repetition but is signifi-
cantly more computationally intensive [Perkins 1998].
7.3.4 Case Study: VoIP with Skype
Skype is an immensely popular VoIP application with over 50 million accounts
active on a daily basis. In addition to providing host-to-host VoIP service, Skype
offers host-to-phone services, phone-to-host services, and multi-party host-to-host
620 CHAPTER 7 MULTIMEDIA NETWORKING
Original
stream
Interleaved
stream
Received
stream
Reconstructed
stream
5 9 131
2 41
2 3 41
5 9 131
2 10 146
5 86
5 7 86
2 10 14 loss6
7 11 153
10 129
10 11 129
4 12 168
13 1614
13 15 1614
4 12 168
Figure 7.9 Sending interleaved audio
video conferencing services. (Here, a host is again any Internet connected IP device,
including PCs, tablets, and smartphones.) Skype was acquired by Microsoft in 2011
for over $8 billion.
Because the Skype protocol is proprietary, and because all Skypes control and
media packets are encrypted, it is difficult to precisely determine how Skype operates.
Nevertheless, from the Skype Web site and several measurement studies, researchers
have learned how Skype generally works [Baset 2006; Guha 2006; Chen 2006; Suh
2006; Ren 2006; Zhang X 2012]. For both voice and video, the Skype clients have at
their disposal many different codecs, which are capable of encoding the media at a wide
range of rates and qualities. For example, video rates for Skype have been measured to
be as low as 30 kbps for a low-quality session up to almost1 Mbps for a high quality
session [Zhang X 2012]. Typically, Skypes audio quality is better than the POTS
(Plain Old Telephone Service) quality provided by the wire-line phone system. (Skype
codecs typically sample voice at 16,000 samples/sec or higher, which provides richer
tones than POTS, which samples at 8,000/sec.) By default, Skype sends audio and
video packets over UDP. However, control packets are sent over TCP, and media pack-
ets are also sent over TCP when firewalls block UDP streams. Skype uses FEC for loss
recovery for both voice and video streams sent over UDP. The Skype client also adapts
the audio and video streams it sends to current network conditions, by changing video
quality and FEC overhead [Zhang X 2012].
Skype uses P2P techniques in a number of innovative ways, nicely illustrating
how P2P can be used in applications that go beyond content distribution and file
sharing. As with instant messaging, host-to-host Internet telephony is inherently P2P
since, at the heart of the application, pairs of users (that is, peers) communicate with
each other in real time. But Skype also employs P2P techniques for two other impor-
tant functions, namely, for user location and for NAT traversal.
As shown in Figure 7.10, the peers (hosts) in Skype are organized into a hierar-
chical overlay network, with each peer classified as a super peer or an ordinary peer.
Skype maintains an index that maps Skype usernames to current IP addresses (and
port numbers). This index is distributed over the super peers. When Alice wants to
call Bob, her Skype client searches the distributed index to determine Bobs current
IP address. Because the Skype protocol is proprietary, it is currently not known how
the index mappings are organized across the super peers, although some form of
DHT organization is very possible.
P2P techniques are also used in Skype relays, which are useful for establish-
ing calls between hosts in home networks. Many home network configurations
provide access to the Internet through NATs, as discussed in Chapter 4. Recall that
a NAT prevents a host from outside the home network from initiating a connec-
tion to a host within the home network. If both Skype callers have NATs, then
there is a problemneither can accept a call initiated by the other, making a call
seemingly impossible. The clever use of super peers and relays nicely solves this
problem. Suppose that when Alice signs in, she is assigned to a non-NATed super
peer and initiates a session to that super peer. (Since Alice is initiating the session,
her NAT permits this session.) This session allows Alice and her super peer to
7.3 VOICE-OVER-IP 621
exchange control messages. The same happens for Bob when he signs in. Now,
when Alice wants to call Bob, she informs her super peer, who in turn informs
Bobs super peer, who in turn informs Bob of Alices incoming call. If Bob
accepts the call, the two super peers select a third non-NATed super peerthe
relay peerwhose job will be to relay data between Alice and Bob. Alices and
Bobs super peers then instruct Alice and Bob respectively to initiate a session
with the relay. As shown in Figure 7.10, Alice then sends voice packets to the
relay over the Alice-to-relay connection (which was initiated by Alice), and the
relay then forwards these packets over the relay-to-Bob connection (which was
initiated by Bob); packets from Bob to Alice flow over these same two relay con-
nections in reverse. And voila!Bob and Alice have an end-to-end connection
even though neither can accept a session originating from outside.
Up to now, our discussion on Skype has focused on calls involving two persons.
Now lets examine multi-party audio conference calls. With N > 2 participants, if each
user were to senda copy of its audio stream to each of the N 1 other users, then a
total of N(N 1) audio streams would need to be sent into the network to support the
audio conference. To reduce this bandwidth usage, Skype employs a clever distribution
622 CHAPTER 7 MULTIMEDIA NETWORKING
Callee
peer
Caller
peer
Relay
peer
Super
peer
Skype
Skype Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Skype
Figure 7.10 Skype peers
technique. Specifically, each user sends its audio stream to the conference initiator. The
conference initiator combines the audio streams into one stream (basically by adding
all the audio signals together) and then sends a copy of each combined stream to each
of the other N 1 participants. In this manner, the number of streams is reduced to
2(N 1). For ordinary two-person video conversations, Skype routes the call peer-to-
peer, unless NAT traversal is required, in which case the call is relayed through a non-
NATed peer, as described earlier. For a video conference call involving N > 2
participants, due to the nature of the video medium, Skype does not combine the call
into one stream at one location and then redistribute the stream to all the participants,
as it does for voice calls. Instead, each participants video stream is routed to a server
cluster (located in Estonia as of 2011), which in turn relays to each participant the
N 1 streams of the N 1 other participants [Zhang X 2012]. You may be wonder-
ing why each participant sends a copy to a server rather than directly sending a copy of
its video stream to each of the other N 1 participants? Indeed, for both approaches,
N(N 1) video streams are being collectively received by the N participants in the
conference.The reason is, becauseupstream link bandwidths aresignificantly lower
than downstream link bandwidths in most access links, the upstream links may not be
able to support the N 1 streams with the P2P approach.
VoIP systems such as Skype, QQ, and Google Talk introduce new privacy
concerns. Specifically, when Alice and Bob communicate over VoIP, Alice can sniff
Bobs IP address and then use geo-location services [MaxMind 2012; Quova 2012]
to determine Bobs current location and ISP (for example, his work or home ISP). In
fact, with Skype it is possible for Alice to block the transmission of certain packets
during call establishment so that she obtains Bobs current IP address, say every hour,
without Bob knowing that he is being tracked and without being on Bobs contact
list. Furthermore, the IP address discovered from Skype can be correlated with IP
addresses found in BitTorrent, so that Alice can determine the files that Bob is down-
loading [LeBlond 2011]. Moreover, it is possible to partially decrypt a Skype call by
doing a traffic analysis of the packet sizes in a stream [White 2011].
7.4 Protocols for Real-Time Conversational
Applications
Real-time conversational applications, including VoIP and video conferencing, are
compelling and very popular. It is therefore not surprising that standards bodies,
such as the IETF and ITU, have been busy for many years (and continue to be
busy!) at hammering out standards for this class of applications. With the appropri-
ate standards in place for real-time conversational applications, independent compa-
nies are creating new products that interoperate with each other. In this section we
examine RTP and SIP for real-time conversational applications. Both standards are
enjoying widespread implementation in industry products.
7.4 PROTOCOLS FOR REAL-TIME CONVERSATIONAL APPLICATIONS 623
7.4.1 RTP
In the previous section, we learned that the sender side of a VoIP application appends
header fields to the audio chunks before passing them to the transport layer. These
header fields include sequence numbers and timestamps. Since most multimedia
networking applications can make use of sequence numbers and timestamps, it is con-
venient to have a standardized packet structure that includes fields for audio/video data,
sequence number, and timestamp, as well as other potentially useful fields. RTP,
defined in RFC 3550, is such a standard. RTP can be used for transporting common
formats such as PCM, ACC, and MP3 for sound and MPEG and H.263 for video. It can
also be used for transporting proprietary sound and video formats. Today, RTP enjoys
widespread implementation in many products and research prototypes. It is also com-
plementary to other important real-time interactive protocols, such as SIP.
In this section, we provide an introduction to RTP. We also encourage you to
visit Henning Schulzrinnes RTP site [Schulzrinne-RTP 2012], which provides a
wealth of information on the subject. Also, you may want to visit the RAT site
[RAT 2012], which documents VoIP application that uses RTP.
RTP Basics
RTP typically runs on top of UDP. The sending side encapsulates a media chunk
within an RTP packet, then encapsulates the packet in a UDP segment, and then
hands the segment to IP. The receiving side extracts the RTP packet from the UDP
segment, then extracts the media chunk from the RTP packet, and then passes the
chunk to the media player for decoding and rendering.
As an example, consider the use of RTP to transport voice. Suppose the voice
source is PCM-encoded (that is, sampled, quantized, and digitized) at 64 kbps. Fur-
ther suppose that the application collects the encoded data in 20-msec chunks, that is,
160 bytes in a chunk. The sending side precedes each chunk of the audio data with
an RTP header that includes the type of audio encoding, a sequence number, and a
timestamp. The RTP header is normally 12 bytes. The audio chunk along with the
RTP header form the RTP packet. The RTP packet is then sent into the UDP socket
interface. At the receiver side, the application receives the RTP packet from its socket
interface. The application extracts the audio chunk from the RTP packet and uses the
header fields of the RTP packet to properly decode and play back the audio chunk.
If an application incorporates RTPinstead of a proprietary scheme to provide
payload type, sequence numbers, or timestampsthen the application will more easily
interoperate with other networked multimedia applications. For example, if two differ-
ent companies develop VoIP software and they both incorporate RTP into their product,
there may be some hope that a user using one of the VoIP products will be able to com-
municate with a user using the other VoIP product. In Section 7.4.2, well see that RTP
is often used in conjunction with SIP, an important standard for Internet telephony.
It should be emphasized that RTP does not provide any mechanism to ensure
timely delivery of data or provide other quality-of-service (QoS) guarantees; it
624 CHAPTER 7 MULTIMEDIA NETWORKING
does not even guarantee delivery of packets or prevent out-of-order delivery of
packets. Indeed, RTP encapsulation is seen only at the end systems. Routers do
not distinguish between IP datagrams that carry RTP packets and IP datagrams
that dont.
RTP allows each source (for example, a camera or a microphone) to be assigned
its own independent RTP stream of packets. For example, for a video conference
between two participants, four RTP streams could be openedtwo streams for
transmitting the audio (one in each direction) and two streams for transmitting the
video (again, one in each direction). However, many popular encoding techniques
including MPEG 1 and MPEG 2bundle the audio and video into a single stream
during the encoding process. When the audio and video are bundled by the encoder,
then only one RTP stream is generated in each direction.
RTP packets are not limited to unicast applications. They can also be sent over
one-to-many and many-to-many multicast trees. For a many-to-many multicast
session, all of the sessions senders and sources typically use the same multicast
group for sending their RTP streams. RTP multicast streams belonging together,
such as audio and video streams emanating from multiple senders in a video confer-
ence application, belong to an RTP session.
RTP Packet Header Fields
As shown in Figure 7.11, the four main RTP packet header fields are the payload
type, sequence number, timestamp, and source identifier fields.
The payload type field in the RTP packet is 7 bits long. For an audio stream, the
payload type field is used to indicate the type of audio encoding (for example, PCM,
adaptive delta modulation, linear predictive encoding) that is being used. If a sender
decides to change the encoding in the middle of a session, the sender can inform the
receiver of the change through this payload type field. The sender may want to change
the encoding in order to increase the audio quality or to decrease the RTP stream bit
rate. Table 7.2 lists some of the audio payload types currently supported by RTP.
For a video stream, the payload type is used to indicate the type of video encoding
(for example, motion JPEG, MPEG 1, MPEG 2, H.261). Again, the sender can change
video encoding on the fly during a session. Table 7.3 lists some of the video payload
types currently supported by RTP. The other important fields are the following:
Sequence number field. The sequence number field is 16 bits long. The sequence
number increments by one for each RTP packet sent, and may be used by the
Payload
type
Sequence
number
Synchronization
source identifier
Miscellaneous
fields
Timestamp
Figure 7.11 RTP header fields
7.4 PROTOCOLS FOR REAL-TIME CONVERSATIONAL APPLICATIONS 625
receiver to detect packet loss and to restore packet sequence. For example, if the
receiver side of the application receives a stream of RTP packets with a gap
between sequence numbers 86 and 89, then the receiver knows that packets 87
and 88 are missing. The receiver can then attempt to conceal the lost data.
Timestamp field. The timestamp field is 32 bits long. It reflects the sampling
instant of the first byte in the RTP data packet. As we saw in the preceding
section, the receiver can use timestamps to remove packet jitter introduced in
the network and to provide synchronous playout at the receiver. The time-
stamp is derived from a sampling clock at the sender. As an example, for
audio the timestamp clock increments by one for each sampling period (for
example, each 125 sec for an 8 kHz sampling clock); if the audio applica-
tion generates chunks consisting of 160 encoded samples, then the timestamp
increases by 160 for each RTP packet when the source is active. The
626 CHAPTER 7 MULTIMEDIA NETWORKING
Payload-Type Number Audio Format Sampling Rate Rate
0 PCM -law 8 kHz 64 kbps
1 1016 8 kHz 4.8 kbps
3 GSM 8 kHz 13 kbps
7 LPC 8 kHz 2.4 kbps
9 G.722 16 kHz 4864 kbps
14 MPEG Audio 90 kHz
15 G.728 8 kHz 16 kbps
Table 7.2 Audio payload types supported by RTP
Payload-Type Number Video Format
26 Motion JPEG
31 H.261
32 MPEG 1 video
33 MPEG 2 video
Table 7.3 Some video payload types supported by RTP
timestamp clock continues to increase at a constant rate even if the source is
inactive.
Synchronization source identifier (SSRC). The SSRC field is 32 bits long. It iden-
tifies the source of the RTP stream. Typically, each stream in an RTP session has
a distinct SSRC. The SSRC is not the IP address of the sender, but instead is a
number that the source assigns randomly when the new stream is started. The
probability that two streams get assigned the same SSRC is very small. Should
this happen, the two sources pick a new SSRC value.
7.4.2 SIP
The Session Initiation Protocol (SIP), defined in [RFC 3261; RFC 5411], is an
open and lightweight protocol that does the following:
It provides mechanisms for establishing calls between a caller and a callee over
an IP network. It allows the caller to notify the callee that it wants to start a call.
It allows the participants to agree on media encodings. It also allows participants
to end calls.
It provides mechanisms for the caller to determine the current IP address of the
callee. Users do not have a single, fixed IP address because they may be assigned
addresses dynamically (using DHCP) and because they may have multiple IP
devices, each with a different IP address.
It provides mechanisms for call management, such as adding new media streams
during the call, changing the encoding during the call, inviting new participants
during the call, call transfer, and call holding.
Setting Up a Call to a Known IP Address
To understand the essence of SIP, it is best to take a look at a concrete example. In
this example, Alice is at her PC and she wants to call Bob, who is also working at
his PC. Alices and Bobs PCs are both equipped with SIP-based software for mak-
ing and receiving phone calls. In this initial example, well assume that Alice knows
the IP address of Bobs PC. Figure 7.12 illustrates the SIP call-establishment
process.
In Figure 7.12, we see that an SIP session begins when Alice sends Bob an
INVITE message, which resembles an HTTP request message. This INVITE mes-
sage is sent over UDP to the well-known port 5060 for SIP. (SIP messages can also
be sent over TCP.) The INVITE message includes an identifier for Bob
([email protected]), an indication of Alices current IP address, an indication that
Alice desires to receive audio, which is to be encoded in format AVP 0 (PCM
encoded -law) and encapsulated in RTP, and an indication that she wants to receive
7.4 PROTOCOLS FOR REAL-TIME CONVERSATIONAL APPLICATIONS 627
the RTP packets on port 38060. After receiving Alices INVITE message, Bob sends
an SIP response message, which resembles an HTTP response message. This
response SIP message is also sent to the SIP port 5060. Bobs response includes a
200 OK as well as an indication of his IP address, his desired encoding and packeti-
zation for reception, and his port number to which the audio packets should be sent.
Note that in this example Alice and Bob are going to use different audio-encoding
mechanisms: Alice is asked to encode her audio with GSM whereas Bob is asked to
encode his audio with PCM -law. After receiving Bobs response, Alice sends Bob
an SIP acknowledgment message. After this SIP transaction, Bob and Alice can talk.
(For visual convenience, Figure 7.12 shows Alice talking after Bob, but in truth they
628 CHAPTER 7 MULTIMEDIA NETWORKING
Time Time
167.180.112.24
INVITE [email protected]=IN IP4 167.180.112.24m=audio 38060 RTP/AVP 0
200 OK
c=In IP4 193
.64.210.89
m=audio 48
753 RTP/AV
P 3
Bobs
terminal rings
193.64.210.89
Law audio
port 5060
port 5060
port 38060
Alice Bob
port 5060
port 48753
ACK
GSM
Figure 7.12 SIP call establishment when Alice knows Bobs IP address
would normally talk at the same time.) Bob will encode and packetize the audio as
requested and send the audio packets to port number 38060 at IP address
167.180.112.24. Alice will also encode and packetize the audio as requested and
send the audio packets to port number 48753 at IP address 193.64.210.89.
From this simple example, we have learned a number of key characteristics
of SIP. First, SIP is an out-of-band protocol: The SIP messages are sent and
received in sockets that are different from those used for sending and receiving
the media data. Second, the SIP messages themselves are ASCII-readable and
resemble HTTP messages. Third, SIP requires all messages to be acknowledged,
so it can run over UDP or TCP.
In this example, lets consider what would happen if Bob does not have a
PCM -law codec for encoding audio. In this case, instead of responding with 200
OK, Bob would likely respond with a 600 Not Acceptable and list in the message
all the codecs he can use. Alice would then choose one of the listed codecs and
send another INVITE message, this time advertising the chosen codec. Bob could
also simply reject the call by sending one of many possible rejection reply codes.
(There are many such codes, including busy, gone, payment required, and
forbidden.)
SIP Addresses
In the previous example, Bobs SIP address is sip:[email protected]. However,
we expect manyif not mostSIP addresses to resemble e-mail addresses. For
example, Bobs address might be sip:[email protected]. When Alices SIP device
sends an INVITE message, the message would include this e-mail-like address;
the SIP infrastructure would then route the message to the IP device that Bob is
currently using (as well discuss below). Other possible forms for the SIP address
could be Bobs legacy phone number or simply Bobs first/middle/last name
(assuming it is unique).
An interesting feature of SIP addresses is that they can be included in Web
pages, just as peoples e-mail addresses are included in Web pages with the mailto
URL. For example, suppose Bob has a personal homepage, and he wants to pro-
vide a means for visitors to the homepage to call him. He could then simply
include the URL sip:[email protected]. When the visitor clicks on the URL, the
SIP application in the visitors device is launched and an INVITE message is sent
to Bob.
SIP Messages
In this short introduction to SIP, well not cover all SIP message types and headers.
Instead, well take a brief look at the SIP INVITE message, along with a few com-
mon header lines. Let us again suppose that Alice wants to initiate a VoIP call to
Bob, and this time Alice knows only Bobs SIP address, [email protected], and
7.4 PROTOCOLS FOR REAL-TIME CONVERSATIONAL APPLICATIONS 629
does not know the IP address of the device that Bob is currently using. Then her
message might look something like this:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 167.180.112.24
From: sip:[email protected]
To: sip:[email protected]
Call-ID: [email protected]
Content-Type: application/sdp
Content-Length: 885
c=IN IP4 167.180.112.24
m=audio 38060 RTP/AVP 0
The INVITE line includes the SIP version, as does an HTTP request message.
Whenever an SIP message passes through an SIP device (including the device that orig-
inates the message), it attaches a Via header, which indicates the IP address of the
device. (Well see soon that the typical INVITE message passes through many SIP
devices before reaching the callees SIP application.) Similar to an e-mail message, the
SIP message includes a From header line and a To header line. The message includes a
Call-ID, which uniquely identifies the call (similar to the message-ID in e-mail). It
includes a Content-Type header line, which defines the format used to describe the con-
tent contained in the SIP message. It also includes a Content-Length header line, which
provides the length in bytes of the content in the message. Finally, after a carriage return
and line feed, the message contains the content. In this case, the content provides infor-
mation about Alices IP address and how Alice wants to receive the audio.
Name Translation and User Location
In the example in Figure 7.12, we assumed that Alices SIP device knew the IP
address where Bob could be contacted. But this assumption is quite unrealistic, not
only because IP addresses are often dynamically assigned with DHCP, but also
because Bob may have multiple IP devices (for example, different devices for his
home, work, and car). So now let us suppose that Alice knows only Bobs e-mail
address, [email protected], and that this same address is used for SIP-based calls.
In this case, Alice needs to obtain the IP address of the device that the user
[email protected] is currently using. To find this out, Alice creates an INVITE mes-
sage that begins with INVITE [email protected] SIP/2.0 and sends this message to
an SIP proxy. The proxy will respond with an SIP reply that might include the IP
address of the device that [email protected] is currently using. Alternatively, the
reply might include the IP address of Bobs voicemail box, or it might include a
URL of a Web page (that says Bob is sleeping. Leave me alone!). Also, the result
returned by the proxy might depend on the caller: If the call is from Bobs wife, he
630 CHAPTER 7 MULTIMEDIA NETWORKING
might accept the call and supply his IP address; if the call is from Bobs mother-in-
law, he might respond with the URL that points to the I-am-sleeping Web page!
Now, you are probably wondering, how can the proxy server determine the cur-
rent IP address for [email protected]? To answer this question, we need to say a few
words about another SIP device, the SIP registrar. Every SIP user has an associated
registrar. Whenever a user launches an SIP application on a device, the application
sends an SIP register message to the registrar, informing the registrar of its current
IP address. For example, when Bob launches his SIP application on his PDA, the
application would send a message along the lines of:
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 193.64.210.89
From: sip:[email protected]
To: sip:[email protected]
Expires: 3600
Bobs registrar keeps track of Bobs current IP address. Whenever Bob switches
to a new SIP device, the new device sends a new register message, indicating the
new IP address. Also, if Bob remains at the same device for an extended period of
time, the device will send refresh register messages, indicating that the most
recently sent IP address is still valid. (In the example above, refresh messages need
to be sent every 3600 seconds to maintain the address at the registrar server.) It is
worth noting that the registrar is analogous to a DNS authoritative name server: The
DNS server translates fixed host names to fixed IP addresses; the SIP registrar trans-
lates fixed human identifiers (for example, [email protected]) to dynamic IP
addresses. Often SIP registrars and SIP proxies are run on the same host.
Now lets examine how Alices SIP proxy server obtains Bobs current IP
address. From the preceding discussion we see that the proxy server simply needs to
forward Alices INVITE message to Bobs registrar/proxy. The registrar/proxy
could then forward the message to Bobs current SIP device. Finally, Bob, having
now received Alices INVITE message, could send an SIP response to Alice.
As an example, consider Figure 7.13, in which [email protected], currently
working on 217.123.56.89, wants to initiate a Voice-over-IP (VoIP) session with
[email protected], currently working on 197.87.54.21. The following steps are
taken: (1) Jim sends an INVITE message to the umass SIP proxy. (2) The proxy
does a DNS lookup on the SIP registrar upenn.edu (not shown in diagram) and then
forwards the message to the registrar server. (3) Because [email protected] is no
longer registered at the upenn registrar, the upenn registrar sends a redirect response,
indicating that it should try [email protected]. (4) The umass proxy sends an
INVITE message to the eurecom SIP registrar. (5) The eurecom registrar knows the
IP address of [email protected] and forwards the INVITE message to the host
197.87.54.21, which is running Keiths SIP client. (68) An SIP response is sent back
through registrars/proxies to the SIP client on 217.123.56.89. (9) Media is sent
7.4 PROTOCOLS FOR REAL-TIME CONVERSATIONAL APPLICATIONS 631
directly between the two clients. (There is also an SIP acknowledgment message,
which is not shown.)
Our discussion of SIP has focused on call initiation for voice calls. SIP, being a
signaling protocol for initiating and ending calls in general, can be used for video
conference calls as well as for text-based sessions. In fact, SIP has become a funda-
mental component in many instant messaging applications. Readers desiring to
learn more about SIP are encouraged to visit Henning Schulzrinnes SIP Web site
[Schulzrinne-SIP 2012]. In particular, on this site you will find open source software
for SIP clients and servers [SIP Software 2012].
7.5 Network Support for Multimedia
In Sections 7.2 through 7.4, we learned how application-level mechanisms such as
client buffering, prefetching, adapting media quality to available bandwidth, adap-
tive playout, and loss mitigation techniques can be used by multimedia applications
632 CHAPTER 7 MULTIMEDIA NETWORKING
9
5
6
4
7
2
3
1
8
SIP registrar
upenn.edu
SIP proxy
umass.edu
SIP client
217.123.56.89
SIP client
197.87.54.21
SIP registrar
eurcom.fr
Figure 7.13 Session initiation, involving SIP proxies and registrars
to improve a multimedia applications performance. We also learned how content
distribution networks and P2P overlay networks can be used to provide a system-
level approach for delivering multimedia content. These techniques and approaches
are all designed to be used in todays best-effort Internet. Indeed, they are in use
today precisely because the Internet provides only a single, best-effort class of serv-
ice. But as designers of computer networks, we cant help but ask whether the
network (rather than the applications or application-level infrastructure alone) might
provide mechanisms to support multimedia content delivery. As well see shortly,
the answer is, of course, yes! But well also see that a number of these new net-
work-level mechanisms have yet to be widely deployed. This may be due to their
complexity and to the fact that application-level techniques together with best-effort
service and properly dimensioned network resources (for example, bandwidth) can
indeed provide a good-enough (even if not-always-perfect) end-to-end multimedia
delivery service.
Table 7.4 summarizes three broad approaches towards providing network-level
support for multimedia applications.
Making the best of best-effort service. The application-level mechanisms and
infrastructure that we studied in Sections 7.2 through 7.4 can be successfully
used in a well-dimensioned network where packet loss and excessive end-to-end
7.5 NETWORK SUPPORT FOR MULTIMEDIA 633
Approach Granularity Guarantee Mechanisms Complexity Deployment to date
Making the all traffic none, or application- minimal everywhere
best of best- treated soft layer support,
effort service. equally CDNs, overlays,
network-level
resource
provisioning
Differentiated different none, packet marking, medium some
service classes of or soft policing,
traffic scheduling
treated
differently
Per-connection each soft or hard, packet marking, light little
Quality-of- source- once flow policing, scheduling;
Service (QoS) destination is admitted call admission and
Guarantees flows treated signaling
differently
Table 7.4 Three network-level approaches to supporting multimedia
applications
delay rarely occur. When demand increases are forecasted, the ISPs deploy addi-
tional bandwidth and switching capacity to continue to ensure satisfactory delay
and packet-loss performance [Huang 2005]. Well discuss such network dimen-
sioning further in Section 7.5.1.
Differentiated service. Since the early days of the Internet, its been envisioned
that different types of traffic (for example, as indicated in the Type-of-Service
field in the IP4v packet header) could be provided with different classes of serv-
ice, rather than a single one-size-fits-all best-effort service. With differentiated
service, one type of traffic might be given strict priority over another class of
traffic when both types of traffic are queued at a router. For example, packets
belonging to a real-time conversational application might be given priority over
other packets due to their stringent delay constraints. Introducing differentiated
service into the network will require new mechanisms for packet marking (indi-
cating a packets class of service), packet scheduling, and more. Well cover dif-
ferentiated service, and new network mechanisms needed to implement this
service, in Section 7.5.2.
Per-connection Quality-of-Service (QoS) Guarantees. With per-connection
QoS guarantees, each instance of an application explicitly reserves end-to-end
bandwidth and thus has a guaranteed end-to-end performance. A hard guarantee
means the application will receive its requested quality of service (QoS) with
certainty. A soft guarantee means the application will receive its requested
quality of service with high probability. For example, if a user wants to make a
VoIP call from Host A to Host B, the users VoIP application reserves band-
width explicitly in each link along a route between the two hosts. But permit-
ting applications to make reservations and requiring the network to honor the
reservations requires some big changes. First, we need a protocol that, on
behalf of the applications, reserves link bandwidth on the paths from the
senders to their receivers. Second, well need new scheduling policies in the
router queues so that per-connection bandwidth reservations can be honored.
Finally, in order to make a reservation, the applications must give the network
a description of the traffic that they intend to send into the network and the net-
work will need to police each applications traffic to make sure that it abides
by that description. These mechanisms, when combined, require new and com-
plex software in hosts and routers. Because per-connection QoS guaranteed
service has not seen significant deployment, well cover these mechanisms
only briefly in Section 7.5.3.
7.5.1 Dimensioning Best-Effort Networks
Fundamentally, the difficulty in supporting multimedia applications arises from
their stringent performance requirementslow end-to-end packet delay, delay
634 CHAPTER 7 MULTIMEDIA NETWORKING
jitter, and lossand the fact that packet delay, delay jitter, and loss occur when-
ever the network becomes congested. A first approach to improving the quality
of multimedia applicationsan approach that can often be used to solve just
about any problem where resources are constrainedis simply to throw money
at the problem and thus simply avoid resource contention. In the case of net-
worked multimedia, this means providing enough link capacity throughout the
network so that network congestion, and its consequent packet delay and loss,
never (or only very rarely) occurs. With enough link capacity, packets could zip
through todays Internet without queuing delay or loss. From many perspectives
this is an ideal situationmultimedia applications would perform perfectly, users
would be happy, and this could all be achieved with no changes to Internets best-
effort architecture.
The question, of course, is how much capacity is enough to achieve this
nirvana, and whether the costs of providing enough bandwidth are practical
from a business standpoint to the ISPs. The question of how much capacity to
provide at network links in a given topology to achieve a given level of perform-
ance is often known as bandwidth provisioning. The even more complicated
problem of how to design a network topology (where to place routers, how to
interconnect routers with links, and what capacity to assign to links) to achieve a
given level of end-to-end performance is a network design problem often referred
to as network dimensioning. Both bandwidth provisioning and network dimen-
sioning are complex topics, well beyond the scope of this textbook. We note here,
however, that the following issues must be addressed in order to predict applica-
tion-level performance between two network end points, and thus provision
enough capacity to meet an applications performance requirements.
Models of traffic demand between network end points. Models may need to be
specified at both the call level (for example, users arriving to the network and
starting up end-to-end applications) and at the packet level (for example, packets
being generated by ongoing applications). Note that workload may change over
time.
Well-defined performance requirements. For example, a performance require-
ment for supporting delay-sensitive traffic, such as a conversational multimedia
application, might be that the probability that the end-to-end delay of the packet
is greater than a maximum tolerable delay be less than some small value
[Fraleigh 2003].
Models to predict end-to-end performance for a given workload model, and tech-
niques to find a minimal cost bandwidth allocation that will result in all user
requirements being met. Here, researchers are busy developing performance
models that can quantify performance for a given workload, and optimization
techniques to find minimal-cost bandwidth allocations meeting performance
requirements.
7.5 NETWORK SUPPORT FOR MULTIMEDIA 635
Given that todays best-effort Internet could (from a technology standpoint)
support multimedia traffic at an appropriate performance level if it were dimen-
sioned to do so, the natural question is why todays Internet doesnt do so. The
answers are primarily economic and organizational. From an economic standpoint,
would users be willing to pay their ISPs enough for the ISPs to install sufficient
bandwidth to support multimedia applications over a best-effort Internet? The orga-
nizational issues are perhaps even more daunting. Note that an end-to-end path
between two multimedia end points will pass through the networks of multiple ISPs.
From an organizational standpoint, would these ISPs be willing to cooperate
(perhaps with revenue sharing) to ensure that the end-to-end path is properly dimen-
sioned to support multimedia applications? For a perspective on these economic and
organizational issues, see [Davies 2005]. For a perspective on provisioning tier-1
backbone networks to support delay-sensitive traffic, see [Fraleigh 2003].
7.5.2 Providing Multiple Classes of Service
Perhaps the simplest enhancement to the one-size-fits-all best-effort service in
todays Internet is to divide traffic into classes, and provide different levels of serv-
ice to these different classes of traffic. For example, an ISP might well want to pro-
vide a higher class of service to delay-sensitive Voice-over-IP or teleconferencing
traffic (and charge more for this service!) than to elastic traffic such as email or
HTTP. Alternatively, an ISP may simply want to provide a higher quality of service
to customers willing to pay more for this improved service. A number of residential
wired-access ISPs and cellular wireless-access ISPs have adopted such tiered levels
of servicewith platinum-service subscribers receiving better performance than
gold- or silver-service subscribers.
Were all familiar with different classes of service from our everyday lives
first-class airline passengers get better service than business-class passengers, who
in turn get better service than those of us who fly economy class; VIPs are provided
immediate entry to events while everyone else waits in line; elders are revered in
some countries and provided seats of honor and the finest food at a table. Its impor-
tant to note that such differential service is provided among aggregates of traffic,
that is, among classes of traffic, not among individual connections. For example, all
first-class passengers are handled the same (with no first-class passenger receiving
any better treatment than any other first-class passenger), just as all VoIP packets
would receive the same treatment within the network, independent of the particular
end-to-end connection to which they belong. As we will see, by dealing with a small
number of traffic aggregates, rather than a large number of individual connections,
the new network mechanisms required to provide better-than-best service can be
kept relatively simple.
The early Internet designers clearly had this notion of multiple classes of serv-
ice in mind. Recall the type-of-service (ToS) field in the IPv4 header in Figure 4.13.
636 CHAPTER 7 MULTIMEDIA NETWORKING
IEN123 [ISI 1979] describes the ToS field also present in an ancestor of the IPv4
datagram as follows: The Type of Service [field] provides an indication of the
abstract parameters of the quality of service desired. These parameters are to be used
to guide the selection of the actual service parameters when transmitting a datagram
through a particular network. Several networks offer service precedence, which
somehow treats high precedence traffic as more important that other traffic. More
than four decades ago, the vision of providing different levels of service to different
classes of traffic was clear! However, its taken us an equally long period of time to
realize this vision.
Motivating Scenarios
Lets begin our discussion of network mechanisms for providing multiple classes of
service with a few motivating scenarios.
Figure 7.14 shows a simple network scenario in which two application packet
flows originate on Hosts H1 and H2 on one LAN and are destined for Hosts H3 and
H4 on another LAN. The routers on the two LANs are connected by a 1.5 Mbps
link. Lets assume the LAN speeds are significantly higher than 1.5 Mbps, and focus
on the output queue of router R1; it is here that packet delay and packet loss will
occur if the aggregate sending rate of H1 and H2 exceeds 1.5 Mbps. Lets further
suppose that a 1 Mbps audio application (for example, a CD-quality audio call)
shares the 1.5 Mbps link between R1 and R2 with an HTTP Web-browsing applica-
tion that is downloading a Web page from H2 to H4.
R1
1.5 Mbps link R2
H2
H1
H4
H3
Figure 7.14 Competing audio and HTTP applications
7.5 NETWORK SUPPORT FOR MULTIMEDIA 637
In the best-effort Internet, the audio and HTTP packets are mixed in the out-
put queue at R1 and (typically) transmitted in a first-in-first-out (FIFO) order. In
this scenario, a burst of packets from the Web server could potentially fill up the
queue, causing IP audio packets to be excessively delayed or lost due to buffer
overflow at R1. How should we solve this potential problem? Given that the
HTTP Web-browsing application does not have time constraints, our intuition
might be to give strict priority to audio packets at R1. Under a strict priority
scheduling discipline, an audio packet in the R1 output buffer would always be
transmitted before any HTTP packet in the R1 output buffer. The link from R1 to
R2 would look like a dedicated link of 1.5 Mbps to the audio traffic, with HTTP
traffic using the R1-to-R2 link only when no audio traffic is queued. In order for
R1 to distinguish between the audio and HTTP packets in its queue, each packet
must be marked as belonging to one of these two classes of traffic. This was the
original goal of the type-of-service (ToS) field in IPv4. As obvious as this might
seem, this then is our first insight into mechanisms needed to provide multiple
classes of traffic:
Insight 1: Packet marking allows a router to distinguish among packets
belonging to different classes of traffic.
Note that although our example considers a competing multimedia and elastic
flow, the same insight applies to the case that platinum, gold, and silver classes of
service are implementeda packet-marking mechanism is still needed to indicate
that class of service to which a packet belongs.
Now suppose that the router is configured to give priority to packets marked as
belonging to the 1 Mbps audio application. Since the outgoing link speed is
1.5 Mbps, even though the HTTP packets receive lower priority, they can still, on
average, receive 0.5 Mbps of transmission service. But what happens if the audio
application starts sending packets at a rate of 1.5 Mbps or higher (either maliciously
or due to an error in the application)? In this case, the HTTP packets will starve, that
is, they will not receive any service on the R1-to-R2 link. Similar problems would
occur if multiple applications (for example, multiple audio calls), all with the same
class of service as the audio application, were sharing the links bandwidth; they too
could collectively starve the FTP session. Ideally, one wants a degree of isolation
among classes of traffic so that one class of traffic can be protected from the other.
This protection could be implemented at different places in the networkat each
and every router, at first entry to the network, or at inter-domain network bound-
aries. This then is our second insight:
Insight 2: It is desirable to provide a degree of traffic isolation among classes
so that one class is not adversely affected by another class of traffic that misbe-
haves.
638 CHAPTER 7 MULTIMEDIA NETWORKING
Well examine several specific mechanisms for providing such isolation
among traffic classes. We note here that two broad approaches can be taken.
First, it is possible to perform traffic policing, as shown in Figure 7.15. If a traf-
fic class or flow must meet certain criteria (for example, that the audio flow not
exceed a peak rate of 1 Mbps), then a policing mechanism can be put into place
to ensure that these criteria are indeed observed. If the policed application mis-
behaves, the policing mechanism will take some action (for example, drop or
delay packets that are in violation of the criteria) so that the traffic actually enter-
ing the network conforms to the criteria. The leaky bucket mechanism that well
examine shortly is perhaps the most widely used policing mechanism. In Figure
7.15, the packet classification and marking mechanism (Insight 1) and the polic-
ing mechanism (Insight 2) are both implemented together at the networks edge,
either in the end system or at an edge router.
A complementary approach for providing isolation among traffic classes is
for the link-level packet-scheduling mechanism to explicitly allocate a fixed
R1
1.5 Mbps link
Packet marking
and policing
Metering and policing Marks
R2
H2
H1
Key:
H4
H3
Figure 7.15 Policing (and marking) the audio and HTTP traffic classes
7.5 NETWORK SUPPORT FOR MULTIMEDIA 639
amount of link bandwidth to each class. For example, the audio class could be
allocated 1 Mbps at R1, and the HTTP class could be allocated 0.5 Mbps. In this
case, the audio and HTTP flows see a logical link with capacity 1.0 and 0.5
Mbps, respectively, as shown in Figure 7.16. With strict enforcement of the link-
level allocation of bandwidth, a class can use only the amount of bandwidth that
has been allocated; in particular, it cannot utilize bandwidth that is not currently
being used by others. For example, if the audio flow goes silent (for example, if
the speaker pauses and generates no audio packets), the HTTP flow would still
not be able to transmit more than 0.5 Mbps over the R1-to-R2 link, even though
the audio flows 1 Mbps bandwidth allocation is not being used at that moment.
Since bandwidth is a use-it-or-lose-it resource, there is no reason to prevent
HTTP traffic from using bandwidth not used by the audio traffic. Wed like to use
bandwidth as efficiently as possible, never wasting it when it could be otherwise
used. This gives rise to our third insight:
Insight 3: While providing isolation among classes or flows, it is desirable
to use resources (for example, link bandwidth and buffers) as efficiently as
possible.
Scheduling Mechanisms
Recall from our discussion in Section 1.3 and Section 4.3 that packets belonging
to various network flows are multiplexed and queued for transmission at the
640 CHAPTER 7 MULTIMEDIA NETWORKING
R1
1.5 Mbps link
1.0 Mbps
logical link
0.5 Mbps
logical link
R2
H2
H1
H4
H3
Figure 7.16 Logical isolation of audio and HTTP traffic classes
output buffers associated with a link. The manner in which queued packets are
selected for transmission on the link is known as the link-scheduling discipline.
Let us now consider several of the most important link-scheduling disciplines in
more detail.
First-In-First-Out (FIFO)
Figure 7.17 shows the queuing model abstractions for the FIFO link-scheduling dis-
cipline. Packets arriving at the link output queue wait for transmission if the link is
currently busy transmitting another packet. If there is not sufficient buffering space
to hold the arriving packet, the queues packet-discarding policy then determines
whether the packet will be dropped (lost) or whether other packets will be removed
from the queue to make space for the arriving packet. In our discussion below, we
will ignore packet discard. When a packet is completely transmitted over the out-
going link (that is, receives service) it is removed from the queue.
The FIFO (also known as first-come-first-served, or FCFS) scheduling disci-
pline selects packets for link transmission in the same order in which they arrived at
the output link queue. Were all familiar with FIFO queuing from bus stops (partic-
ularly in England, where queuing seems to have been perfected) or other service
centers, where arriving customers join the back of the single waiting line, remain in
order, and are then served when they reach the front of the line.
Figure 7.18 shows the FIFO queue in operation. Packet arrivals are indicated
by numbered arrows above the upper timeline, with the number indicating the order
R1
1.5 Mbps link
R1 output
interface queue
R2
H2
H1
H4
H3
Figure 7.17 FIFO queuing abstraction
7.5 NETWORK SUPPORT FOR MULTIMEDIA 641
in which the packet arrived. Individual packet departures are shown below the lower
timeline. The time that a packet spends in service (being transmitted) is indicated by
the shaded rectangle between the two timelines. Because of the FIFO discipline,
packets leave in the same order in which they arrived. Note that after the departure
of packet 4, the link remains idle (since packets 1 through 4 have been transmitted
and removed from the queue) until the arrival of packet 5.
Priority Queuing
Under priority queuing, packets arriving at the output link are classified into priority
classes at the output queue, as shown in Figure 7.19. As discussed in the previous sec-
tion, a packets priority class may depend on an explicit marking that it carries in its
packet header (for example, the value of the ToS bits in an IPv4 packet), its source or
destination IP address, its destination port number, or other criteria. Each priority class
typically has its own queue. When choosing a packet to transmit, the priority queuing
discipline will transmit a packet from the highest priority class that has a nonempty
queue (that is, has packets waiting for transmission). The choice among packets in the
same priority class is typically done in a FIFO manner.
Figure 7.20 illustrates the operation of a priority queue with two priority
classes. Packets 1, 3, and 4 belong to the high-priority class, and packets 2 and 5
belong to the low-priority class. Packet 1 arrives and, finding the link idle, begins
transmission. During the transmission of packet 1, packets 2 and 3 arrive and are
queued in the low- and high-priority queues, respectively. After the transmission
of packet 1, packet 3 (a high-priority packet) is selected for transmission over
packet 2 (which, even though it arrived earlier, is a low-priority packet). At the end
of the transmission of packet 3, packet 2 then begins transmission. Packet 4 (a
high-priority packet) arrives during the transmission of packet 2 (a low-priority
packet). Under a nonpreemptive priority queuing discipline, the transmission of
642 CHAPTER 7 MULTIMEDIA NETWORKING
Time
Arrivals
Departures
Packet
in service
Time
1
1 2 3 4 5
2 3
1
t = 0 t = 2 t = 4 t = 6 t = 8 t = 10 t = 12 t = 14
2 3 4 5
4 5
Figure 7.18 The FIFO queue in operation
a packet is not interrupted once it has begun. In this case, packet 4 queues for
transmission and begins being transmitted after the transmission of packet 2 is
completed.
Round Robin and Weighted Fair Queuing (WFQ)
Under the round robin queuing discipline, packets are sorted into classes as
with priority queuing. However, rather than there being a strict priority of service
among classes, a round robin scheduler alternates service among the classes. In
the simplest form of round robin scheduling, a class 1 packet is transmitted, fol-
lowed by a class 2 packet, followed by a class 1 packet, followed by a class 2
packet, and so on. A so-called work-conserving queuing discipline will never
allow the link to remain idle whenever there are packets (of any class) queued for
Arrivals
Departures
Packet
in service
1
1 23 4 5
2 3
1 23 4 5
4 5
Time
Time
t = 0 t = 2 t = 4 t = 6 t = 8 t = 10 t = 12 t = 14
Figure 7.20 Operation of the priority queue
Arrivals Departures
Low-priority queue
(waiting area)
Classify
High-priority queue
(waiting area)
Link
(server)
Figure 7.19 Priority queuing model
7.5 NETWORK SUPPORT FOR MULTIMEDIA 643
transmission. A work-conserving round robin discipline that looks for a packet
of a given class but finds none will immediately check the next class in the round
robin sequence.
Figure 7.21 illustrates the operation of a two-class round robin queue. In
this example, packets 1, 2, and 4 belong to class 1, and packets 3 and 5 belong to the
second class. Packet 1 begins transmission immediately upon arrival at the output
queue. Packets 2 and 3 arrive during the transmission of packet 1 and thus queue for
transmission. After the transmission of packet 1, the link scheduler looks for a class
2 packet and thus transmits packet 3. After the transmission of packet 3, the sched-
uler looks for a class 1 packet and thus transmits packet 2. After the transmission of
packet 2, packet 4 is the only queued packet; it is thus transmitted immediately after
packet 2.
A generalized abstraction of round robin queuing that has found considerable
use in QoS architectures is the so-called weighted fair queuing (WFQ) discipline
[Demers 1990; Parekh 1993]. WFQ is illustrated in Figure 7.22. Arriving packets
are classified and queued in the appropriate per-class waiting area. As in round robin
scheduling, a WFQ scheduler will serve classes in a circular mannerfirst serving
class 1, then serving class 2, then serving class 3, and then (assuming there are three
classes) repeating the service pattern. WFQ is also a work-conserving queuing
discipline and thus will immediately move on to the next class in the service
sequence when it finds an empty class queue.
WFQ differs from round robin in that each class may receive a differential
amount of service in any interval of time. Specifically, each class, i, is assigned a
weight, wi. Under WFQ, during any interval of time during which there are class i
packets to send, class i will then be guaranteed to receive a fraction of service equal
to wi/(wj), where the sum in the denominator is taken over all classes that also have
packets queued for transmission. In the worst case, even if all classes have queued
packets, class i will still be guaranteed to receive a fraction wi /(wj) of the
644 CHAPTER 7 MULTIMEDIA NETWORKING
Arrivals
Packet
in service
1
1 23 4 5
2 3
1 23 4 5
4 5
Departures
Time
Time
t = 0 t = 2 t = 4 t = 6 t = 8 t = 10 t = 12 t = 14
Figure 7.21 Operation of the two-class round robin queue
bandwidth. Thus, for a link with transmission rate R, class i will always achieve a
throughput of at least R wi /(wj). Our description of WFQ has been an idealized
one, as we have not considered the fact that packets are discrete units of data and a
packets transmission will not be interrupted to begin transmission of another
packet; [Demers 1990] and [Parekh 1993] discuss this packetization issue. As we
will see in the following sections, WFQ plays a central role in QoS architectures. It
is also available in todays router products [Cisco QoS 2012].
Policing: The Leaky Bucket
One of our earlier insights was that policing, the regulation of the rate at which a
class or flow (we will assume the unit of policing is a flow in our discussion below)
is allowed to inject packets into the network, is an important QoS mechanism. But
what aspects of a flows packet rate should be policed? We can identify three impor-
tant policing criteria, each differing from the other according to the time scale over
which the packet flow is policed:
Average rate. The network may wish to limit the long-term average rate (packets
per time interval) at which a flows packets can be sent into the network. A
crucial issue here is the interval of time over which the average rate will be
policed. A flow whose average rate is limited to 100 packets per second is
more constrained than a source that is limited to 6,000 packets per minute, even
though both have the same average rate over a long enough interval of time. For
example, the latter constraint would allow a flow to send 1,000 packets in a given
second-long interval of time, while the former constraint would disallow this
sending behavior.
Classify
Arrivals Departures
w1
w2
w3
Link
Figure 7.22 Weighted fair queuing (WFQ)
7.5 NETWORK SUPPORT FOR MULTIMEDIA 645
Peak rate. While the average-rate constraint limits the amount of traffic that can
be sent into the network over a relatively long period of time, a peak-rate con-
straint limits the maximum number of packets that can be sent over a shorter
period of time. Using our example above, the network may police a flow at an
average rate of 6,000 packets per minute, while limiting the flows peak rate to
1,500 packets per second.
Burst size. The network may also wish to limit the maximum number of packets
(the burst of packets) that can be sent into the network over an extremely short
interval of time. In the limit, as the interval length approaches zero, the burst size
limits the number of packets that can be instantaneously sent into the network.
Even though it is physically impossible to instantaneously send multiple packets
into the network (after all, every link has a physical transmission rate that cannot
be exceeded!), the abstraction of a maximum burst size is a useful one.
The leaky bucket mechanism is an abstraction that can be used to characterize
these policing limits. As shown in Figure 7.23, a leaky bucket consists of a bucket
that can hold up to b tokens. Tokens are added to this bucket as follows. New tokens,
which may potentially be added to the bucket, are always being generated at a rate
of r tokens per second. (We assume here for simplicity that the unit of time is a sec-
ond.) If the bucket is filled with less than b tokens when a token is generated, the
newly generated token is added to the bucket; otherwise the newly generated token
is ignored, and the token bucket remains full with b tokens.
Let us now consider how the leaky bucket can be used to police a packet flow.
Suppose that before a packet is transmitted into the network, it must first remove a
646 CHAPTER 7 MULTIMEDIA NETWORKING
To network
Packets
Remove
token
Token
wait area
Bucket holds
up to
b tokens
r tokens/sec
Figure 7.23 The leaky bucket policer
token from the token bucket. If the token bucket is empty, the packet must wait for
a token. (An alternative is for the packet to be dropped, although we will not consider
that option here.) Let us now consider how this behavior polices a traffic flow. Because
there can be at most b tokens in the bucket, the maximum burst size for a leaky-bucket-
policed flow is b packets. Furthermore, because the token generation rate is r, the max-
imum number of packets that can enter the network of any interval of time of length t
is rt + b. Thus, the token-generation rate, r, serves to limit the long-term average rate
at which packets can enter the network. It is also possible to use leaky buckets (specif-
ically, two leaky buckets in series) to police a flows peak rate in addition to the long-
term average rate; see the homework problems at the end of this chapter.
Leaky Bucket + Weighted Fair Queuing = Provable Maximum Delay in a
Queue
Lets close our discussion of scheduling and policing by showing how the two can
be combined to provide a bound on the delay through a routers queue. Lets con-
sider a routers output link that multiplexes n flows, each policed by a leaky bucket
with parameters bi and ri, i = 1, . . . , n, using WFQ scheduling. We use the term flow
here loosely to refer to the set of packets that are not distinguished from each other
by the scheduler. In practice, a flow might be comprised of traffic from a single end-
to-end connection or a collection of many such connections, see Figure 7.24.
Recall from our discussion of WFQ that each flow, i, is guaranteed to receive a
share of the link bandwidth equal to at least R wi/(wj), where R is the transmission
b1
r1
w1
wn
bn
rn
Figure 7.24 n multiplexed leaky bucket flows with WFQ scheduling
7.5 NETWORK SUPPORT FOR MULTIMEDIA 647
rate of the link in packets/sec. What then is the maximum delay that a packet will
experience while waiting for service in the WFQ (that is, after passing through the
leaky bucket)? Let us focus on flow 1. Suppose that flow 1s token bucket is initially
full. A burst of b1 packets then arrives to the leaky bucket policer for flow 1. These
packets remove all of the tokens (without wait) from the leaky bucket and then join
the WFQ waiting area for flow 1. Since these b1 packets are served at a rate of at least
R wi /(wj) packet/sec, the last of these packets will then have a maximum delay,
dmax, until its transmission is completed, where
The rationale behind this formula is that if there are b1 packets in the queue and
packets are being serviced (removed) from the queue at a rate of at least R w1/
(wj) packets per second, then the amount of time until the last bit of the last packet
is transmitted cannot be more than b1/(R w1/(wj)). A homework problem asks you
to prove that as long as r1 < R w1/(wj), then dmax is indeed the maximum delaythat any packet in flow 1 will ever experience in the WFQ queue.7.5.3 Diffserv Having seen the motivation, insights, and specific mechanisms for providing multi-ple classes of service, lets wrap up our study of approaches toward proving multi-ple classes of service with an examplethe Internet Diffserv architecture [RFC2475; RFC Kilkki 1999]. Diffserv provides service differentiationthat is, the abil-ity to handle different classes of traffic in different ways within the Internet in a scal-able manner. The need for scalability arises from the fact that millions ofsimultaneous source-destination traffic flows may be present at a backbone router.Well see shortly that this need is met by placing only simple functionality withinthe network core, with more complex control operations being implemented at thenetworks edge.Lets begin with the simple network shown in Figure 7.25. Well describe onepossible use of Diffserv here; other variations are possible, as described in RFC2475. The Diffserv architecture consists of two sets of functional elements: Edge functions: packet classification and traffic conditioning. At the incom-ing edge of the network (that is, at either a Diffserv-capable host that generatestraffic or at the first Diffserv-capable router that the traffic passes through), arriv-ing packets are marked. More specifically, the differentiated service (DS) field inthe IPv4 or IPv6 packet header is set to some value [RFC 3260]. The definitionof the DS field is intended to supersede the earlier definitions of the IPv4 type-of-service field and the IPv6 traffic class fields that we discussed in Chapter 4.For example, in Figure 7.25, packets being sent from H1 to H3 might be markeddmax =b1R w1>gwj
648 CHAPTER 7 MULTIMEDIA NETWORKING
at R1, while packets being sent from H2 to H4 might be marked at R2. The mark
that a packet receives identifies the class of traffic to which it belongs. Different
classes of traffic will then receive different service within the core network.
Core function: forwarding. When a DS-marked packet arrives at a Diffserv-
capable router, the packet is forwarded onto its next hop according to the so-called
per-hop behavior (PHB) associated with that packets class. The per-hop behavior
influences how a routers buffers and link bandwidth are shared among the compet-
ing classes of traffic. A crucial tenet of the Diffserv architecture is that a routers per-
hop behavior will be based only on packet markings, that is, the class of traffic to
which a packet belongs. Thus, if packets being sent from H1 to H3 in Figure 7.25
receive the same marking as packets being sent from H2 to H4, then the network
routers treat these packets as an aggregate, without distinguishing whether the pack-
ets originated at H1 or H2. For example, R3 would not distinguish between packets
from H1 and H2 when forwarding these packets on to R4. Thus, the Diffserv archi-
tecture obviates the need to keep router state for individual source-destination
pairsa critical consideration in making Diffserv scalable.
An analogy might prove useful here. At many large-scale social events (for example, a
large public reception, a large dance club or discothque, a concert, or a football game),
people entering the event receive a pass of one type or another: VIP passes for Very
7.5 NETWORK SUPPORT FOR MULTIMEDIA 649
R4
Leaf router
Key:
Core router
R2
R1 R6
R7
R3 R5
H1
H2
H4
H3
R2 R3
Figure 7.25 A simple Diffserv network example
Important People; over-21 passes for people who are 21 years old or older (for exam-
ple, if alcoholic drinks are to be served); backstage passes at concerts; press passes for
reporters; even an ordinary pass for the Ordinary Person. These passes are typically dis-
tributed upon entry to the event, that is, at the edge of the event. It is here at the edge
where computationally intensive operations, such as paying for entry, checking for the
appropriate type of invitation, and matching an invitation against a piece of identifica-
tion, are performed. Furthermore, there may be a limit on the number of people of a
given type that are allowed into an event. If there is such a limit, people may have to
wait before entering the event. Once inside the event, ones pass allows one to receive
differentiated service at many locations around the eventa VIP is provided with free
drinks, a better table, free food, entry to exclusive rooms, and fawning service. Con-
versely, an ordinary person is excluded from certain areas, pays for drinks, and receives
only basic service. In both cases, the service received within the event depends solely
on the type of ones pass. Moreover, all people within a class are treated alike.
Figure 7.26 provides a logical view of the classification and marking functions
within the edge router. Packets arriving to the edge router are first classified. The
classifier selects packets based on the values of one or more packet header fields
(for example, source address, destination address, source port, destination port, and
protocol ID) and steers the packet to the appropriate marking function. As noted
above, a packets marking is carried in the DS field in the packet header.
In some cases, an end user may have agreed to limit its packet-sending rate to con-
form to a declared traffic profile. The traffic profile might contain a limit on the peak
rate, as well as the burstiness of the packet flow, as we saw previously with the leaky
bucket mechanism. As long as the user sends packets into the network in a way that
conforms to the negotiated traffic profile, the packets receive their priority marking and
are forwarded along their route to the destination. On the other hand, if the traffic pro-
file is violated, out-of-profile packets might be marked differently, might be shaped (for
example, delayed so that a maximum rate constraint would be observed), or might be
dropped at the network edge. The role of the metering function, shown in Figure 7.26,
is to compare the incoming packet flow with the negotiated traffic profile and to deter-
mine whether a packet is within the negotiated traffic profile. The actual decision about
whether to immediately remark, forward, delay, or drop a packet is a policy issue deter-
mined by the network administrator and is not specified in the Diffserv architecture.
So far, we have focused on the marking and policing functions in the Diffserv
architecture. The second key component of the Diffserv architecture involves the
per-hop behavior (PHB) performed by Diffserv-capable routers. PHB is rather cryp-
tically, but carefully, defined as a description of the externally observable forward-
ing behavior of a Diffserv node applied to a particular Diffserv behavior aggregate
[RFC 2475]. Digging a little deeper into this definition, we can see several impor-
tant considerations embedded within:
A PHB can result in different classes of traffic receiving different performance
(that is, different externally observable forwarding behaviors).
650 CHAPTER 7 MULTIMEDIA NETWORKING
While a PHB defines differences in performance (behavior) among classes, it
does not mandate any particular mechanism for achieving these behaviors. As
long as the externally observable performance criteria are met, any implementa-
tion mechanism and any buffer/bandwidth allocation policy can be used. For
example, a PHB would not require that a particular packet-queuing discipline
(for example, a priority queue versus a WFQ queue versus a FCFS queue) be
used to achieve a particular behavior. The PHB is the end, to which resource allo-
cation and implementation mechanisms are the means.
Differences in performance must be observable and hence measurable.
Two PHBs have been defined: an expedited forwarding (EF) PHB [RFC 3246] and
an assured forwarding (AF) PHB [RFC 2597]. The expedited forwarding PHB
specifies that the departure rate of a class of traffic from a router must equal or
exceed a configured rate. The assured forwarding PHB divides traffic into four
classes, where each AF class is guaranteed to be provided with some minimum
amount of bandwidth and buffering.
Lets close our discussion of Diffserv with a few observations regarding its
service model. First, we have implicitly assumed that Diffserv is deployed within a
single administrative domain, but typically an end-to-end service must be fashioned
from multiple ISPs sitting between communicating end systems. In order to provide
end-to-end Diffserv service, all the ISPs between the end systems must not only pro-
vide this service, but most also cooperate and make settlements in order to offer end
customers true end-to-end service. Without this kind of cooperation, ISPs directly
selling Diffserv service to customers will find themselves repeatedly saying: Yes,
we know you paid extra, but we dont have a service agreement with the ISP that
dropped and delayed your traffic. Im sorry that there were so many gaps in your
Packets Forward
Classifier Marker
Drop
Shaper/
Dropper
Meter
Figure 7.26 A simple Diffserv network example
7.5 NETWORK SUPPORT FOR MULTIMEDIA 651
VoIP call! Second, if Diffserv were actually in place and the network ran at only
moderate load, most of the time there would be no perceived difference between a
best-effort service and a Diffserv service. Indeed, end-to-end delay is usually domi-
nated by access rates and router hops rather than by queuing delays in the routers.
Imagine the unhappy Diffserv customer who has paid more for premium service but
finds that the best-effort service being provided to others almost always has the
same performance as premium service!
7.5.4 Per-Connection Quality-of-Service (QoS) Guarantees:
Resource Reservation and Call Admission
In the previous section, we have seen that packet marking and policing, traffic isola-
tion, and link-level scheduling can provide one class of service with better perform-
ance than another. Under certain scheduling disciplines, such as priority scheduling,
the lower classes of traffic are essentially invisible to the highest-priority class of
traffic. With proper network dimensioning, the highest class of service can indeed
achieve extremely low packet loss and delayessentially circuit-like performance.
But can the network guarantee that an ongoing flow in a high-priority traffic class
will continue to receive such service throughout the flows duration using only the
mechanisms that we have described so far? It cannot. In this section, well see why
yet additional network mechanisms and protocols are required when a hard service
guarantee is provided to individual connections.
Lets return to our scenario from Section 7.5.2 and consider two 1 Mbps
audio applications transmitting their packets over the 1.5 Mbps link, as shown in
Figure 7.27. The combined data rate of the two flows (2 Mbps) exceeds the link
652 CHAPTER 7 MULTIMEDIA NETWORKING
R1
1.5 Mbps link
1 Mbps
audio
1 Mbps
audio
R2
H2
H1
H4
H3
Figure 7.27 Two competing audio applications overloading
the R1-to-R2 link
capacity. Even with classification and marking, isolation of flows, and sharing of
unused bandwidth (of which there is none), this is clearly a losing proposition.
There is simply not enough bandwidth to accommodate the needs of both appli-
cations at the same time. If the two applications equally share the bandwidth,
each application would lose 25 percent of its transmitted packets. This is such an
unacceptably low QoS that both audio applications are completely unusable;
theres no need even to transmit any audio packets in the first place.
Given that the two applications in Figure 7.27 cannot both be satisfied simulta-
neously, what should the network do? Allowing both to proceed with an unusable
QoS wastes network resources on application flows that ultimately provide no util-
ity to the end user. The answer is hopefully clearone of the application flows
should be blocked (that is, denied access to the network), while the other should be
allowed to proceed on, using the full 1 Mbps needed by the application. The tele-
phone network is an example of a network that performs such call blockingif the
required resources (an end-to-end circuit in the case of the telephone network) can-
not be allocated to the call, the call is blocked (prevented from entering the network)
and a busy signal is returned to the user. In our example, there is no gain in allowing
a flow into the network if it will not receive a sufficient QoS to be considered
usable. Indeed, there is a cost to admitting a flow that does not receive its needed
QoS, as network resources are being used to support a flow that provides no utility
to the end user.
By explicitly admitting or blocking flows based on their resource requirements,
and the source requirements of already-admitted flows, the network can guarantee
that admitted flows will be able to receive their requested QoS. Implicit in the need
to provide a guaranteed QoS to a flow is the need for the flow to declare its QoS
requirements. This process of having a flow declare its QoS requirement, and then
having the network either accept the flow (at the required QoS) or block the flow is
referred to as the call admission process. This then is our fourth insight (in addition
to the three earlier insights from Section 7.5.2) into the mechanisms needed to pro-
vide QoS.
Insight 4: If sufficient resources will not always be available, and QoS is to be
guaranteed, a call admission process is needed in which flows declare their
QoS requirements and are then either admitted to the network (at the required
QoS) or blocked from the network (if the required QoS cannot be provided by
the network).
Our motivating example in Figure 7.27 highlights the need for several new network
mechanisms and protocols if a call (an end-to-end flow) is to be guaranteed a given
quality of service once it begins:
Resource reservation. The only way to guarantee that a call will have the
resources (link bandwidth, buffers) needed to meet its desired QoS is to explicitly
7.5 NETWORK SUPPORT FOR MULTIMEDIA 653
allocate those resources to the calla process known in networking parlance as
resource reservation. Once resources are reserved, the call has on-demand access
to these resources throughout its duration, regardless of the demands of all other
calls. If a call reserves and receives a guarantee of x Mbps of link bandwidth, and
never transmits at a rate greater than x, the call will see loss- and delay-free per-
formance.
Call admission. If resources are to be reserved, then the network must have a
mechanism for calls to request and reserve resources. Since resources are not
infinite, a call making a call admission request will be denied admission, that is,
be blocked, if the requested resources are not available. Such a call admission is
performed by the telephone networkwe request resources when we dial a num-
ber. If the circuits (TDMA slots) needed to complete the call are available, the
circuits are allocated and the call is completed. If the circuits are not available,
then the call is blocked, and we receive a busy signal. A blocked call can try
again to gain admission to the network, but it is not allowed to send traffic into
the network until it has successfully completed the call admission process. Of
course, a router that allocates link bandwidth should not allocate more than is
available at that link. Typically, a call may reserve only a fraction of the links
bandwidth, and so a router may allocate link bandwidth to more than one call.
However, the sum of the allocated bandwidth to all calls should be less than the
link capacity if hard quality of service guarantees are to be provided.
Call setup signaling. The call admission process described above requires
that a call be able to reserve sufficient resources at each and every network
router on its source-to-destination path to ensure that its end-to-end QoS
requirement is met. Each router must determine the local resources required by
the session, consider the amounts of its resources that are already committed to
other ongoing sessions, and determine whether it has sufficient resources to
satisfy the per-hop QoS requirement of the session at this router without vio-
lating local QoS guarantees made to an already-admitted session. A signaling
protocol is needed to coordinate these various activitiesthe per-hop alloca-
tion of local resources, as well as the overall end-to-end decision of whether or
not the call has been able to reserve sufficient resources at each and every
router on the end-to-end path. This is the job of the call setup protocol, as
shown in Figure 7.28. The RSVP protocol [Zhang 1993, RFC 2210] was
proposed for this purpose within an Internet architecture for providing quality-
of-service guarantees. In ATM networks, the Q2931b protocol [Black 1995]
carries this information among the ATM networks switches and end point.
Despite a tremendous amount of research and development, and even prod-
ucts that provide for per-connection quality of service guarantees, there has been
almost no extended deployment of such services. There are many possible rea-
sons. First and foremost, it may well be the case that the simple application-level
mechanisms that we studied in Sections 7.2 through 7.4, combined with proper
654 CHAPTER 7 MULTIMEDIA NETWORKING
network dimensioning (Section 7.5.1) provide good enough best-effort network
service for multimedia applications. In addition, the added complexity and cost of
deploying and managing a network that provides per-connection quality of serv-
ice guarantees may be judged by ISPs to be simply too high given predicted cus-
tomer revenues for that service.
7.6 Summary
Multimedia networking is one of the most exciting developments in the Internet
today. People throughout the world are spending less time in front of their radios
and televisions, and are instead turning to the Internet to receive audio and video
transmissions, both live and prerecorded. This trend will certainly continue as high-
speed wireless Internet access becomes more and more prevalent. Moreover, with
sites like YouTube, users have become producers as well as consumers of multime-
dia Internet content. In addition to video distribution, the Internet is also being used
to transport phone calls. In fact, over the next 10 years, the Internet, along with wire-
less Internet access, may make the traditional circuit-switched telephone system a
thing of the past. VoIP not only provides phone service inexpensively, but also pro-
vides numerous value-added services, such as video conferencing, online directory
services, voice messaging, and integration into social networks such as Facebook
and Google+.
7.6 SUMMARY 655
QoS call signaling setup
Request/reply
Figure 7.28 The call setup process
In Section 7.1, we described the intrinsic characteristics of video and voice, and
then classified multimedia applications into three categories: (i) streaming stored
audio/video, (ii) conversational voice/video-over-IP, and (iii) streaming live audio/
video.
In Section 7.2, we studied streaming stored video in some depth. For streaming
video applications, prerecorded videos are placed on servers, and users send
requests to these servers to view the videos on demand. We saw that streaming video
systems can be classified into three categories: UDP streaming, HTTP streaming,
and adaptive HTTP streaming. Although all three types of systems are used in prac-
tice, the majority of todays systems employ HTTP streaming and adaptive HTTP
streaming. We observed that the most important performance measure for streaming
video is average throughput. In Section 7.2 we also investigated CDNs, which help
distribute massive amounts of video data to users around the world. We also sur-
veyed the technology behind three major Internet video-streaming companies: Net-
flix, YouTube, and Kankan.
In Section 7.3, we examined how conversational multimedia applications, such as
VoIP, can be designed to run over a best-effort network. For conversational multimedia,
timing considerations are important because conversational applications are highly
delay-sensitive. On the other hand, conversational multimedia applications are loss-
tolerantoccasional loss only causes occasional glitches in audio/video playback, and
these losses can often be partially or fully concealed. We saw how a combination of
client buffers, packet sequence numbers, and timestamps can greatly alleviate the
effects of network-induced jitter. We also surveyed the technology behind Skype, one
of the leading voice- and video-over-IP companies. In Section 7.4, we examined two of
the most important standardized protocols for VoIP, namely, RTP and SIP.
In Section 7.5, we introduced how several network mechanisms (link-level
scheduling disciplines and traffic policing) can be used to provide differentiated
service among several classes of traffic.
Homework Problems and Questions
Chapter 7 Review Questions
SECTION 7.1
R1. Reconstruct Table 7.1 for when Victor Video is watching a 4 Mbps video,
Facebook Frank is looking at a new 100 Kbyte image every 20 seconds, and
Martha Music is listening to 200 kbps audio stream.
R2. There are two types of redundancy in video. Describe them, and discuss how
they can be exploited for efficient compression.
R3. Suppose an analog audio signal is sampled 16,000 times per second, and each
sample is quantized into one of 1024 levels. What would be the resulting bit
rate of the PCM digital audio signal?
656 CHAPTER 7 MULTIMEDIA NETWORKING
R4. Multimedia applications can be classified into three categories. Name and
describe each category.
SECTION 7.2
R5. Streaming video systems can be classified into three categories. Name and
briefly describe each of these categories.
R6. List three disadvantages of UDP streaming.
R7. With HTTP streaming, are the TCP receive buffer and the clients application
buffer the same thing? If not, how do they interact?
R8. Consider the simple model for HTTP streaming. Suppose the server sends
bits at a constant rate of 2 Mbps and playback begins when 8 million bits
have been received. What is the initial buffering delay ?
R9. CDNs typically adopt one of two different server placement philosophies.
Name and briefly describe these two philosophies.
R10. Several cluster selection strategies were described in Section 7.2.4. Which of
these strategies finds a good cluster with respect to the clients LDNS? Which
of these strategies finds a good cluster with respect to the client itself?
R11. Besides network-related considerations such as delay, loss, and bandwidth
performance, there are many additional important factors that go into design-
ing a cluster selection strategy. What are they?
SECTION 7.3
R12. What is the difference between end-to-end delay and packet jitter? What are
the causes of packet jitter?
R13. Why is a packet that is received after its scheduled playout time considered
lost?
R14. Section 7.3 describes two FEC schemes. Briefly summarize them. Both
schemes increase the transmission rate of the stream by adding overhead.
Does interleaving also increase the transmission rate?
SECTION 7.4
R15. How are different RTP streams in different sessions identified by a receiver?
How are different streams from within the same session identified?
R16. What is the role of a SIP registrar? How is the role of an SIP registrar different
from that of a home agent in Mobile IP?
SECTION 7.5
R17. In Section 7.5, we discussed non-preemptive priority queuing. What would
be preemptive priority queuing? Does preemptive priority queuing make
sense for computer networks?
R18. Give an example of a scheduling discipline that is not work conserving.
tp
HOMEWORK PROBLEMS AND QUESTIONS 657
R19. Give an example from queues you experience in your everyday life of FIFO,
priority, RR, and WFQ.
Problems
P1. Consider the figure below. Similar to our discussion of Figure 7.1, suppose
that video is encoded at a fixed bit rate, and thus each video block contains
video frames that are to be played out over the same fixed amount of
time, . The server transmits the first video block at , the second block
at , the third block at , and so on. Once the client begins
playout, each block should be played out time units after the previous
block.
a. Suppose that the client begins playout as soon as the first block arrives at
. In the figure below, how many blocks of video (including the first
block) will have arrived at the client in time for their playout? Explain
how you arrived at your answer.
b. Suppose that the client begins playout now at . How many blocks
of video (including the first block) will have arrived at the client in time
for their playout? Explain how you arrived at your answer.
c. In the same scenario at (b) above, what is the largest number of blocks
that is ever stored in the client buffer, awaiting playout? Explain how you
arrived at your answer.
d. What is the smallest playout delay at the client, such that every video
block has arrived in time for its playout? Explain how you arrived at your
answer.
t1 +
t1
t0 + 2t0 +
t0
658 CHAPTER 7 MULTIMEDIA NETWORKING
Constant bit
rate video
transmission
by server
1
2
3
4
5
6
7
8
9
Time
V
id
e
o
b
lo
ck
n
u
m
b
e
r
t0 t1
Video
reception
at client
P2. Recall the simple model for HTTP streaming shown in Figure 7.3. Recall that
B denotes the size of the clients application buffer, and Q denotes the num-
ber of bits that must be buffered before the client application begins playout.
Also r denotes the video consumption rate. Assume that the server sends bits
at a constant rate x whenever the client buffer is not full.
a. Suppose that x < r. As discussed in the text, in this case playout will alter-nate between periods of continuous playout and periods of freezing.Determine the length of each continuous playout and freezing period as afunction of Q, r, and x.b. Now suppose that x > r. At what time does the client application
buffer become full?
P3. Recall the simple model for HTTP streaming shown in Figure 7.3. Suppose
the buffer size is infinite but the server sends bits at variable rate x(t). Specifi-
cally, suppose x(t) has the following saw-tooth shape. The rate is initially
zero at time t = 0 and linearly climbs to H at time t = T. It then repeats this
pattern again and again, as shown in the figure below.
a. What is the servers average send rate?
b. Suppose that Q = 0, so that the client starts playback as soon as it receives
a video frame. What will happen?
c. Now suppose Q > 0. Determine as a function of Q, H, and T the time at
which playback first begins.
d. Suppose H > 2r and Q = HT/2. Prove there will be no freezing after the
initial playout delay.
e. Suppose H > 2r. Find the smallest value of Q such that there will be no
freezing after the initial playback delay.
f. Now suppose that the buffer size B is finite. Suppose H > 2r. As a func-
tion of Q, B, T, and H, determine the time when the client applica-
tion buffer first becomes full.
t = tf
t = tf
H
Time
T 2T 3T 4T
B
it
r
a
te
x
(t
)
PROBLEMS 659
P4. Recall the simple model for HTTP streaming shown in Figure 7.3. Suppose
the client application buffer is infinite, the server sends at the constant rate x,
and the video consumption rate is r with r < x. Also suppose playback beginsimmediately. Suppose that the user terminates the video early at time t = E.At the time of termination, the server stops sending bits (if it hasnt alreadysent all the bits in the video).a. Suppose the video is infinitely long. How many bits are wasted (that is,sent but not viewed)?b. Suppose the video is T seconds long with T > E. How many bits are
wasted (that is, sent but not viewed)?
P5. Consider a DASH system for which there are N video versions (at N different
rates and qualities) and N audio versions (at N different rates and versions).
Suppose we want to allow the player to choose at any time any of the N video
versions and any of the N audio versions.
a. If we create files so that the audio is mixed in with the video, so server
sends only one media stream at given time, how many files will the server
need to store (each a different URL)?
b. If the server instead sends the audio and video streams separately and has
the client synchronize the streams, how many files will the server need to
store?
P6. In the VoIP example in Section 7.3, let h be the total number of header bytes
added to each chunk, including UDP and IP header.
a. Assuming an IP datagram is emitted every 20 msecs, find the transmission
rate in bits per second for the datagrams generated by one side of this appli-
cation.
b. What is a typical value of h when RTP is used?
P7. Consider the procedure described in Section 7.3 for estimating average delay
di. Suppose that u = 0.1. Let r1 t1 be the most recent sample delay, let r2 t2
be the next most recent sample delay, and so on.
a. For a given audio application suppose four packets have arrived at the
receiver with sample delays r4 t4, r3 t3, r2 t2, and r1 t1. Express the
estimate of delay d in terms of the four samples.
b. Generalize your formula for n sample delays.
c. For the formula in Part b, let n approach infinity and give the resulting
formula. Comment on why this averaging procedure is called an exponen-
tial moving average.
P8. Repeat Parts a and b in Question P7 for the estimate of average delay deviation.
P9. For the VoIP example in Section 7.3, we introduced an online procedure
(exponential moving average) for estimating delay. In this problem we will
660 CHAPTER 7 MULTIMEDIA NETWORKING
examine an alternative procedure. Let ti be the timestamp of the ith packet
received; let ri be the time at which the ith packet is received. Let dn be our
estimate of average delay after receiving the nth packet. After the first packet
is received, we set the delay estimate equal to d1 = r1 t1.
a. Suppose that we would like dn = (r1 t1 + r2 t2 + . . . + rn tn)/n for all
n. Give a recursive formula for dn in terms of dn1, rn, and tn.
b. Describe why for Internet telephony, the delay estimate described in Sec-
tion 7.3 is more appropriate than the delay estimate outlined in Part a.
P10. Compare the procedure described in Section 7.3 for estimating average delay
with the procedure in Section 3.5 for estimating round-trip time. What do the
procedures have in common? How are they different?
P11. Consider the figure below (which is similar to Figure 7.7). A sender begins
sending packetized audio periodically at t = 1. The first packet arrives at the
receiver at t = 8.
Packets
generated
Time
P
a
ck
e
ts
1 8
Packets
received
a. What are the delays (from sender to receiver, ignoring any playout delays)
of packets 2 through 8? Note that each vertical and horizontal line seg-
ment in the figure has a length of 1, 2, or 3 time units.
b. If audio playout begins as soon as the first packet arrives at the receiver at
t = 8, which of the first eight packets sent will not arrive in time for playout?
c. If audio playout begins at t = 9, which of the first eight packets sent will
not arrive in time for playout?
d. What is the minimum playout delay at the receiver that results in all of the
first eight packets arriving in time for their playout?
PROBLEMS 661
P12. Consider again the figure in P11, showing packet audio transmission and
reception times.
a. Compute the estimated delay for packets 2 through 8, using the formula
for di from Section 7.3.2. Use a value of u = 0.1.
b. Compute the estimated deviation of the delay from the estimated average
for packets 2 through 8, using the formula for vi from Section 7.3.2. Use a
value of u = 0.1.
P13. Recall the two FEC schemes for VoIP described in Section 7.3. Suppose the
first scheme generates a redundant chunk for every four original chunks.
Suppose the second scheme uses a low-bit rate encoding whose transmission
rate is 25 percent of the transmission rate of the nominal stream.
a. How much additional bandwidth does each scheme require? How much
playback delay does each scheme add?
b. How do the two schemes perform if the first packet is lost in every group
of five packets? Which scheme will have better audio quality?
c. How do the two schemes perform if the first packet is lost in every group
of two packets? Which scheme will have better audio quality?
P14. a. Consider an audio conference call in Skype with N > 2 participants.
Suppose each participant generates a constant stream of rate r bps. How
many bits per second will the call initiator need to send? How many bits
per second will each of the other N 1 participants need to send?What
is the total send rate, aggregated over all participants?
b. Repeat part (a) for a Skype video conference call using a central server.
c. Repeat part (b), but now for when each peer sends a copy of its video
stream to each of the N 1 other peers.
P15. a. Suppose we send into the Internet two IP datagrams, each carrying a differ-
ent UDP segment. The first datagram has source IP address A1, destination
IP address B, source port P1, and destination port T. The second datagram
has source IP address A2, destination IP address B, source port P2, and des-
tination port T. Suppose that A1 is different from A2 and that P1 is different
from P2. Assuming that both datagrams reach their final destination, will
the two UDP datagrams be received by the same socket? Why or why not?
b. Suppose Alice, Bob, and Claire want to have an audio conference call
using SIP and RTP. For Alice to send and receive RTP packets to and from
Bob and Claire, is only one UDP socket sufficient (in addition to the
socket needed for the SIP messages)? If yes, then how does Alices SIP
client distinguish between the RTP packets received from Bob and Claire?
P16. True or false:
a. If stored video is streamed directly from a Web server to a media player, then
the application is using TCP as the underlying transport protocol.
662 CHAPTER 7 MULTIMEDIA NETWORKING
b. When using RTP, it is possible for a sender to change encoding in the mid-
dle of a session.
c. All applications that use RTP must use port 87.
d. If an RTP session has a separate audio and video stream for each sender,
then the audio and video streams use the same SSRC.
e. In differentiated services, while per-hop behavior defines differences in
performance among classes, it does not mandate any particular mecha-
nism for achieving these performances.
f. Suppose Alice wants to establish an SIP session with Bob. In her INVITE
message she includes the line: m=audio 48753 RTP/AVP 3 (AVP 3
denotes GSM audio). Alice has therefore indicated in this message that
she wishes to send GSM audio.
g. Referring to the preceding statement, Alice has indicated in her INVITE
message that she will send audio to port 48753.
h. SIP messages are typically sent between SIP entities using a default SIP
port number.
i. In order to maintain registration, SIP clients must periodically send
REGISTER messages.
j. SIP mandates that all SIP clients support G.711 audio encoding.
P17. Suppose that the WFQ scheduling policy is applied to a buffer that supports three
classes, and suppose the weights are 0.5, 0.25, and 0.25 for the three classes.
a. Suppose that each class has a large number of packets in the buffer. In what
sequence might the three classes be served in order to achieve the WFQ
weights? (For round robin scheduling, a natural sequence is 123123123 . . .).
b. Suppose that classes 1 and 2 have a large number of packets in the buffer,
and there are no class 3 packets in the buffer. In what sequence might the
three classes be served in to achieve the WFQ weights?
P18. Consider the figure below. Answer the following questions:
Time
Arrivals
Departures
Packet
in service
Time
1 6
5 932
8 10
11
74 12
t = 0
1
t = 2 t = 4 t = 6 t = 8 t = 10 t = 12 t = 14
1
PROBLEMS 663
a. Assuming FIFO service, indicate the time at which packets 2 through 12
each leave the queue. For each packet, what is the delay between its
arrival and the beginning of the slot in which it is transmitted? What is the
average of this delay over all 12 packets?
b. Now assume a priority service, and assume that odd-numbered packets
are high priority, and even-numbered packets are low priority. Indicate
the time at which packets 2 through 12 each leave the queue. For each
packet, what is the delay between its arrival and the beginning of the
slot in which it is transmitted? What is the average of this delay over all
12 packets?
c. Now assume round robin service. Assume that packets 1, 2, 3, 6, 11, and
12 are from class 1, and packets 4, 5, 7, 8, 9, and 10 are from class 2. Indi-
cate the time at which packets 2 through 12 each leave the queue. For
each packet, what is the delay between its arrival and its departure? What
is the average delay over all 12 packets?
d. Now assume weighted fair queueing (WFQ) service. Assume that odd-
numbered packets are from class 1, and even-numbered packets are from
class 2. Class 1 has a WFQ weight of 2, while class 2 has a WFQ weight
of 1. Note that it may not be possible to achieve an idealized WFQ sched-
ule as described in the text, so indicate why you have chosen the particu-
lar packet to go into service at each time slot. For each packet what is the
delay between its arrival and its departure? What is the average delay over
all 12 packets?
e. What do you notice about the average delay in all four cases (FIFO, RR,
priority, and WFQ)?
P19. Consider again the figure for P18.
a. Assume a priority service, with packets 1, 4, 5, 6, and 11 being high-
priority packets. The remaining packets are low priority. Indicate the slots
in which packets 2 through 12 each leave the queue.
b. Now suppose that round robin service is used, with packets 1, 4, 5, 6, and
11 belonging to one class of traffic, and the remaining packets belonging
to the second class of traffic. Indicate the slots in which packets 2 through
12 each leave the queue.
c. Now suppose that WFQ service is used, with packets 1, 4, 5, 6, and 11
belonging to one class of traffic, and the remaining packets belonging to
the second class of traffic. Class 1 has a WFQ weight of 1, while class 2
has a WFQ weight of 2 (note that these weights are different than in the
previous question). Indicate the slots in which packets 2 through 12 each
leave the queue. See also the caveat in the question above regarding WFQ
service.
664 CHAPTER 7 MULTIMEDIA NETWORKING
P20. Consider the figure below, which shows a leaky bucket policer being fed by a
stream of packets. The token buffer can hold at most two tokens, and is ini-
tially full at t = 0. New tokens arrive at a rate of one token per slot. The out-
put link speed is such that if two packets obtain tokens at the beginning of a
time slot, they can both go to the output link in the same slot. The timing
details of the system are as follows:
PROBLEMS 665
Arrivals
Packet queue
(wait for tokens)
9
10
7 6 4
8 5
1
3
2
t = 8 t = 6 t = 4 t = 2 t = 0 t = 4 t = 2 t = 0
r = 1 token/slot
b = 2 tokens
1. Packets (if any) arrive at the beginning of the slot. Thus in the figure,
packets 1, 2, and 3 arrive in slot 0. If there are already packets in the
queue, then the arriving packets join the end of the queue. Packets
proceed towards the front of the queue in a FIFO manner.
2. After the arrivals have been added to the queue, if there are any queued
packets, one or two of those packets (depending on the number of avail-
able tokens) will each remove a token from the token buffer and go to the
output link during that slot. Thus, packets 1 and 2 each remove a token
from the buffer (since there are initially two tokens) and go to the output
link during slot 0.
3. A new token is added to the token buffer if it is not full, since the token
generation rate is r = 1 token/slot.
4. Time then advances to the next time slot, and these steps repeat.
Answer the following questions:
a. For each time slot, identify the packets that are in the queue and the num-
ber of tokens in the bucket, immediately after the arrivals have been
processed (step 1 above) but before any of the packets have passed
through the queue and removed a token. Thus, for the t = 0 time slot in the
example above, packets 1, 2 and 3 are in the queue, and there are two
tokens in the buffer.
b. For each time slot indicate which packets appear on the output after the
token(s) have been removed from the queue. Thus, for the t = 0 time
slot in the example above, packets 1 and 2 appear on the output link
from the leaky buffer during slot 0.
P21. Repeat P20 but assume that r = 2. Assume again that the bucket is initially
full.
P22. Consider P21 and suppose now that r = 3, and that b = 2 as before. Will your
answer to the question above change?
P23. Consider the leaky-bucket policer that polices the average rate and burst size
of a packet flow. We now want to police the peak rate, p, as well. Show how
the output of this leaky-bucket policer can be fed into a second leaky bucket
policer so that the two leaky buckets in series police the average rate, peak
rate, and burst size. Be sure to give the bucket size and token generation rate
for the second policer.
P24. A packet flow is said to conform to a leaky-bucket specification (r,b) with
burst size b and average rate r if the number of packets that arrive to the leaky
bucket is less than rt + b packets in every interval of time of length t for all t.
Will a packet flow that conforms to a leaky-bucket specification (r,b) ever
have to wait at a leaky bucket policer with parameters r and b? Justify your
answer.
P25. Show that as long as r1 < R w1/( wj), then dmax is indeed the maximum delaythat any packet in flow 1 will ever experience in the WFQ queue.Programming AssignmentIn this lab, you will implement a streaming video server and client. The client willuse the real-time streaming protocol (RTSP) to control the actions of the server. Theserver will use the real-time protocol (RTP) to packetize the video for transport overUDP. You will be given Python code that partially implements RTSP and RTP at theclient and server. Your job will be to complete both the client and server code. Whenyou are finished, you will have created a client-server application that does the fol-lowing: The client sends SETUP, PLAY, PAUSE, and TEARDOWN RTSP commands,and the server responds to the commands. When the server is in the playing state, it periodically grabs a stored JPEG frame,packetizes the frame with RTP, and sends the RTP packet into a UDP socket. The client receives the RTP packets, removes the JPEG frames, decompressesthe frames, and renders the frames on the clients monitor.666 CHAPTER 7 MULTIMEDIA NETWORKINGThe code you will be given implements the RTSP protocol in the server and theRTP depacketization in the client. The code also takes care of displaying the trans-mitted video. You will need to implement RTSP in the client and RTP server. Thisprogramming assignment will significantly enhance the students understanding ofRTP, RTSP, and streaming video. It is highly recommended. The assignment alsosuggests a number of optional exercises, including implementing the RTSPDESCRIBE command at both client and server. You can find full details of the assignment, as well as an overview of the RTSP protocol, at the Web site http://www.awl.com/kurose-ross.PROGRAMMING ASSIGNMENT 667http://www.awl.com/kurose-rossWhat made you decide to specialize in multimedia networking?This happened almost by accident. As a PhD student, I got involved with DARTnet, anexperimental network spanning the United States with T1 lines. DARTnet was used as a proving ground for multicast and Internet real-time tools. That led me to write my firstaudio tool, NeVoT. Through some of the DARTnet participants, I became involved in theIETF, in the then-nascent Audio Video Transport working group. This group later ended upstandardizing RTP.What was your first job in the computer industry? What did it entail?My first job in the computer industry was soldering together an Altair computer kit when Iwas a high school student in Livermore, California. Back in Germany, I started a little con-sulting company that devised an address management program for a travel agencystoringdata on cassette tapes for our TRS-80 and using an IBM Selectric typewriter with a home-brew hardware interface as a printer.My first real job was with AT&T Bell Laboratories, developing a network emulator forconstructing experimental networks in a lab environment.What are the goals of the Internet Real-Time Lab?Our goal is to provide components and building blocks for the Internet as the single futurecommunications infrastructure. This includes developing new protocols, such as GIST(for network-layer signaling) and LoST (for finding resources by location), or enhancingprotocols that we have worked on earlier, such as SIP, through work on rich presence,peer-to-peer systems, next-generation emergency calling, and service creation tools.Recently, we have also looked extensively at wireless systems for VoIP, as 802.11b and802.11n networks and maybe WiMax networks are likely to become important last-miletechnologies for telephony. We are also trying to greatly improve the ability of users todiagnose faults in the complicated tangle of providers and equipment, using a peer-to-peerfault diagnosis system called DYSWIS (Do You See What I See).668Henning SchulzrinneHenning Schulzrinne is a professor, chair of the Department ofComputer Science, and head of the Internet Real-Time Laboratory atColumbia University. He is the co-author of RTP, RTSP, SIP, andGISTkey protocols for audio and video communications over theInternet. Henning received his BS in electrical and industrial engineer-ing at TU Darmstadt in Germany, his MS in electrical and computerengineering at the University of Cincinnati, and his PhD in electricalengineering at the University of Massachusetts, Amherst.AN INTERVIEW WITH…We try to do practically relevant work, by building prototypes and open source sys-tems, by measuring performance of real systems, and by contributing to IETF standards.What is your vision for the future of multimedia networking?We are now in a transition phase; just a few years shy of when IP will be the universal plat-form for multimedia services, from IPTV to VoIP. We expect radio, telephone, and TV to beavailable even during snowstorms and earthquakes, so when the Internet takes over the roleof these dedicated networks, users will expect the same level of reliability.We will have to learn to design network technologies for an ecosystem of competingcarriers, service and content providers, serving lots of technically untrained users anddefending them against a small, but destructive, set of malicious and criminal users.Changing protocols is becoming increasingly hard. They are also becoming more complex,as they need to take into account competing business interests, security, privacy, and thelack of transparency of networks caused by firewalls and network address translators.Since multimedia networking is becoming the foundation for almost all of consumerentertainment, there will be an emphasis on managing very large networks, at low cost.Users will expect ease of use, such as finding the same content on all of their devices.Why does SIP have a promising future?As the current wireless network upgrade to 3G networks proceeds, there is the hope of asingle multimedia signaling mechanism spanning all types of networks, from cablemodems, to corporate telephone networks and public wireless networks. Together withsoftware radios, this will make it possible in the future that a single device can be used ona home network, as a cordless BlueTooth phone, in a corporate network via 802.11 and inthe wide area via 3G networks. Even before we have such a single universal wirelessdevice, the personal mobility mechanisms make it possible to hide the differences betweennetworks. One identifier becomes the universal means of reaching a person, rather thanremembering or passing around half a dozen technology- or location-specific telephonenumbers.SIP also breaks apart the provision of voice (bit) transport from voice services. It nowbecomes technically possible to break apart the local telephone monopoly, where onecompany provides neutral bit transport, while others provide IP dial tone and the classicaltelephone services, such as gateways, call forwarding, and caller ID.Beyond multimedia signaling, SIP offers a new service that has been missing in theInternet: event notification. We have approximated such services with HTTP kludges ande-mail, but this was never very satisfactory. Since events are a common abstraction fordistributed systems, this may simplify the construction of new services.669Do you have any advice for students entering the networking field?Networking bridges disciplines. It draws from electrical engineering, all aspects of com-puter science, operations research, statistics, economics, and other disciplines. Thus,networking researchers have to be familiar with subjects well beyond protocols and rout-ing algorithms.Given that networks are becoming such an important part of everyday life, studentswanting to make a difference in the field should think of the new resource constraints innetworks: human time and effort, rather than just bandwidth or storage.Work in networking research can be immensely satisfying since it is about allowingpeople to communicate and exchange ideas, one of the essentials of being human. TheInternet has become the third major global infrastructure, next to the transportation systemand energy distribution. Almost no part of the economy can work without high-performancenetworks, so there should be plenty of opportunities for the foreseeable future.670CHAPTER 8Security inComputerNetworks671Way back in Section 1.6 we described some of the more prevalent and damagingclasses of Internet attacks, including malware attacks, denial of service, sniffing,source masquerading, and message modification and deletion. Although we havesince learned a tremendous amount about computer networks, we still haventexamined how to secure networks from those attacks. Equipped with our newlyacquired expertise in computer networking and Internet protocols, well now studyin-depth secure communication and, in particular, how computer networks can bedefended from those nasty bad guys.Let us introduce Alice and Bob, two people who want to communicate andwish to do so securely. This being a networking text, we should remark that Aliceand Bob could be two routers that want to exchange routing tables securely, a clientand server that want to establish a secure transport connection, or two e-mail appli-cations that want to exchange secure e-mailall case studies that we will considerlater in this chapter. Alice and Bob are well-known fixtures in the security commu-nity, perhaps because their names are more fun than a generic entity named Athat wants to communicate securely with a generic entity named B. Love affairs,wartime communication, and business transactions are the commonly cited humanneeds for secure communications; preferring the first to the latter two, were happyto use Alice and Bob as our sender and receiver, and imagine them in this first scenario.We said that Alice and Bob want to communicate and wish to do so securely,but what precisely does this mean? As we will see, security (like love) is a many-splendored thing; that is, there are many facets to security. Certainly, Alice andBob would like for the contents of their communication to remain secret from aneavesdropper. They probably would also like to make sure that when they arecommunicating, they are indeed communicating with each other, and that if theircommunication is tampered with by an eavesdropper, that this tampering isdetected. In the first part of this chapter, well cover the fundamental cryptographytechniques that allow for encrypting communication, authenticating the party withwhom one is communicating, and ensuring message integrity.In the second part of this chapter, well examine how the fundamental crypto-graphy principles can be used to create secure networking protocols. Once againtaking a top-down approach, well examine secure protocols in each of the (topfour) layers, beginning with the application layer. Well examine how to secure e-mail, how to secure a TCP connection, how to provide blanket security at the net-work layer, and how to secure a wireless LAN. In the third part of this chapter wellconsider operational security, which is about protecting organizational networksfrom attacks. In particular, well take a careful look at how firewalls and intrusiondetection systems can enhance the security of an organizational network.8.1 What Is Network Security?Lets begin our study of network security by returning to our lovers, Alice and Bob,who want to communicate securely. What precisely does this mean? Certainly,Alice wants only Bob to be able to understand a message that she has sent, eventhough they are communicating over an insecure medium where an intruder(Trudy, the intruder) may intercept whatever is transmitted from Alice to Bob. Bobalso wants to be sure that the message he receives from Alice was indeed sent byAlice, and Alice wants to make sure that the person with whom she is communicat-ing is indeed Bob. Alice and Bob also want to make sure that the contents of theirmessages have not been altered in transit. They also want to be assured that theycan communicate in the first place (i.e., that no one denies them access to theresources needed to communicate). Given these considerations, we can identify thefollowing desirable properties of secure communication. Confidentiality. Only the sender and intended receiver should be able to under-stand the contents of the transmitted message. Because eavesdroppers may inter-cept the message, this necessarily requires that the message be somehowencrypted so that an intercepted message cannot be understood by an intercep-tor. This aspect of confidentiality is probably the most commonly perceived672 CHAPTER 8 SECURITY IN COMPUTER NETWORKSmeaning of the term secure communication. Well study cryptographic tech-niques for encrypting and decrypting data in Section 8.2. Message integrity. Alice and Bob want to ensure that the content of their com-munication is not altered, either maliciously or by accident, in transit. Extensionsto the checksumming techniques that we encountered in reliable transport anddata link protocols can be used to provide such message integrity. We will studymessage integrity in Section 8.3. End-point authentication. Both the sender and receiver should be able to confirm the identity of the other party involved in the communicationto confirm that the other party is indeed who or what they claim to be. Face-to-face human communication solves this problem easily by visualrecognition. When communicating entities exchange messages over amedium where they cannot see the other party, authentication is not sosimple. When a user wants to access an inbox, how does the mail server ver-ify that the user is the person he or she claims to be? We study end-pointauthentication in Section 8.4. Operational security. Almost all organizations (companies, universities, andso on) today have networks that are attached to the public Internet. These net-works therefore can potentially be compromised. Attackers can attempt to deposit worms into the hosts in the network, obtain corporate secrets, mapthe internal network configurations, and launch DoS attacks. Well see in Section 8.9 that operational devices such as firewalls and intrusion detectionsystems are used to counter attacks against an organizations network. Afirewall sits between the organizations network and the public network, controlling packet access to and from the network. An intrusion detection sys-tem performs deep packet inspection, alerting the network administratorsabout suspicious activity.Having established what we mean by network security, lets next considerexactly what information an intruder may have access to, and what actions can betaken by the intruder. Figure 8.1 illustrates the scenario. Alice, the sender, wants tosend data to Bob, the receiver. In order to exchange data securely, while meeting therequirements of confidentiality, end-point authentication, and message integrity,Alice and Bob will exchange control messages and data messages (in much thesame way that TCP senders and receivers exchange control segments and data seg-ments). All or some of these messages will typically be encrypted. As discussed inSection 1.6, an intruder can potentially perform eavesdroppingsniffing and recording control and data messages on thechannel. modification, insertion, or deletion of messages or message content.8.1 WHAT IS NETWORK SECURITY? 673As well see, unless appropriate countermeasures are taken, these capabilitiesallow an intruder to mount a wide variety of security attacks: snooping on commu-nication (possibly stealing passwords and data), impersonating another entitity,hijacking an ongoing session, denying service to legitimate network users by over-loading system resources, and so on. A summary of reported attacks is maintained atthe CERT Coordination Center [CERT 2012].Having established that there are indeed real threats loose in the Internet,what are the Internet equivalents of Alice and Bob, our friends who need to com-municate securely? Certainly, Bob and Alice might be human users at two endsystems, for example, a real Alice and a real Bob who really do want to exchangesecure e-mail. They might also be participants in an electronic commerce transac-tion. For example, a real Bob might want to transfer his credit card numbersecurely to a Web server to purchase an item online. Similarly, a real Alice mightwant to interact with her bank online. The parties needing secure communicationmight themselves also be part of the network infrastructure. Recall that thedomain name system (DNS, see Section 2.5) or routing daemons that exchangerouting information (see Section 4.6) require secure communication between twoparties. The same is true for network management applications, a topic we exam-ine in Chapter 9. An intruder that could actively interfere with DNS lookups (asdiscussed in Section 2.5), routing computations [RFC 4272], or network manage-ment functions [RFC 3414] could wreak havoc in the Internet.Having now established the framework, a few of the most important defi-nitions, and the need for network security, let us next delve into cryptography.While the use of cryptography in providing confidentiality is self-evident, well seeshortly that it is also central to providing end-point authentication and messageintegritymaking cryptography a cornerstone of network security.674 CHAPTER 8 SECURITY IN COMPUTER NETWORKSSecuresenderAliceTrudyChannelControl, data messagesSecurereceiverBobData DataFigure 8.1 Sender, receiver, and intruder (Alice, Bob, and Trudy)8.2 Principles of CryptographyAlthough cryptography has a long history dating back at least as far as Julius Caesar,modern cryptographic techniques, including many of those used in the Internet,are based on advances made in the past 30 years. Kahns book, The Codebreakers[Kahn 1967], and Singhs book, The Code Book: The Science of Secrecy fromAncient Egypt to Quantum Cryptography [Singh 1999], provide a fascinating lookat the long history of cryptography. A complete discussion of cryptography itselfrequires a complete book [Kaufman 1995; Schneier 1995] and so we only touchon the essential aspects of cryptography, particularly as they are practiced on theInternet. We also note that while our focus in this section will be on the use ofcryptography for confidentiality, well see shortly that cryptographic techniquesare inextricably woven into authentication, message integrity, nonrepudiation,and more.Cryptographic techniques allow a sender to disguise data so that an intruder cangain no information from the intercepted data. The receiver, of course, must be ableto recover the original data from the disguised data. Figure 8.2 illustrates some ofthe important terminology.Suppose now that Alice wants to send a message to Bob. Alices message inits original form (for example, Bob, I love you. Alice) is known asplaintext, or cleartext. Alice encrypts her plaintext message using an encryptionalgorithm so that the encrypted message, known as ciphertext, looks unintelligibleto any intruder. Interestingly, in many modern cryptographic systems, includingthose used in the Internet, the encryption technique itself is knownpublished, stan-dardized, and available to everyone (for example, [RFC 1321; RFC 3447; RFC8.2 PRINCIPLES OF CRYPTOGRAPHY 675EncryptionalgorithmCiphertextChannelTrudyAlice BobDecryptionalgorithmPlaintextKey:KeyPlaintextKA KBFigure 8.2 Cryptographic components2420; NIST 2001]), even a potential intruder! Clearly, if everyone knows themethod for encoding data, then there must be some secret information that preventsan intruder from decrypting the transmitted data. This is where keys come in.In Figure 8.2, Alice provides a key, KA, a string of numbers or characters, asinput to the encryption algorithm. The encryption algorithm takes the key and theplaintext message, m, as input and produces ciphertext as output. The notationKA(m) refers to the ciphertext form (encrypted using the key KA) of the plaintextmessage, m. The actual encryption algorithm that uses key KA will be evident fromthe context. Similarly, Bob will provide a key, KB, to the decryption algorithmthat takes the ciphertext and Bobs key as input and produces the original plain-text as output. That is, if Bob receives an encrypted message KA(m), he decrypts itby computing KB(KA(m)) = m. In symmetric key systems, Alices and Bobs keysare identical and are secret. In public key systems, a pair of keys is used. One ofthe keys is known to both Bob and Alice (indeed, it is known to the whole world).The other key is known only by either Bob or Alice (but not both). In the follow-ing two subsections, we consider symmetric key and public key systems in moredetail.8.2.1 Symmetric Key CryptographyAll cryptographic algorithms involve substituting one thing for another, for exam-ple, taking a piece of plaintext and then computing and substituting the appropriateciphertext to create the encrypted message. Before studying a modern key-basedcryptographic system, let us first get our feet wet by studying a very old, very sim-ple symmetric key algorithm attributed to Julius Caesar, known as the Caesarcipher (a cipher is a method for encrypting data).For English text, the Caesar cipher would work by taking each letter in theplaintext message and substituting the letter that is k letters later (allowing wrap-around; that is, having the letter z followed by the letter a) in the alphabet. Forexample if k = 3, then the letter a in plaintext becomes d in ciphertext; b in plaintextbecomes e in ciphertext, and so on. Here, the value of k serves as the key. As anexample, the plaintext message bob, i love you. alice becomes ere,l oryh brx. dolfh in ciphertext. While the ciphertext does indeed look likegibberish, it wouldnt take long to break the code if you knew that the Caesar cipherwas being used, as there are only 25 possible key values.An improvement on the Caesar cipher is the monoalphabetic cipher, whichalso substitutes one letter of the alphabet with another letter of the alphabet. How-ever, rather than substituting according to a regular pattern (for example, substitu-tion with an offset of k for all letters), any letter can be substituted for any otherletter, as long as each letter has a unique substitute letter, and vice versa. The substi-tution rule in Figure 8.3 shows one possible rule for encoding plaintext.The plaintext message bob, i love you. alice becomes nkn, sgktc wky. mgsbc. Thus, as in the case of the Caesar cipher, this looks like676 CHAPTER 8 SECURITY IN COMPUTER NETWORKSgibberish. A monoalphabetic cipher would also appear to be better than the Caesarcipher in that there are 26! (on the order of 1026) possible pairings of lettersrather than 25 possible pairings. A brute-force approach of trying all 1026 possiblepairings would require far too much work to be a feasible way of breaking theencryption algorithm and decoding the message. However, by statistical analysisof the plaintext language, for example, knowing that the letters e and t are the mostfrequently occurring letters in typical English text (accounting for 13 percent and 9percent of letter occurrences), and knowing that particular two- and three-letteroccurrences of letters appear quite often together (for example, in, it, the,ion, ing, and so forth) make it relatively easy to break this code. If the intruderhas some knowledge about the possible contents of the message, then it is even eas-ier to break the code. For example, if Trudy the intruder is Bobs wife and suspectsBob of having an affair with Alice, then she might suspect that the names boband alice appear in the text. If Trudy knew for certain that those two namesappeared in the ciphertext and had a copy of the example ciphertext messageabove, then she could immediately determine seven of the 26 letter pairings,requiring 109 fewer possibilities to be checked by a brute-force method. Indeed, ifTrudy suspected Bob of having an affair, she might well expect to find some otherchoice words in the message as well.When considering how easy it might be for Trudy to break Bob and Alicesencryption scheme, one can distinguish three different scenarios, depending on whatinformation the intruder has. Ciphertext-only attack. In some cases, the intruder may have access only to theintercepted ciphertext, with no certain information about the contents of theplaintext message. We have seen how statistical analysis can help in a cipher-text-only attack on an encryption scheme. Known-plaintext attack. We saw above that if Trudy somehow knew for sure thatbob and alice appeared in the ciphertext message, then she could have deter-mined the (plaintext, ciphertext) pairings for the letters a, l, i, c, e, b, and o.Trudy might also have been fortunate enough to have recorded all of the cipher-text transmissions and then found Bobs own decrypted version of one of thetransmissions scribbled on a piece of paper. When an intruder knows some of the(plaintext, ciphertext) pairings, we refer to this as a known-plaintext attack onthe encryption scheme.8.2 PRINCIPLES OF CRYPTOGRAPHY 677Plaintext letter: a b c d e f g h i j k l m n o p q r s t u v w x y zCiphertext letter: m n b v c x z a s d f g h j k l p o i u y t r e w qFigure 8.3 A monoalphabetic cipher Chosen-plaintext attack. In a chosen-plaintext attack, the intruder is able tochoose the plaintext message and obtain its corresponding ciphertext form. Forthe simple encryption algorithms weve seen so far, if Trudy could get Alice tosend the message, The quick brown fox jumps over the lazydog, she could completely break the encryption scheme. Well see shortly thatfor more sophisticated encryption techniques, a chosen-plaintext attack does notnecessarily mean that the encryption technique can be broken.Five hundred years ago, techniques improving on monoalphabetic encryption,known as polyalphabetic encryption, were invented. The idea behind polyalpha-betic encryption is to use multiple monoalphabetic ciphers, with a specific monoal-phabetic cipher to encode a letter in a specific position in the plaintext message.Thus, the same letter, appearing in different positions in the plaintext message,might be encoded differently. An example of a polyalphabetic encryption scheme isshown in Figure 8.4. It has two Caesar ciphers (with k = 5 and k = 19), shown asrows. We might choose to use these two Caesar ciphers, C1 and C2, in the repeatingpattern C1, C2, C2, C1, C2. That is, the first letter of plaintext is to be encoded usingC1, the second and third using C2, the fourth using C1, and the fifth using C2. Thepattern then repeats, with the sixth letter being encoded using C1, the seventh withC2, and so on. The plaintext message bob, i love you. is thus encryptedghu, n etox dhz. Note that the first b in the plaintext message is encryptedusing C1, while the second b is encrypted using C2. In this example, the encryptionand decryption key is the knowledge of the two Caesar keys (k = 5, k = 19) andthe pattern C1, C2, C2, C1, C2.Block CiphersLet us now move forward to modern times and examine how symmetric key encryp-tion is done today. There are two broad classes of symmetric encryption techniques:stream ciphers and block ciphers. Well briefly examine stream ciphers in Section8.7 when we investigate security for wireless LANs. In this section, we focus onblock ciphers, which are used in many secure Internet protocols, including PGP(for secure e-mail), SSL (for securing TCP connections), and IPsec (for securing thenetwork-layer transport).678 CHAPTER 8 SECURITY IN COMPUTER NETWORKSPlaintext letter: a b c d e f g h i j k l m n o p q r s t u v w x y zC1(k = 5): C2(k = 19): f g h i j k l m n o p q r s t u v w x y z a b c d et u v w x y z a b c d e f g h i j k l m n o p q r sFigure 8.4 A polyalphabetic cipher using two Caesar ciphers8.2 PRINCIPLES OF CRYPTOGRAPHY 679In a block cipher, the message to be encrypted is processed in blocks of k bits.For example, if k = 64, then the message is broken into 64-bit blocks, and each blockis encrypted independently. To encode a block, the cipher uses a one-to-one map-ping to map the k-bit block of cleartext to a k-bit block of ciphertext. Lets look atan example. Suppose that k = 3, so that the block cipher maps 3-bit inputs (clear-text) to 3-bit outputs (ciphertext). One possible mapping is given in Table 8.1.Notice that this is a one-to-one mapping; that is, there is a different output for eachinput. This block cipher breaks the message up into 3-bit blocks and encrypts eachblock according to the above mapping. You should verify that the message010110001111 gets encrypted into 101000111001.Continuing with this 3-bit block example, note that the mapping in Table 8.1 isjust one mapping of many possible mappings. How many possible mappings are there?To answer this question, observe that a mapping is nothing more than a permuta-tion of all the possible inputs. There are 23 (= 8) possible inputs (listed under theinput columns). These eight inputs can be permuted in 8! = 40,320 different ways.Since each of these permutations specifies a mapping, there are 40,320 possiblemappings. We can view each of these mappings as a keyif Alice and Bob bothknow the mapping (the key), they can encrypt and decrypt the messages sentbetween them.The brute-force attack for this cipher is to try to decrypt ciphtertext by using allmappings. With only 40,320 mappings (when k = 3), this can quickly be accom-plished on a desktop PC. To thwart brute-force attacks, block ciphers typically usemuch larger blocks, consisting of k = 64 bits or even larger. Note that the number ofpossible mappings for a general k-block cipher is 2k!, which is astronomical for evenmoderate values of k (such as k = 64).Although full-table block ciphers, as just described, with moderate values ofk can produce robust symmetric key encryption schemes, they are unfortunatelydifficult to implement. For k = 64 and for a given mapping, Alice and Bobwould need to maintain a table with 264 input values, which is an infeasible task.Moreover, if Alice and Bob were to change keys, they would have to each regenerateTable 8.1 A specific 3-bit block cipherinput output input output000 110 100 011001 111 101 010010 101 110 000011 100 111 001the table. Thus, a full-table block cipher, providing predetermined mappingsbetween all inputs and outputs (as in the example above), is simply out of thequestion.Instead, block ciphers typically use functions that simulate randomly permutedtables. An example (adapted from [Kaufman 1995]) of such a function for k = 64bits is shown in Figure 8.5. The function first breaks a 64-bit block into 8 chunks,with each chunk consisting of 8 bits. Each 8-bit chunk is processed by an 8-bit to 8-bit table, which is of manageable size. For example, the first chunk is processed bythe table denoted by T1. Next, the 8 output chunks are reassembled into a 64-bitblock. The positions of the 64 bits in the block are then scrambled (permuted) toproduce a 64-bit output. This output is fed back to the 64-bit input, where anothercycle begins. After n such cycles, the function provides a 64-bit block of ciphertext.The purpose of the rounds is to make each input bit affect most (if not all) of thefinal output bits. (If only one round were used, a given input bit would affect only 8of the 64 output bits.) The key for this block cipher algorithm would be the eightpermutation tables (assuming the scramble function is publicly known).Today there are a number of popular block ciphers, including DES (standing forData Encryption Standard), 3DES, and AES (standing for Advanced EncryptionStandard). Each of these standards uses functions, rather than predetermined tables,along the lines of Figure 8.5 (albeit more complicated and specific to each cipher).Each of these algorithms also uses a string of bits for a key. For example, DES uses64-bit blocks with a 56-bit key. AES uses 128-bit blocks and can operate with keysthat are 128, 192, and 256 bits long. An algorithms key determines the specific680 CHAPTER 8 SECURITY IN COMPUTER NETWORKS64-bit outputLoopfor nrounds8 bits8 bitsT18 bits8 bitsT28 bits8 bitsT38 bits64-bit input8 bitsT48 bits8 bitsT58 bits8 bitsT68 bits8 bitsT78 bits8 bitsT864-bit scramblerFigure 8.5 An example of a block cipher8.2 PRINCIPLES OF CRYPTOGRAPHY 681mini-table mappings and permutations within the algorithms internals. The brute-force attack for each of these ciphers is to cycle through all the keys, applyingthe decryption algorithm with each key. Observe that with a key length of n, thereare 2n possible keys. NIST [NIST 2001] estimates that a machine that could crack56-bit DES in one second (that is, try all 256 keys in one second) would take approx-imately 149 trillion years to crack a 128-bit AES key.Cipher-Block ChainingIn computer networking applications, we typically need to encrypt long messages(or long streams of data). If we apply a block cipher as described by simply chop-ping up the message into k-bit blocks and independently encrypting each block, asubtle but important problem occurs. To see this, observe that two or more of the clear-text blocks can be identical. For example, the cleartext in two or more blocks couldbe HTTP/1.1. For these identical blocks, a block cipher would, of course, producethe same ciphertext. An attacker could potentially guess the cleartext when it seesidentical ciphertext blocks and may even be able to decrypt the entire message byidentifying identical ciphtertext blocks and using knowledge about the underlyingprotocol structure [Kaufman 1995].To address this problem, we can mix some randomness into the ciphertext sothat identical plaintext blocks produce different ciphertext blocks. To explain thisidea, let m(i) denote the ith plaintext block, c(i) denote the ith ciphertext block,and a b denote the exclusive-or (XOR) of two bit strings, a and b. (Recall thatthe 0 0 = 1 1 = 0 and 0 1 = 1 0 = 1, and the XOR of two bit strings is doneon a bit-by-bit basis. So, for example, 10101010 11110000 = 01011010.) Also,denote the block-cipher encryption algorithm with key S as KS. The basic idea isas follows. The sender creates a random k-bit number r(i) for the ith block and cal-culates c(i) = KS(m(i) r(i)). Note that a new k-bit random number is chosen foreach block. The sender then sends c(1), r(1), c(2), r(2), c(3), r(3), and so on. Sincethe receiver receives c(i) and r(i), it can recover each block of the plaintext bycomputing m(i) = KS(c(i)) r(i). It is important to note that, although r(i) is sentin the clear and thus can be sniffed by Trudy, she cannot obtain the plaintext m(i),since she does not know the key KS. Also note that if two plaintext blocks m(i) andm( j) are the same, the corresponding ciphertext blocks c(i) and c( j) will be differ-ent (as long as the random numbers r(i) and r( j) are different, which occurs withvery high probability).As an example, consider the 3-bit block cipher in Table 8.1. Suppose the plain-text is 010010010. If Alice encrypts this directly, without including the randomness,the resulting ciphertext becomes 101101101. If Trudy sniffs this ciphertext, becauseeach of the three cipher blocks is the same, she can correctly surmise that each ofthe three plaintext blocks are the same. Now suppose instead Alice generates therandom blocks r(1) = 001, r(2) =111, and r(3) = 100 and uses the above techniqueto generate the ciphertext c(1) = 100, c(2) = 010, and c(3) = 000. Note that the threeciphertext blocks are different even though the plaintext blocks are the same. Alicethen sends c(1), r(1), c(2), and r(2). You should verify that Bob can obtain the origi-nal plaintext using the shared key KS.The astute reader will note that introducing randomness solves one problem butcreates another: namely, Alice must transmit twice as many bits as before. Indeed,for each cipher bit, she must now also send a random bit, doubling the requiredbandwidth. In order to have our cake and eat it too, block ciphers typically use atechnique called Cipher Block Chaining (CBC). The basic idea is to send only onerandom value along with the very first message, and then have the sender andreceiver use the computed coded blocks in place of the subsequent random number.Specifically, CBC operates as follows:1. Before encrypting the message (or the stream of data), the sender generates arandom k-bit string, called the Initialization Vector (IV). Denote this initial-ization vector by c(0). The sender sends the IV to the receiver in cleartext.2. For the first block, the sender calculates m(1) c(0), that is, calculates theexclusive-or of the first block of cleartext with the IV. It then runs the resultthrough the block-cipher algorithm to get the corresponding ciphertext block;that is, c(1) = KS(m(1) c(0)). The sender sends the encrypted block c(1) tothe receiver.3. For the ith block, the sender generates the ith ciphertext block from c(i) =KS(m(i) c(i 1)).Lets now examine some of the consequences of this approach. First, thereceiver will still be able to recover the original message. Indeed, when the receiverreceives c(i), it decrypts it with KS to obtain s(i) = m(i) c(i 1); since the receiveralso knows c(i 1), it then obtains the cleartext block from m(i) = s(i) c(i 1). Sec-ond, even if two cleartext blocks are identical, the corresponding ciphtertexts(almost always) will be different. Third, although the sender sends the IV in theclear, an intruder will still not be able to decrypt the ciphertext blocks, since theintruder does not know the secret key, S. Finally, the sender only sends one over-head block (the IV), thereby negligibly increasing the bandwidth usage for longmessages (consisting of hundreds of blocks).As an example, lets now determine the ciphertext for the 3-bit block cipher inTable 8.1 with plaintext 010010010 and IV = c(0) = 001. The sender first uses theIV to calculate c(1) = KS(m(1) c(0)) = 100. The sender then calculates c(2) =KS(m(2) c(1)) = KS(010 100) = 000, and c(3) = KS(m(3) c(2)) = KS(010 000) = 101. The reader should verify that the receiver, knowing the IV and KS canrecover the original plaintext.CBC has an important consequence when designing secure network protocols:well need to provide a mechanism within the protocol to distribute the IV fromsender to receiver. Well see how this is done for several protocols later in thischapter.682 CHAPTER 8 SECURITY IN COMPUTER NETWORKS8.2.2 Public Key EncryptionFor more than 2,000 years (since the time of the Caesar cipher and up to the1970s), encrypted communication required that the two communicating partiesshare a common secretthe symmetric key used for encryption and decryption.One difficulty with this approach is that the two parties must somehow agree onthe shared key; but to do so requires (presumably secure) communication! Perhapsthe parties could first meet and agree on the key in person (for example, two ofCaesars centurions might meet at the Roman baths) and thereafter communicatewith encryption. In a networked world, however, communicating parties may nevermeet and may never converse except over the network. Is it possible for two par-ties to communicate with encryption without having a shared secret key that isknown in advance? In 1976, Diffie and Hellman [Diffie 1976] demonstrated analgorithm (known now as Diffie-Hellman Key Exchange) to do just thata radi-cally different and marvelously elegant approach toward secure communicationthat has led to the development of todays public key cryptography systems. Wellsee shortly that public key cryptography systems also have several wonderfulproperties that make them useful not only for encryption, but for authenticationand digital signatures as well. Interestingly, it has recently come to light thatideas similar to those in [Diffie 1976] and [RSA 1978] had been independentlydeveloped in the early 1970s in a series of secret reports by researchers at theCommunications-Electronics Security Group in the United Kingdom [Ellis1987]. As is often the case, great ideas can spring up independently in manyplaces; fortunately, public key advances took place not only in private, but alsoin the public view, as well.The use of public key cryptography is conceptually quite simple. Suppose Alicewants to communicate with Bob. As shown in Figure 8.6, rather than Bob and Alice8.2 PRINCIPLES OF CRYPTOGRAPHY 683EncryptionalgorithmCiphertextDecryptionalgorithmPlaintextmessage, mPlaintextmessage, mPrivate decryption keym = KB(KB+(m))KBKB+(m)Public encryption keyKB+Figure 8.6 Public key cryptographysharing a single secret key (as in the case of symmetric key systems), Bob (the recip-ient of Alices messages) instead has two keysa public key that is available toeveryone in the world (including Trudy the intruder) and a private key that is knownonly to Bob. We will use the notation KB+ and KB to refer to Bobs public and privatekeys, respectively. In order to communicate with Bob, Alice first fetches Bobs pub-lic key. Alice then encrypts her message, m, to Bob using Bobs public key and aknown (for example, standardized) encryption algorithm; that is, Alice computesKB+(m). Bob receives Alices encrypted message and uses his private key and a known(for example, standardized) decryption algorithm to decrypt Alices encrypted mes-sage. That is, Bob computes KB(KB+(m)). We will see below that there are encryp-tion/decryption algorithms and techniques for choosing public and private keys suchthat KB(KB+(m)) = m; that is, applying Bobs public key, KB+, to a message, m (to getKB+(m)), and then applying Bobs private key, KB, to the encrypted version of m (thatis, computing KB(KB+(m))) gives back m. This is a remarkable result! In this manner,Alice can use Bobs publicly available key to send a secret message to Bob withouteither of them having to distribute any secret keys! We will see shortly that we caninterchange the public key and private key encryption and get the same remarkableresultthat is, KB (B+(m)) = KB+ (KB(m)) = m.The use of public key cryptography is thus conceptually simple. But two imme-diate worries may spring to mind. A first concern is that although an intruder inter-cepting Alices encrypted message will see only gibberish, the intruder knows boththe key (Bobs public key, which is available for all the world to see) and the algo-rithm that Alice used for encryption. Trudy can thus mount a chosen-plaintextattack, using the known standardized encryption algorithm and Bobs publicly avail-able encryption key to encode any message she chooses! Trudy might well try, forexample, to encode messages, or parts of messages, that she suspects that Alicemight send. Clearly, if public key cryptography is to work, key selection and encryp-tion/decryption must be done in such a way that it is impossible (or at least so hardas to be nearly impossible) for an intruder to either determine Bobs private key orsomehow otherwise decrypt or guess Alices message to Bob. A second concern isthat since Bobs encryption key is public, anyone can send an encrypted message toBob, including Alice or someone claiming to be Alice. In the case of a single sharedsecret key, the fact that the sender knows the secret key implicitly identifies thesender to the receiver. In the case of public key cryptography, however, this is nolonger the case since anyone can send an encrypted message to Bob using Bobspublicly available key. A digital signature, a topic we will study in Section 8.3, isneeded to bind a sender to a message.RSAWhile there may be many algorithms that address these concerns, the RSA algo-rithm (named after its founders, Ron Rivest, Adi Shamir, and Leonard Adleman)has become almost synonymous with public key cryptography. Lets first see howRSA works and then examine why it works.684 CHAPTER 8 SECURITY IN COMPUTER NETWORKSRSA makes extensive use of arithmetic operations using modulo-n arithmetic.So lets briefly review modular arithmetic. Recall that x mod n simply means theremainder of x when divided by n; so, for example, 19 mod 5 = 4. In modular arith-metic, one performs the usual operations of addition, multiplication, and exponenti-ation. However, the result of each operation is replaced by the integer remainder thatis left when the result is divided by n. Adding and multiplying with modular arith-metic is facilitated with the following handy facts:[(a mod n) + (b mod n)] mod n = (a + b) mod n[(a mod n) (b mod n)] mod n = (a b) mod n[(a mod n) (b mod n)] mod n = (a b) mod nIt follows from the third fact that (a mod n)d mod n = ad mod n, which is an identitythat we will soon find very useful.Now suppose that Alice wants to send to Bob an RSA-encrypted message, asshown in Figure 8.6. In our discussion of RSA, lets always keep in mind that a mes-sage is nothing but a bit pattern, and every bit pattern can be uniquely representedby an integer number (along with the length of the bit pattern). For example, supposea message is the bit pattern 1001; this message can be represented by the decimalinteger 9. Thus, when encrypting a message with RSA, it is equivalent to encryptingthe unique integer number that represents the message.There are two interrelated components of RSA: The choice of the public key and the private key The encryption and decryption algorithmTo generate the public and private RSA keys, Bob performs the following steps:1. Choose two large prime numbers, p and q. How large should p and q be? Thelarger the values, the more difficult it is to break RSA, but the longer it takes toperform the encoding and decoding. RSA Laboratories recommends that theproduct of p and q be on the order of 1,024 bits. For a discussion of how tofind large prime numbers, see [Caldwell 2012].2. Compute n = pq and z = (p 1)(q 1).3. Choose a number, e, less than n, that has no common factors (other than 1)with z. (In this case, e and z are said to be relatively prime.) The letter e is usedsince this value will be used in encryption.4. Find a number, d, such that ed 1 is exactly divisible (that is, with no remain-der) by z. The letter d is used because this value will be used in decryption. Putanother way, given e, we choose d such thated mod z = 15. The public key that Bob makes available to the world, KB+, is the pair of num-bers (n, e); his private key, KB, is the pair of numbers (n, d).8.2 PRINCIPLES OF CRYPTOGRAPHY 685The encryption by Alice and the decryption by Bob are done as follows: Suppose Alice wants to send Bob a bit pattern represented by the integer number m(with m < n). To encode, Alice performs the exponentiation me, and then computesthe integer remainder when me is divided by n. In other words, the encryptedvalue, c, of Alices plaintext message, m, isc = me mod nThe bit pattern corresponding to this ciphertext c is sent to Bob. To decrypt the received ciphertext message, c, Bob computesm = cd mod nwhich requires the use of his private key (n,d).As a simple example of RSA, suppose Bob chooses p = 5 and q = 7. (Admit-tedly, these values are far too small to be secure.) Then n = 35 and z = 24. Bobchooses e = 5, since 5 and 24 have no common factors. Finally, Bob chooses d = 29,since 5 29 1 (that is, ed 1) is exactly divisible by 24. Bob makes the two val-ues, n = 35 and e = 5, public and keeps the value d = 29 secret. Observing these twopublic values, suppose Alice now wants to send the letters l, o, v, and e to Bob. Inter-preting each letter as a number between 1 and 26 (with a being 1, and z being 26),Alice and Bob perform the encryption and decryption shown in Tables 8.2 and 8.3,respectively. Note that in this example, we consider each of the four letters as a dis-tinct message. A more realistic example would be to convert the four letters intotheir 8-bit ASCII representations and then encrypt the integer corresponding to theresulting 32-bit bit pattern. (Such a realistic example generates numbers that aremuch too long to print in a textbook!)Given that the toy example in Tables 8.2 and 8.3 has already produced someextremely large numbers, and given that we saw earlier that p and q should each beseveral hundred bits long, several practical issues regarding RSA come to mind.686 CHAPTER 8 SECURITY IN COMPUTER NETWORKSTable 8.2 Alices RSA encryption, e = 5, n = 35Plaintext Letter m: numeric representation m e Ciphertext c = m e mod nl 12 248832 17o 15 759375 15v 22 5153632 22e 5 3125 10How does one choose large prime numbers? How does one then choose e and d?How does one perform exponentiation with large numbers? A discussion of theseimportant issues is beyond the scope of this book; see [Kaufman 1995] and the ref-erences therein for details.Session KeysWe note here that the exponentiation required by RSA is a rather time-consumingprocess. By contrast, DES is at least 100 times faster in software and between 1,000and 10,000 times faster in hardware [RSA Fast 2012]. As a result, RSA is often usedin practice in combination with symmetric key cryptography. For example, if Alicewants to send Bob a large amount of encrypted data, she could do the following.First Alice chooses a key that will be used to encode the data itself; this key isreferred to as a session key, and is denoted by KS. Alice must inform Bob of the ses-sion key, since this is the shared symmetric key they will use with a symmetric keycipher (e.g., with DES or AES). Alice encrypts the session key using Bobs publickey, that is, computes c = (KS)e mod n. Bob receives the RSA-encrypted session key,c, and decrypts it to obtain the session key, KS. Bob now knows the session key thatAlice will use for her encrypted data transfer.Why Does RSA Work?RSA encryption/decryption appears rather magical. Why should it be that by apply-ing the encryption algorithm and then the decryption algorithm, one recovers theoriginal message? In order to understand why RSA works, again denote n = pq,where p and q are the large prime numbers used in the RSA algorithm.Recall that, under RSA encryption, a message (uniquely represented by an inte-ger), m, is exponentiated to the power e using modulo-n arithmetic, that is,c = me mod nDecryption is performed by raising this value to the power d, again using modulo-narithmetic. The result of an encryption step followed by a decryption step is thus8.2 PRINCIPLES OF CRYPTOGRAPHY 687Table 8.3 Bobs RSA decryption, d = 29, n = 35Ciphertext c c d m = c d mod n Plaintext Letter17 4819685721067509150915091411825223071697 12 l15 127834039403948858939111232757568359375 15 o22 851643319086537701956194499721106030592 22 v10 1000000000000000000000000000000 5 e(me mod n)d mod n . Lets now see what we can say about this quantity. As mentionedearlier, one important property of modulo arithmetic is (a mod n)d mod n = ad modn for any values a, n, and d. Thus, using a = me in this property, we have(me mod n)d mod n = med mod nIt therefore remains to show that med mod n = m. Although were trying to removesome of the magic about why RSA works, to establish this, well need to use a rathermagical result from number theory here. Specifically, well need the result that says ifp and q are prime, n = pq, and z = (p 1)(q 1), then xy mod n is the same as x(y mod z)mod n [Kaufman 1995]. Applying this result with x = m and y = ed we havemed mod n = m(ed mod z) mod nBut remember that we have chosen e and d such that ed mod z = 1. This gives usmed mod n = m1 mod n = mwhich is exactly the result we are looking for! By first exponentiating to the powerof e (that is, encrypting) and then exponentiating to the power of d (that is, decrypt-ing), we obtain the original value, m. Even more wonderful is the fact that if we firstexponentiate to the power of d and then exponentiate to the power of ethat is, wereverse the order of encryption and decryption, performing the decryption operationfirst and then applying the encryption operationwe also obtain the original value,m. This wonderful result follows immediately from the modular arithmetic:(md mod n)e mod n = mde mod n = med mod n = (me mod n)d mod n688 CHAPTER 8 SECURITY IN COMPUTER NETWORKSThe security of RSA relies on the fact that there are no known algorithms forquickly factoring a number, in this case the public value n, into the primes p and q. Ifone knew p and q, then given the public value e, one could easily compute the secretkey, d. On the other hand, it is not known whether or not there exist fast algorithms forfactoring a number, and in this sense, the security of RSA is not guaranteed.Another popular public-key encryption algorithm is the Diffie-Hellman algo-rithm, which we will briefly explore in the homework problems. Diffie-Hellman isnot as versatile as RSA in that it cannot be used to encrypt messages of arbitrarylength; it can be used, however, to establish a symmetric session key, which is inturn used to encrypt messages.8.3 Message Integrity and Digital SignaturesIn the previous section we saw how encryption can be used to provide confidential-ity to two communicating entities. In this section we turn to the equally importantcryptography topic of providing message integrity (also known as message authen-tication). Along with message integrity, we will discuss two related topics in thissection: digital signatures and end-point authentication.We define the message integrity problem using, once again, Alice and Bob.Suppose Bob receives a message (which may be encrypted or may be in plaintext)and he believes this message was sent by Alice. To authenticate this message, Bobneeds to verify:1. The message indeed originated from Alice.2. The message was not tampered with on its way to Bob.Well see in Sections 8.4 through 8.7 that this problem of message integrity is acritical concern in just about all secure networking protocols.As a specific example, consider a computer network using a link-state routingalgorithm (such as OSPF) for determining routes between each pair of routers inthe network (see Chapter 4). In a link-state algorithm, each router needs to broad-cast a link-state message to all other routers in the network. A routers link-statemessage includes a list of its directly connected neighbors and the direct costs tothese neighbors. Once a router receives link-state messages from all of the otherrouters, it can create a complete map of the network, run its least-cost routingalgorithm, and configure its forwarding table. One relatively easy attack on therouting algorithm is for Trudy to distribute bogus link-state messages with incor-rect link-state information. Thus the need for message integritywhen router Breceives a link-state message from router A, router B should verify that router Aactually created the message and, further, that no one tampered with the messagein transit.In this section, we describe a popular message integrity technique that is usedby many secure networking protocols. But before doing so, we need to coveranother important topic in cryptographycryptographic hash functions.8.3.1 Cryptographic Hash FunctionsAs shown in Figure 8.7, a hash function takes an input, m, and computes a fixed-size string H(m) known as a hash. The Internet checksum (Chapter 3) and CRCs(Chapter 4) meet this definition. A cryptographic hash function is required to havethe following additional property: It is computationally infeasible to find any two different messages x and y suchthat H(x) = H(y).Informally, this property means that it is computationally infeasible for anintruder to substitute one message for another message that is protected by the hashfunction. That is, if (m, H(m)) are the message and the hash of the message created8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 689by the sender, then an intruder cannot forge the contents of another message, y, thathas the same hash value as the original message.Lets convince ourselves that a simple checksum, such as the Internet check-sum, would make a poor cryptographic hash function. Rather than performing 1scomplement arithmetic (as in the Internet checksum), let us compute a checksumby treating each character as a byte and adding the bytes together using 4-bytechunks at a time. Suppose Bob owes Alice $100.99 and sends an IOU to Aliceconsisting of the text string IOU100.99BOB. The ASCII representation (inhexadecimal notation) for these letters is 49, 4F, 55, 31, 30, 30, 2E, 39, 39,42, 4F, 42.Figure 8.8 (top) shows that the 4-byte checksum for this message is B2 C1D2 AC. A slightly different message (and a much more costly one for Bob) isshown in the bottom half of Figure 8.8. The messages IOU100.99BOB andIOU900.19BOB have the same checksum. Thus, this simple checksum algo-rithm violates the requirement above. Given the original data, it is simple to findanother set of data with the same checksum. Clearly, for security purposes, we aregoing to need a more powerful hash function than a checksum.The MD5 hash algorithm of Ron Rivest [RFC 1321] is in wide use today. Itcomputes a 128-bit hash in a four-step process consisting of a padding step(adding a one followed by enough zeros so that the length of the message satisfiescertain conditions), an append step (appending a 64-bit representation of the mes-sage length before padding), an initialization of an accumulator, and a final loop-ing step in which the messages 16-word blocks are processed (mangled) in fourrounds. For a description of MD5 (including a C source code implementation) see[RFC 1321].690 CHAPTER 8 SECURITY IN COMPUTER NETWORKSMany-to-onehash functionLong message: mDear Alice:This is a VERY long lettersince there is so much tosay…………………….BobFixed-lengthhash: H(m)OpgmdvboijrtnsdgghPPdogm;lcvkbFigure 8.7 Hash functionsThe second major hash algorithm in use today is the Secure Hash Algorithm(SHA-1) [FIPS 1995]. This algorithm is based on principles similar to those used inthe design of MD4 [RFC 1320], the predecessor to MD5. SHA-1, a US federalstandard, is required for use whenever a cryptographic hash algorithm is needed forfederal applications. It produces a 160-bit message digest. The longer output lengthmakes SHA-1 more secure.8.3.2 Message Authentication CodeLets now return to the problem of message integrity. Now that we understand hashfunctions, lets take a first stab at how we might perform message integrity:1. Alice creates message m and calculates the hash H(m) (for example withSHA-1).2. Alice then appends H(m) to the message m, creating an extended message(m, H(m)), and sends the extended message to Bob.3. Bob receives an extended message (m, h) and calculates H(m). If H(m) = h,Bob concludes that everything is fine.This approach is obviously flawed. Trudy can create a bogus message m in whichshe says she is Alice, calculate H(m), and send Bob (m, H(m)). When Bob receivesthe message, everything checks out in step 3, so Bob doesnt suspect any funnybusiness.Figure 8.8 Initial message and fraudulent message have the same checksum!MessageI O U 10 0 . 99 B O BASCIIRepresentation49 4F 55 3130 30 2E 3939 42 4F 42B2 C1 D2 AC ChecksumMessageI O U 90 0 . 19 B O BASCIIRepresentation49 4F 55 3930 30 2E 3139 42 4F 42B2 C1 D2 AC Checksum8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 691To perform message integrity, in addition to using cryptographic hash func-tions, Alice and Bob will need a shared secret s. This shared secret, which is nothingmore than a string of bits, is called the authentication key. Using this shared secret,message integrity can be performed as follows:1. Alice creates message m, concatenates s with m to create m + s, and calculatesthe hash H(m + s) (for example with SHA-1). H(m + s) is called the messageauthentication code (MAC).2. Alice then appends the MAC to the message m, creating an extended message(m, H(m + s)), and sends the extended message to Bob.3. Bob receives an extended message (m, h) and knowing s, calculates the MACH(m + s). If H(m + s) = h, Bob concludes that everything is fine.A summary of the procedure is shown in Figure 8.9. Readers should note that theMAC here (standing for message authentication code) is not the same MAC usedin link-layer protocols (standing for medium access control)!One nice feature of a MAC is that it does not require an encryption algorithm.Indeed, in many applications, including the link-state routing algorithm describedearlier, communicating entities are only concerned with message integrity and arenot concerned with message confidentiality. Using a MAC, the entities can authenti-cate the messages they send to each other without having to integrate complexencryption algorithms into the integrity process.As you might expect, a number of different standards for MACs have been pro-posed over the years. The most popular standard today is HMAC, which can be692 CHAPTER 8 SECURITY IN COMPUTER NETWORKSH(.)H(.)mmmmsss+ InternetCompareKey:= Message= Shared secretH(m+s)H(m+s)Figure 8.9 Message authentication code (MAC)used either with MD5 or SHA-1. HMAC actually runs data and the authenticationkey through the hash function twice [Kaufman 1995; RFC 2104].There still remains an important issue. How do we distribute the shared authen-tication key to the communicating entities? For example, in the link-state routingalgorithm, we would somehow need to distribute the secret authentication key toeach of the routers in the autonomous system. (Note that the routers can all usethe same authentication key.) A network administrator could actually accomplishthis by physically visiting each of the routers. Or, if the network administrator isa lazy guy, and if each router has its own public key, the network administratorcould distribute the authentication key to any one of the routers by encrypting itwith the routers public key and then sending the encrypted key over the networkto the router.8.3.3 Digital SignaturesThink of the number of the times youve signed your name to a piece of paper dur-ing the last week. You sign checks, credit card receipts, legal documents, and letters.Your signature attests to the fact that you (as opposed to someone else) haveacknowledged and/or agreed with the documents contents. In a digital world, oneoften wants to indicate the owner or creator of a document, or to signify ones agree-ment with a documents content. A digital signature is a cryptographic techniquefor achieving these goals in a digital world.Just as with handwritten signatures, digital signing should be done in a way thatis verifiable and nonforgeable. That is, it must be possible to prove that a documentsigned by an individual was indeed signed by that individual (the signature must beverifiable) and that only that individual could have signed the document (the signa-ture cannot be forged).Lets now consider how we might design a digital signature scheme. Observethat when Bob signs a message, Bob must put something on the message that isunique to him. Bob could consider attaching a MAC for the signature, where theMAC is created by appending his key (unique to him) to the message, and then takingthe hash. But for Alice to verify the signature, she must also have a copy of the key,in which case the key would not be unique to Bob. Thus, MACs are not going to getthe job done here.Recall that with public-key cryptography, Bob has both a public and privatekey, with both of these keys being unique to Bob. Thus, public-key cryptography isan excellent candidate for providing digital signatures. Let us now examine how itis done.Suppose that Bob wants to digitally sign a document, m. We can think of thedocument as a file or a message that Bob is going to sign and send. As shown inFigure 8.10, to sign this document, Bob simply uses his private key, KB, to com-pute KB(m). At first, it might seem odd that Bob is using his private key (which, aswe saw in Section 8.2, was used to decrypt a message that had been encrypted8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 693with his public key) to sign a document. But recall that encryption and decryptionare nothing more than mathematical operations (exponentiation to the power of eor d in RSA; see Section 8.2) and recall that Bobs goal is not to scramble orobscure the contents of the document, but rather to sign the document in a man-ner that is verifiable and nonforgeable. Bobs digital signature of the document isKB(m).Does the digital signature KB(m) meet our requirements of being verifiable andnonforgeable? Suppose Alice has m and KB(m). She wants to prove in court (beinglitigious) that Bob had indeed signed the document and was the only person whocould have possibly signed the document. Alice takes Bobs public key, KB+, andapplies it to the digital signature, KB(m), associated with the document, m. That is,she computes KB+(KB(m)), and voil, with a dramatic flurry, she produces m, whichexactly matches the original document! Alice then argues that only Bob could havesigned the document, for the following reasons: Whoever signed the message must have used the private key, KB, in computingthe signature KB(m), such that KB+(KB(m)) = m. The only person who could have known the private key, KB, is Bob. Recall fromour discussion of RSA in Section 8.2 that knowing the public key, KB+, is of nohelp in learning the private key, KB. Therefore, the only person who could knowKB is the person who generated the pair of keys, (KB+, KB), in the first place, Bob.(Note that this assumes, though, that Bob has not given KB to anyone, nor hasanyone stolen KB from Bob.)694 CHAPTER 8 SECURITY IN COMPUTER NETWORKSEncryptionalgorithmMessage: mBobs privatekey, KBDear Alice:Sorry I have been unableto write for so long. Sincewe…………………….BobSigned message:KB (m)fadfg54986fgnzmcnvT98734ngldskg02jser09tugkjdflg……….Figure 8.10 Creating a digital signature for a documentIt is also important to note that if the original document, m, is ever modified tosome alternate form, m, the signature that Bob created for m will not be valid for m,since KB+(KB(m)) does not equal m. Thus we see that digital signatures also providemessage integrity, allowing the receiver to verify that the message was unaltered aswell as the source of the message.One concern with signing data by encryption is that encryption and decryptionare computationally expensive. Given the overheads of encryption and decryption,signing data via complete encryption/decryption can be overkill. A more efficientapproach is to introduce hash functions into the digital signature. Recall from Sec-tion 8.3.2 that a hash algorithm takes a message, m, of arbitrary length and computesa fixed-length fingerprint of the message, denoted by H(m). Using a hash func-tion, Bob signs the hash of a message rather than the message itself, that is, Bob cal-culates KB(H(m)). Since H(m) is generally much smaller than the original messagem, the computational effort required to create the digital signature is substantiallyreduced.In the context of Bob sending a message to Alice, Figure 8.11 provides a sum-mary of the operational procedure of creating a digital signature. Bob puts his origi-nal long message through a hash function. He then digitally signs the resulting hashBobs privatekey, KBMany-to-onehash functionLong messageDear Alice: This is a VERY long letter since there is so much to say….. ………………..BobFixed-lengthhashOpgmdvboijrtnsdgghPPdogm;lcvkbSignedhashPackage to sendto AliceFgkopdgoo69cmxw54psdterma[asofmzEncryptionalgorithmFigure 8.11 Sending a digitally signed message8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 695with his private key. The original message (in cleartext) along with the digitallysigned message digest (henceforth referred to as the digital signature) is then sent toAlice. Figure 8.12 provides a summary of the operational procedure of the signa-ture. Alice applies the senders public key to the message to obtain a hash result.Alice also applies the hash function to the cleartext message to obtain a second hashresult. If the two hashes match, then Alice can be sure about the integrity and authorof the message.Before moving on, lets briefly compare digital signatures with MACs, sincethey have parallels, but also have important subtle differences. Both digital signaturesand MACs start with a message (or a document). To create a MAC out of the mes-sage, we append an authentication key to the message, and then take the hash of theresult. Note that neither public key nor symmetric key encryption is involved in cre-ating the MAC. To create a digital signature, we first take the hash of the messageand then encrypt the message with our private key (using public key cryptography).696 CHAPTER 8 SECURITY IN COMPUTER NETWORKSBobs publickey, KB+Long messageDear Alice: This is a VERY long letter since there is so much to say….. ………………..BobFixed-lengthhashOpgmdvboijrtnsdgghPPdogm;lcvkbSignedhashFgkopdgoo69cmxw54psdterma[asofmzMany-to-onehash functionCompareFixed-lengthhashOpgmdvboijrtnsdgghPPdogm;lcvkbEncryptionalgorithmFigure 8.12 Verifying a signed messageThus, a digital signature is a heavier technique, since it requires an underlying Public Key Infrastructure (PKI) with certification authorities as described below.Well see in Section 8.4 that PGPa popular secure e-mail systemuses digital signatures for message integrity. Weve seen already that OSPF uses MACs for message integrity. Well see in Sections 8.5 and 8.6 that MACs are also used for pop-ular transport-layer and network-layer security protocols.Public Key CertificationAn important application of digital signatures is public key certification, that is,certifying that a public key belongs to a specific entity. Public key certification isused in many popular secure networking protocols, including IPsec and SSL.To gain insight into this problem, lets consider an Internet-commerce versionof the classic pizza prank. Alice is in the pizza delivery business and acceptsorders over the Internet. Bob, a pizza lover, sends Alice a plaintext message thatincludes his home address and the type of pizza he wants. In this message, Bob alsoincludes a digital signature (that is, a signed hash of the original plaintext message)to prove to Alice that he is the true source of the message. To verify the signature,Alice obtains Bobs public key (perhaps from a public key server or from the e-mailmessage) and checks the digital signature. In this manner she makes sure that Bob,rather than some adolescent prankster, placed the order.This all sounds fine until clever Trudy comes along. As shown in Figure 8.13,Trudy is indulging in a prank. She sends a message to Alice in which she says she isBob, gives Bobs home address, and orders a pizza. In this message she alsoincludes her (Trudys) public key, although Alice naturally assumes it is Bobs pub-lic key. Trudy also attaches a digital signature, which was created with her own(Trudys) private key. After receiving the message, Alice applies Trudys public key(thinking that it is Bobs) to the digital signature and concludes that the plaintextmessage was indeed created by Bob. Bob will be very surprised when the deliveryperson brings a pizza with pepperoni and anchovies to his home!We see from this example that for public key cryptography to be useful, youneed to be able to verify that you have the actual public key of the entity (person,router, browser, and so on) with whom you want to communicate. For example, whenAlice wants to communicate with Bob using public key cryptography, she needs toverify that the public key that is supposed to be Bobs is indeed Bobs.Binding a public key to a particular entity is typically done by a CertificationAuthority (CA), whose job is to validate identities and issue certificates. A CA hasthe following roles:1. A CA verifies that an entity (a person, a router, and so on) is who it says it is.There are no mandated procedures for how certification is done. When dealingwith a CA, one must trust the CA to have performed a suitably rigorous identityverification. For example, if Trudy were able to walk into the Fly-by-Night CA8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 697and simply announce I am Alice and receive certificates associated with theidentity of Alice, then one shouldnt put much faith in public keys certified bythe Fly-by-Night CA. On the other hand, one might (or might not!) be morewilling to trust a CA that is part of a federal or state program. You can trust theidentity associated with a public key only to the extent to which you can trust aCA and its identity verification techniques. What a tangled web of trust we spin!2. Once the CA verifies the identity of the entity, the CA creates a certificate thatbinds the public key of the entity to the identity. The certificate contains thepublic key and globally unique identifying information about the owner of thepublic key (for example, a human name or an IP address). The certificate isdigitally signed by the CA. These steps are shown in Figure 8.14.Let us now see how certificates can be used to combat pizza-orderingpranksters, like Trudy, and other undesirables. When Bob places his order he alsosends his CA-signed certificate. Alice uses the CAs public key to check the validityof Bobs certificate and extract Bobs public key.698 CHAPTER 8 SECURITY IN COMPUTER NETWORKSTrudys privatekey, KTTrudys publickey, KT+Signed (usingTrudy’s private key)message digestFgkopdgoo69cmxw54psdterma[asofmzMessageAlice,Deliver a pizza to me. BobMany-to-onehash functionAlice uses Trudyspublic key, thinkingit is Bobs, andconcludes themessage is from BobPIZZAEncryptionalgorithmFigure 8.13 Trudy masquerades as Bob using public key cryptographyBoth the International Telecommunication Union (ITU) and the IETF havedeveloped standards for CAs. ITU X.509 [ITU 2005a] specifies an authenticationservice as well as a specific syntax for certificates. [RFC 1422] describes CA-basedkey management for use with secure Internet e-mail. It is compatible with X.509 butgoes beyond X.509 by establishing procedures and conventions for a key manage-ment architecture. Table 8.4 describes some of the important fields in a certificate.Bobs CA-signedcertificate containinghis public key, KB+CertificationAuthority (CA)(KB+, B)CAs privatekey, KCAEncryptionalgorithmFigure 8.14 Bob has his public key certified by the CAField Name DescriptionVersion Version number of X.509 specificationSerial number CA-issued unique identifier for a certificateSignature Specifies the algorithm used by CA to sign this certificateIssuer name Identity of CA issuing this certificate, in distinguished name (DN)[RFC 4514] formatValidity period Start and end of period of validity for certificateSubject name Identity of entity whose public key is associated with this certificate, in DN formatSubject public key The subjects public key as well indication of the public key algorithm (and algorithmparameters) to be used with this keyTable 8.4 Selected fields in an X.509 and RFC 1422 public key8.3 MESSAGE INTEGRITY AND DIGITAL SIGNATURES 6998.4 End-Point AuthenticationEnd-point authentication is the process of one entity proving its identity toanother entity over a computer network, for example, a user proving its identity to an email server. As humans, we authenticate each other in many ways: We rec-ognize each others faces when we meet, we recognize each others voices on thetelephone, we are authenticated by the customs official who checks us against thepicture on our passport.In this section, we consider how one party can authenticate another party when the two are communicating over a network. We focus here on authen-ticating a live party, at the point in time when communication is actually occur-ring. A concrete example is a user authenticating him or herself to an e-mailserver. This is a subtly different problem from proving that a message received atsome point in the past did indeed come from that claimed sender, as studied inSection 8.3.When performing authentication over the network, the communicating par-ties cannot rely on biometric information, such as a visual appearance or a voice-print. Indeed, we will see in our later case studies that it is often networkelements such as routers and client/server processes that must authenticate eachother. Here, authentication must be done solely on the basis of messages and dataexchanged as part of an authentication protocol. Typically, an authenticationprotocol would run before the two communicating parties run some other proto-col (for example, a reliable data transfer protocol, a routing informationexchange protocol, or an e-mail protocol). The authentication protocol firstestablishes the identities of the parties to each others satisfaction; only afterauthentication do the parties get down to the work at hand.As in the case of our development of a reliable data transfer (rdt) protocol inChapter 3, we will find it instructive here to develop various versions of an authen-tication protocol, which we will call ap (authentication protocol), and poke holes ineach version as we proceed. (If you enjoy this stepwise evolution of a design, youmight also enjoy [Bryant 1988], which recounts a fictitious narrative betweendesigners of an open-network authentication system, and their discovery of themany subtle issues involved.)Lets assume that Alice needs to authenticate herself to Bob.8.4.1 Authentication Protocol ap1.0Perhaps the simplest authentication protocol we can imagine is one where Alicesimply sends a message to Bob saying she is Alice. This protocol is shown inFigure 8.15. The flaw here is obviousthere is no way for Bob actually to know700 CHAPTER 8 SECURITY IN COMPUTER NETWORKSAliceI am AliceBobTrudyTrudyAliceI am AliceBobFigure 8.15 Protocol ap1.0 and a failure scenario8.4 END-POINT AUTHENTICATION 701that the person sending the message I am Alice is indeed Alice. For example,Trudy (the intruder) could just as well send such a message.8.4.2 Authentication Protocol ap2.0If Alice has a well-known network address (e.g., an IP address) from which shealways communicates, Bob could attempt to authenticate Alice by verifying that thesource address on the IP datagram carrying the authentication message matchesAlices well-known address. In this case, Alice would be authenticated. This mightstop a very network-naive intruder from impersonating Alice, but it wouldnt stopthe determined student studying this book, or many others!From our study of the network and data link layers, we know that it is not thathard (for example, if one had access to the operating system code and could buildones own operating system kernel, as is the case with Linux and several otherfreely available operating systems) to create an IP datagram, put whatever IP sourceaddress we want (for example, Alices well-known IP address) into the IP datagram,and send the datagram over the link-layer protocol to the first-hop router. From thenon, the incorrectly source-addressed datagram would be dutifully forwarded to Bob.This approach, shown in Figure 8.16, is a form of IP spoofing. IP spoofing can beavoided if Trudys first-hop router is configured to forward only datagrams contain-ing Trudys IP source address [RFC 2827]. However, this capability is not univer-sally deployed or enforced. Bob would thus be foolish to assume that Trudysnetwork manager (who might be Trudy herself) had configured Trudys first-hoprouter to forward only appropriately addressed datagrams.8.4.3 Authentication Protocol ap3.0One classic approach to authentication is to use a secret password. The password isa shared secret between the authenticator and the person being authenticated. Gmail,Facebook, telnet, FTP, and many other services use password authentication. In pro-tocol ap3.0, Alice thus sends her secret password to Bob, as shown in Figure 8.17.702 CHAPTER 8 SECURITY IN COMPUTER NETWORKSAliceI am Alice,passwordOKBobTrudyAliceI am Alice,passwordOKBobTrudyTape recorderKey:Figure 8.17 Protocol ap3.0 and a failure scenarioAliceI am AliceAlices IP addr.BobTrudyAliceI am AliceAlices IP addr.BobTrudyFigure 8.16 Protocol ap2.0 and a failure scenarioSince passwords are so widely used, we might suspect that protocol ap3.0is fairly secure. If so, wed be wrong! The security flaw here is clear. If Trudyeavesdrops on Alices communication, then she can learn Alices password. Lestyou think this is unlikely, consider the fact that when you Telnet to anothermachine and log in, the login password is sent unencrypted to the Telnet server.Someone connected to the Telnet client or servers LAN can possibly sniff (read and store) all packets transmitted on the LAN and thus steal the login pass-word. In fact, this is a well-known approach for stealing passwords (see, forexample, [Jimenez 1997]). Such a threat is obviously very real, so ap3.0 clearlywont do.8.4.4 Authentication Protocol ap3.1Our next idea for fixing ap3.0 is naturally to encrypt the password. By encrypting thepassword, we can prevent Trudy from learning Alices password. If we assume thatAlice and Bob share a symmetric secret key, then Alice can encrypt the password and send her identification message, I am Alice, and her encryptedpassword to Bob. Bob then decrypts the password and, assuming the password is correct, authenticates Alice. Bob feels comfortable in authenticating Alice since Alicenot only knows the password, but also knows the shared secret key value needed toencrypt the password. Lets call this protocol ap3.1.While it is true that ap3.1 prevents Trudy from learning Alices password,the use of cryptography here does not solve the authentication problem. Bob issubject to a playback attack: Trudy need only eavesdrop on Alices communica-tion, record the encrypted version of the password, and play back the encryptedversion of the password to Bob to pretend that she is Alice. The use of anencrypted password in ap3.1 doesnt make the situation manifestly different fromthat of protocol ap3.0 in Figure 8.17.8.4.5 Authentication Protocol ap4.0The failure scenario in Figure 8.17 resulted from the fact that Bob could not dis-tinguish between the original authentication of Alice and the later playback ofAlices original authentication. That is, Bob could not tell if Alice was live (thatis, was currently really on the other end of the connection) or whether the mes-sages he was receiving were a recorded playback of a previous authentication ofAlice. The very (very) observant reader will recall that the three-way TCP hand-shake protocol needed to address the same problemthe server side of a TCPconnection did not want to accept a connection if the received SYN segment wasan old copy (retransmission) of a SYN segment from an earlier connection. HowKA-B,8.4 END-POINT AUTHENTICATION 703did the TCP server side solve the problem of determining whether the client wasreally live? It chose an initial sequence number that had not been used in a verylong time, sent that number to the client, and then waited for the client to respondwith an ACK segment containing that number. We can adopt the same idea herefor authentication purposes.A nonce is a number that a protocol will use only once in a lifetime. That is,once a protocol uses a nonce, it will never use that number again. Our ap4.0protocol uses a nonce as follows:1. Alice sends the message I am Alice to Bob.2. Bob chooses a nonce, R, and sends it to Alice.3. Alice encrypts the nonce using Alice and Bobs symmetric secret key, and sends the encrypted nonce, (R), back to Bob. As in protocol ap3.1,it is the fact that Alice knows and uses it to encrypt a value that lets Bob know that the message he receives was generated by Alice. The nonce is used to ensure that Alice is live.4. Bob decrypts the received message. If the decrypted nonce equals the nonce hesent Alice, then Alice is authenticated.Protocol ap4.0 is illustrated in Figure 8.18. By using the once-in-a-lifetimevalue, R, and then checking the returned value, (R), Bob can be sure that Alice is both who she says she is (since she knows the secret key valueneeded to encrypt R) and live (since she has encrypted the nonce, R, that Bob justcreated).The use of a nonce and symmetric key cryptography forms the basis of ap4.0.A natural question is whether we can use a nonce and public key cryptographyKA-BKA-BKA-BKA-B,704 CHAPTER 8 SECURITY IN COMPUTER NETWORKSAliceRKAB(R)I am AliceBobFigure 8.18 Protocol ap4.0 and a failure scenario(rather than symmetric key cryptography) to solve the authentication problem. Thisissue is explored in the problems at the end of the chapter.8.5 Securing E-MailIn previous sections, we examined fundamental issues in network security,including symmetric key and public key cryptography, end-point authentication,key distribution, message integrity, and digital signatures. We are now going toexamine how these tools are being used to provide security in the Internet.Interestingly, it is possible to provide security services in any of the top fourlayers of the Internet protocol stack. When security is provided for a specific appli-cation-layer protocol, the application using the protocol will enjoy one or moresecurity services, such as confidentiality, authentication, or integrity. When securityis provided for a transport-layer protocol, all applications that use that protocolenjoy the security services of the transport protocol. When security is provided atthe network layer on a host-to-host basis, all transport-layer segments (and hence allapplication-layer data) enjoy the security services of the network layer. When secu-rity is provided on a link basis, then the data in all frames traveling over the linkreceive the security services of the link.In Sections 8.5 through 8.8, we examine how security tools are being used inthe application, transport, network, and link layers. Being consistent with the gen-eral structure of this book, we begin at the top of the protocol stack and discusssecurity at the application layer. Our approach is to use a specific application, e-mail, as a case study for application-layer security. We then move down the proto-col stack. Well examine the SSL protocol (which provides security at the transportlayer), IPsec (which provides security at the network layer), and the security of theIEEE 802.11 wireless LAN protocol.You might be wondering why security functionality is being provided atmore than one layer in the Internet. Wouldnt it suffice simply to provide thesecurity functionality at the network layer and be done with it? There are twoanswers to this question. First, although security at the network layer can offerblanket coverage by encrypting all the data in the datagrams (that is, all thetransport-layer segments) and by authenticating all the source IP addresses, itcant provide user-level security. For example, a commerce site cannot rely onIP-layer security to authenticate a customer who is purchasing goods at the com-merce site. Thus, there is a need for security functionality at higher layers as wellas blanket coverage at lower layers. Second, it is generally easier to deploy newInternet services, including security services, at the higher layers of the protocolstack. While waiting for security to be broadly deployed at the network layer,which is probably still many years in the future, many application developers8.5 SECURING E-MAIL 705just do it and introduce security functionality into their favorite applications. Aclassic example is Pretty Good Privacy (PGP), which provides secure e-mail(discussed later in this section). Requiring only client and server applicationcode, PGP was one of the first security technologies to be broadly used in theInternet.8.5.1 Secure E-MailWe now use the cryptographic principles of Sections 8.2 through 8.3 to create asecure e-mail system. We create this high-level design in an incremental manner, ateach step introducing new security services. When designing a secure e-mail sys-tem, let us keep in mind the racy example introduced in Section 8.1the love affairbetween Alice and Bob. Imagine that Alice wants to send an e-mail message to Bob,and Trudy wants to intrude.Before plowing ahead and designing a secure e-mail system for Alice andBob, we should consider which security features would be most desirable forthem. First and foremost is confidentiality. As discussed in Section 8.1, neitherAlice nor Bob wants Trudy to read Alices e-mail message. The second featurethat Alice and Bob would most likely want to see in the secure e-mail system issender authentication. In particular, when Bob receives the message I dontlove you anymore. I never want to see you again. For-merly yours, Alice, he would naturally want to be sure that the messagecame from Alice and not from Trudy. Another feature that the two lovers wouldappreciate is message integrity, that is, assurance that the message Alice sends isnot modified while en route to Bob. Finally, the e-mail system should providereceiver authentication; that is, Alice wants to make sure that she is indeed send-ing the letter to Bob and not to someone else (for example, Trudy) who is imper-sonating Bob.So lets begin by addressing the foremost concern, confidentiality. The moststraightforward way to provide confidentiality is for Alice to encrypt the messagewith symmetric key technology (such as DES or AES) and for Bob to decrypt themessage on receipt. As discussed in Section 8.2, if the symmetric key is longenough, and if only Alice and Bob have the key, then it is extremely difficult foranyone else (including Trudy) to read the message. Although this approach isstraightforward, it has the fundamental difficulty that we discussed in Section8.2distributing a symmetric key so that only Alice and Bob have copies of it. Sowe naturally consider an alternative approachpublic key cryptography (using,for example, RSA). In the public key approach, Bob makes his public key pub-licly available (e.g., in a public key server or on his personal Web page), Aliceencrypts her message with Bobs public key, and she sends the encrypted messageto Bobs e-mail address. When Bob receives the message, he simply decrypts itwith his private key. Assuming that Alice knows for sure that the public key is706 CHAPTER 8 SECURITY IN COMPUTER NETWORKSBobs public key, this approach is an excellent means to provide the desired confi-dentiality. One problem, however, is that public key encryption is relatively inef-ficient, particularly for long messages.To overcome the efficiency problem, lets make use of a session key (dis-cussed in Section 8.2.2). In particular, Alice (1) selects a random symmetric ses-sion key, KS, (2) encrypts her message, m, with the symmetric key, (3) encryptsthe symmetric key with Bobs public key, KB+, (4) concatenates the encryptedmessage and the encrypted symmetric key to form a package, and (5) sends thepackage to Bobs e-mail address. The steps are illustrated in Figure 8.19. (In thisand the subsequent figures, the circled + represents concatenation and the cir-cled represents deconcatenation.) When Bob receives the package, he (1)uses his private key, KB , to obtain the symmetric key, KS, and (2) uses the sym-metric key KS to decrypt the message m.Having designed a secure e-mail system that provides confidentiality, lets nowdesign another system that provides both sender authentication and messageintegrity. Well suppose, for the moment, that Alice and Bob are no longer concernedwith confidentiality (they want to share their feelings with everyone!), and areconcerned only about sender authentication and message integrity. To accomplishthis task, we use digital signatures and message digests, as described in Section 8.3.Specifically, Alice (1) applies a hash function, H (for example, MD5), to hermessage, m, to obtain a message digest, (2) signs the result of the hash function with her private key, KA, to create a digital signature, (3) concatenates the original(unencrypted) message with the signature to create a package, and (4) sends thepackage to Bobs e-mail address. When Bob receives the package, he (1) appliesAlices public key, KA+, to the signed message digest and (2) compares the result ofthis operation with his own hash, H, of the message. The steps are illustrated in8.5 SECURING E-MAIL 707KS (.) KS (.)KS (m) KS (m)KSKSKB+(.)KB+(KS )KB+(KS )m m+ InternetKB(.)Alice sends e-mail message m Bob receives e-mail message mFigure 8.19 Alice used a symmetric session key, KS, to send a secret e-mail to BobFigure 8.20. As discussed in Section 8.3, if the two results are the same, Bob can bepretty confident that the message came from Alice and is unaltered.Now lets consider designing an e-mail system that provides confidentiality,sender authentication, and message integrity. This can be done by combining theprocedures in Figures 8.19 and 8.20. Alice first creates a preliminary package,exactly as in Figure 8.20, that consists of her original message along with a digi-tally signed hash of the message. She then treats this preliminary package as amessage in itself and sends this new message through the sender steps in Figure 8.19,creating a new package that is sent to Bob. The steps applied by Alice are shownin Figure 8.21. When Bob receives the package, he first applies his side of Figure 8.19and then his side of Figure 8.20. It should be clear that this design achieves thegoal of providing confidentiality, sender authentication, and message integrity.Note that, in this scheme, Alice uses public key cryptography twice: once with her own private key and once with Bobs public key. Similarly, Bob alsouses public key cryptography twiceonce with his private key and once withAlices public key.The secure e-mail design outlined in Figure 8.21 probably provides satisfac-tory security for most e-mail users for most occasions. But there is still oneimportant issue that remains to be addressed. The design in Figure 8.21 requiresAlice to obtain Bobs public key, and requires Bob to obtain Alices public key.The distribution of these public keys is a nontrivial problem. For example, Trudymight masquerade as Bob and give Alice her own public key while saying that itis Bobs public key, enabling her to receive the message meant for Bob. As welearned in Section 8.3, a popular approach for securely distributing public keys isto certify the public keys using a CA.708 CHAPTER 8 SECURITY IN COMPUTER NETWORKSH(.) KA (.) KA+ (.)KA (H(m)) KA (H(m))mmm+ InternetAlice sends e-mail message m Bob receives e-mail message mH(.)CompareFigure 8.20 Using hash functions and digital signatures to providesender authentication and message integrity8.5 SECURING E-MAIL 709PHIL ZIMMERMANN AND PGPPhilip R. Zimmermann is the creator of Pretty Good Privacy (PGP). For that, he wasthe target of a three-year criminal investigation because the government held that USexport restrictions for cryptographic software were violated when PGP spread allaround the world following its 1991 publication as freeware. After releasing PGP asshareware, someone else put it on the Internet and foreign citizens downloaded it.Cryptography programs in the United States are classified as munitions under federallaw and may not be exported.Despite the lack of funding, the lack of any paid staff, and the lack of a companyto stand behind it, and despite government interventions, PGP nonetheless becamethe most widely used e-mail encryption software in the world. Oddly enough, the USgovernment may have inadvertently contributed to PGPs spread because of theZimmermann case.The US government dropped the case in early 1996. The announcement was metwith celebration by Internet activists. The Zimmermann case had become the story ofan innocent person fighting for his rights against the abuses of big government. Thegovernments giving in was welcome news, in part because of the campaign forInternet censorship in Congress and the push by the FBI to allow increased govern-ment snooping.After the government dropped its case, Zimmermann founded PGP Inc., whichwas acquired by Network Associates in December 1997. Zimmermann is now anindependent consultant in matters cryptographic.CASE HISTORYH (.) KA (.)KS (.)KSKA (H (m))mm++ to InternetKB+(.)Figure 8.21 Alice uses symmetric key cyptography, public key cryptography, a hash function, and a digital signature toprovide secrecy, sender authentication, and message integrity8.5.2 PGPWritten by Phil Zimmermann in 1991, Pretty Good Privacy (PGP) is an e-mailencryption scheme that has become a de facto standard. Its Web site serves more thana million pages a month to users in 166 countries [PGPI 2012]. Versions of PGP areavailable in the public domain; for example, you can find the PGP software for yourfavorite platform as well as lots of interesting reading at the International PGPHome Page [PGPI 2012]. (A particularly interesting essay by the author of PGP is[Zimmermann 2012].) The PGP design is, in essence, the same as the design shownin Figure 8.21. Depending on the version, the PGP software uses MD5 or SHA forcalculating the message digest; CAST, triple-DES, or IDEA for symmetric keyencryption; and RSA for the public key encryption.When PGP is installed, the software creates a public key pair for the user. Thepublic key can be posted on the users Web site or placed in a public key server. Theprivate key is protected by the use of a password. The password has to be enteredevery time the user accesses the private key. PGP gives the user the option of digi-tally signing the message, encrypting the message, or both digitally signing andencrypting. Figure 8.22 shows a PGP signed message. This message appears afterthe MIME header. The encoded data in the message is KA (H(m)), that is, the digi-tally signed message digest. As we discussed above, in order for Bob to verify theintegrity of the message, he needs to have access to Alices public key.Figure 8.23 shows a secret PGP message. This message also appears after theMIME header. Of course, the plaintext message is not included within the secrete-mail message. When a sender (such as Alice) wants both confidentiality andintegrity, PGP contains a message like that of Figure 8.23 within the message ofFigure 8.22.PGP also provides a mechanism for public key certification, but the mechanismis quite different from the more conventional CA. PGP public keys are certified by aweb of trust. Alice herself can certify any key/username pair when she believes the710 CHAPTER 8 SECURITY IN COMPUTER NETWORKSFigure 8.22 A PGP signed message—–BEGIN PGP SIGNED MESSAGE—–Hash:SHA1Bob:Can I see you tonight?Passionately yours, Alice—–BEGIN PGP SIGNATURE—–Version: PGP for Personal Privacy 5.0Charset: noconvyhHJRHhGJGhgg/12EpJ+lo8gE4vB3mqJhFEvZP9t6n7G6m5Gw2—–END PGP SIGNATURE—–pair really belong together. In addition, PGP permits Alice to say that she trustsanother user to vouch for the authenticity of more keys. Some PGP users sign eachothers keys by holding key-signing parties. Users physically gather, exchangepublic keys, and certify each others keys by signing them with their private keys.8.6 Securing TCP Connections: SSLIn the previous section, we saw how cryptographic techniques can provide confi-dentiality, data integrity, and end-point authentication to a specific application,namely, e-mail. In this section, well drop down a layer in the protocol stack andexamine how cryptography can enhance TCP with security services, including con-fidentiality, data integrity, and end-point authentication. This enhanced version ofTCP is commonly known as Secure Sockets Layer (SSL). A slightly modified ver-sion of SSL version 3, called Transport Layer Security (TLS), has been standard-ized by the IETF [RFC 4346].The SSL protocol was originally designed by Netscape, but the basic ideasbehind securing TCP had predated Netscapes work (for example, see Woo [Woo1994]). Since its inception, SSL has enjoyed broad deployment. SSL is supportedby all popular Web browsers and Web servers, and it is used by essentially allInternet commerce sites (including Amazon, eBay, Yahoo!, MSN, and so on). Tensof billions of dollars are spent over SSL every year. In fact, if you have ever pur-chased anything over the Internet with your credit card, the communicationbetween your browser and the server for this purchase almost certainly went overSSL. (You can identify that SSL is being used by your browser when the URLbegins with https: rather than http.)To understand the need for SSL, lets walk through a typical Internet com-merce scenario. Bob is surfing the Web and arrives at the Alice Incorporated site,which is selling perfume. The Alice Incorporated site displays a form in whichBob is supposed to enter the type of perfume and quantity desired, his address,and his payment card number. Bob enters this information, clicks on Submit, and8.6 SECURING TCP CONNECTIONS: SSL 711Figure 8.23 A secret PGP message—–BEGIN PGP MESSAGE—–Version: PGP for Personal Privacy 5.0u2R4d+/jKmn8Bc5+hgDsqAewsDfrGdszX68liKm5F6Gc4sDfcXytRfdS10juHgbcfDssWe7/K=lKhnMikLo0+1/BvcX4t==Ujk9PbcD4Thdf2awQfgHbnmKlok8iy6gThlp—–END PGP MESSAGEexpects to receive (via ordinary postal mail) the purchased perfumes; he alsoexpects to receive a charge for his order in his next payment card statement. Thisall sounds good, but if no security measures are taken, Bob could be in for a fewsurprises. If no confidentiality (encryption) is used, an intruder could intercept Bobs orderand obtain his payment card information. The intruder could then make pur-chases at Bobs expense. If no data integrity is used, an intruder could modify Bobs order, having himpurchase ten times more bottles of perfume than desired. Finally, if no server authentication is used, a server could display Alice Incorpo-rateds famous logo when in actuality the site maintained by Trudy, who is mas-querading as Alice Incorporated. After receiving Bobs order, Trudy could takeBobs money and run. Or Trudy could carry out an identity theft by collectingBobs name, address, and credit card number.SSL addresses these issues by enhancing TCP with confidentiality, data integrity,server authentication, and client authentication.SSL is often used to provide security to transactions that take place over HTTP.However, because SSL secures TCP, it can be employed by any application thatruns over TCP. SSL provides a simple Application Programmer Interface (API)with sockets, which is similar and analogous to TCPs API. When an applicationwants to employ SSL, the application includes SSL classes/libraries. As shown inFigure 8.24, although SSL technically resides in the application layer, from thedevelopers perspective it is a transport protocol that provides TCPs servicesenhanced with security services.712 CHAPTER 8 SECURITY IN COMPUTER NETWORKSFigure 8.24 Although SSL technically resides in the application layer,from the developers perspective it is a transport-layer protocolTCPSSL sublayerIPApplicationApplicationlayerTCP enhanced with SSLSSL socketTCP socketTCPIPApplicationTCP APITCP socket8.6.1 The Big PictureWe begin by describing a simplified version of SSL, one that will allow us to get abig-picture understanding of the why and how of SSL. We will refer to this simpli-fied version of SSL as almost-SSL. After describing almost-SSL, in the next sub-section well then describe the real SSL, filling in the details. Almost-SSL (andSSL) has three phases: handshake, key derivation, and data transfer. We nowdescribe these three phases for a communication session between a client (Bob) anda server (Alice), with Alice having a private/public key pair and a certificate thatbinds her identity to her public key.HandshakeDuring the handshake phase, Bob needs to (a) establish a TCP connection withAlice, (b) verify that Alice is really Alice, and (c) send Alice a master secret key,which will be used by both Alice and Bob to generate all the symmetric keys theyneed for the SSL session. These three steps are shown in Figure 8.25. Note that oncethe TCP connection is established, Bob sends Alice a hello message. Alice thenresponds with her certificate, which contains her public key. As discussed in Section8.3, because the certificate has been certified by a CA, Bob knows for sure that the8.6 SECURING TCP CONNECTIONS: SSL 713TCP SYNTCP/SYNACKDecrypts EMS withKA to get MSEMS = KA+(MS)TCP ACKSSL hellocertificate(b)(a)(c)Create MasterSecret (MS)Figure 8.25 The almost-SSL handshake, beginning with a TCP connectionpublic key in the certificate belongs to Alice. Bob then generates a Master Secret(MS) (which will only be used for this SSL session), encrypts the MS with Alicespublic key to create the Encyrpted Master Secret (EMS), and sends the EMS toAlice. Alice decrypts the EMS with her private key to get the MS. After this phase,both Bob and Alice (and no one else) know the master secret for this SSL session.Key DerivationIn principle, the MS, now shared by Bob and Alice, could be used as the symmetricsession key for all subsequent encryption and data integrity checking. It is, however,generally considered safer for Alice and Bob to each use different cryptographickeys, and also to use different keys for encryption and integrity checking. Thus, bothAlice and Bob use the MS to generate four keys: EB = session encryption key for data sent from Bob to Alice MB = session MAC key for data sent from Bob to Alice EA = session encryption key for data sent from Alice to Bob MA = session MAC key for data sent from Alice to BobAlice and Bob each generate the four keys from the MS. This could be done by sim-ply slicing the MS into four keys. (But in real SSL it is a little more complicated, aswell see.) At the end of the key derivation phase, both Alice and Bob have all fourkeys. The two encryption keys will be used to encrypt data; the two MAC keys willbe used to verify the integrity of the data.Data TransferNow that Alice and Bob share the same four session keys (EB, MB, EA, and MA),they can start to send secured data to each other over the TCP connection. SinceTCP is a byte-stream protocol, a natural approach would be for SSL to encryptapplication data on the fly and then pass the encrypted data on the fly to TCP. But ifwe were to do this, where would we put the MAC for the integrity check? We cer-tainly do not want to wait until the end of the TCP session to verify the integrity ofall of Bobs data that was sent over the entire session! To address this issue, SSLbreaks the data stream into records, appends a MAC to each record for integritychecking, and then encrypts the record+MAC. To create the MAC, Bob inputs therecord data along with the key MB into a hash function, as discussed in Section 8.3.To encrypt the package record+MAC, Bob uses his session encryption key EB. Thisencrypted package is then passed to TCP for transport over the Internet.Although this approach goes a long way, it still isnt bullet-proof when it comesto providing data integrity for the entire message stream. In particular, suppose Trudyis a woman-in-the-middle and has the ability to insert, delete, and replace segments714 CHAPTER 8 SECURITY IN COMPUTER NETWORKSin the stream of TCP segments sent between Alice and Bob. Trudy, for example,could capture two segments sent by Bob, reverse the order of the segments, adjustthe TCP sequence numbers (which are not encrypted), and then send the two reverse-ordered segments to Alice. Assuming that each TCP segment encapsulates exactlyone record, lets now take a look at how Alice would process these segments.1. TCP running in Alice would think everything is fine and pass the two recordsto the SSL sublayer.2. SSL in Alice would decrypt the two records.3. SSL in Alice would use the MAC in each record to verify the data integrity ofthe two records.4. SSL would then pass the decrypted byte streams of the two records to theapplication layer; but the complete byte stream received by Alice would not bein the correct order due to reversal of the records!You are encouraged to walk through similar scenarios for when Trudy removes seg-ments or when Trudy replays segments.The solution to this problem, as you probably guessed, is to use sequence num-bers. SSL does this as follows. Bob maintains a sequence number counter, whichbegins at zero and is incremented for each SSL record he sends. Bob doesnt actu-ally include a sequence number in the record itself, but when he calculates theMAC, he includes the sequence number in the MAC calculation. Thus, the MAC isnow a hash of the data plus the MAC key MB plus the current sequence number.Alice tracks Bobs sequence numbers, allowing her to verify the data integrity of arecord by including the appropriate sequence number in the MAC calculation. Thisuse of SSL sequence numbers prevents Trudy from carrying out a woman-in-the-middle attack, such as reordering or replaying segments. (Why?)SSL RecordThe SSL record (as well as the almost-SSL record) is shown in Figure 8.26. Therecord consists of a type field, version field, length field, data field, and MAC field.Note that the first three fields are not encrypted. The type field indicates whether therecord is a handshake message or a message that contains application data. It is also8.6 SECURING TCP CONNECTIONS: SSL 715Figure 8.26 Record format for SSLVersion LengthType Data MACEncrypted with EBused to close the SSL connection, as discussed below. SSL at the receiving end usesthe length field to extract the SSL records out of the incoming TCP byte stream. Theversion field is self-explanatory.8.6.2 A More Complete PictureThe previous subsection covered the almost-SSL protocol; it served to give us a basicunderstanding of the why and how of SSL. Now that we have a basic understandingof SSL, we can dig a little deeper and examine the essentials of the actual SSL proto-col. In parallel to reading this description of the SSL protocol, you are encouraged tocomplete the Wireshark SSL lab, available at the textbooks companion Web site.SSL HandshakeSSL does not mandate that Alice and Bob use a specific symmetric key algorithm, aspecific public-key algorithm, or a specific MAC. Instead, SSL allows Alice andBob to agree on the cryptographic algorithms at the beginning of the SSL session,during the handshake phase. Additionally, during the handshake phase, Alice andBob send nonces to each other, which are used in the creation of the session keys(EB, MB, EA, and MA). The steps of the real SSL handshake are as follows:1. The client sends a list of cryptographic algorithms it supports, along with aclient nonce.2. From the list, the server chooses a symmetric algorithm (for example, AES), apublic key algorithm (for example, RSA with a specific key length), and aMAC algorithm. It sends back to the client its choices, as well as a certificateand a server nonce.3. The client verifies the certificate, extracts the servers public key, generates aPre-Master Secret (PMS), encrypts the PMS with the servers public key, andsends the encrypted PMS to the server.4. Using the same key derivation function (as specified by the SSL standard),the client and server independently compute the Master Secret (MS) fromthe PMS and nonces. The MS is then sliced up to generate the two encryptionand two MAC keys. Furthermore, when the chosen symmetric cipher employsCBC (such as 3DES or AES), then two Initialization Vectors (IVs)one foreach side of the connectionare also obtained from the MS. Henceforth, allmessages sent between client and server are encrypted and authenticated(with the MAC).5. The client sends a MAC of all the handshake messages.6. The server sends a MAC of all the handshake messages.The last two steps protect the handshake from tampering. To see this, observethat in step 1, the client typically offers a list of algorithmssome strong, some716 CHAPTER 8 SECURITY IN COMPUTER NETWORKSweak. This list of algorithms is sent in cleartext, since the encryption algorithms andkeys have not yet been agreed upon. Trudy, as a woman-in-the-middle, could deletethe stronger algorithms from the list, forcing the client to select a weak algorithm.To prevent such a tampering attack, in step 5 the client sends a MAC of the concate-nation of all the handshake messages it sent and received. The server can comparethis MAC with the MAC of the handshake messages it received and sent. If there isan inconsistency, the server can terminate the connection. Similarly, the server sendsa MAC of the handshake messages it has seen, allowing the client to check forinconsistencies.You may be wondering why there are nonces in steps 1 and 2. Dont sequencenumbers suffice for preventing the segment replay attack? The answer is yes, but theydont alone prevent the connection replay attack. Consider the following connectionreplay attack. Suppose Trudy sniffs all messages between Alice and Bob. The nextday, Trudy masquerades as Bob and sends to Alice exactly the same sequence of mes-sages that Bob sent to Alice on the previous day. If Alice doesnt use nonces, she willrespond with exactly the same sequence of messages she sent the previous day. Alicewill not suspect any funny business, as each message she receives will pass theintegrity check. If Alice is an e-commerce server, she will think that Bob is placing asecond order (for exactly the same thing). On the other hand, by including a nonce inthe protocol, Alice will send different nonces for each TCP session, causing theencryption keys to be different on the two days. Therefore, when Alice receivesplayed-back SSL records from Trudy, the records will fail the integrity checks, and thebogus e-commerce transaction will not succeed. In summary, in SSL, nonces are usedto defend against the connection replay attack and sequence numbers are used todefend against replaying individual packets during an ongoing session.Connection ClosureAt some point, either Bob or Alice will want to end the SSL session. One approachwould be to let Bob end the SSL session by simply terminating the underlying TCPconnectionthat is, by having Bob send a TCP FIN segment to Alice. But such anaive design sets the stage for the truncation attack whereby Trudy once again getsin the middle of an ongoing SSL session and ends the session early with a TCP FIN.If Trudy were to do this, Alice would think she received all of Bobs data when actu-ality she only received a portion of it. The solution to this problem is to indicate inthe type field whether the record serves to terminate the SSL session. (Although theSSL type is sent in the clear, it is authenticated at the receiver using the recordsMAC.) By including such a field, if Alice were to receive a TCP FIN before receiv-ing a closure SSL record, she would know that something funny was going on.This completes our introduction to SSL. Weve seen that it uses many of thecryptography principles discussed in Sections 8.2 and 8.3. Readers who want toexplore SSL on yet a deeper level can read Rescorlas highly readable book on SSL[Rescorla 2001].8.6 SECURING TCP CONNECTIONS: SSL 7178.7 Network-Layer Security: IPsec and Virtual Private NetworksThe IP security protocol, more commonly known as IPsec, provides security at thenetwork layer. IPsec secures IP datagrams between any two network-layer entities,including hosts and routers. As we will soon describe, many institutions (corpora-tions, government branches, non-profit organizations, and so on) use IPsec to createvirtual private networks (VPNs) that run over the public Internet.Before getting into the specifics of IPsec, lets step back and consider what itmeans to provide confidentiality at the network layer. With network-layer confiden-tiality between a pair of network entities (for example, between two routers, betweentwo hosts, or between a router and a host), the sending entity encrypts the payloadsof all the datagrams it sends to the receiving entity. The encrypted payload could be aTCP segment, a UDP segment, an ICMP message, and so on. If such a network-layerservice were in place, all data sent from one entity to the otherincluding e-mail,Web pages, TCP handshake messages, and management messages (such as ICMPand SNMP)would be hidden from any third party that might be sniffing the net-work. For this reason, network-layer security is said to provide blanket coverage.In addition to confidentiality, a network-layer security protocol could potentiallyprovide other security services. For example, it could provide source authentication, sothat the receiving entity can verify the source of the secured datagram. A network-layersecurity protocol could provide data integrity, so that the receiving entity can check forany tampering of the datagram that may have occurred while the datagram was in tran-sit. A network-layer security service could also provide replay-attack prevention, mean-ing that Bob could detect any duplicate datagrams that an attacker might insert. We willsoon see that IPsec indeed provides mechanisms for all these security services, that is,for confidentiality, source authentication, data integrity, and replay-attack prevention.8.7.1 IPsec and Virtual Private Networks (VPNs)An institution that extends over multiple geographical regions often desires its ownIP network, so that its hosts and servers can send data to each other in a secure andconfidential manner. To achieve this goal, the institution could actually deploy astand-alone physical networkincluding routers, links, and a DNS infrastructurethatis completely separate from the public Internet. Such a disjoint network, dedicatedto a particular institution, is called a private network. Not surprisingly, a privatenetwork can be very costly, as the institution needs to purchase, install, and main-tain its own physical network infrastructure.Instead of deploying and maintaining a private network, many institutions todaycreate VPNs over the existing public Internet. With a VPN, the institutions inter-office traffic is sent over the public Internet rather than over a physically independentnetwork. But to provide confidentiality, the inter-office traffic is encrypted before itenters the public Internet. A simple example of a VPN is shown in Figure 8.27. Here718 CHAPTER 8 SECURITY IN COMPUTER NETWORKSthe institution consists of a headquarters, a branch office, and traveling salespersonsthat typically access the Internet from their hotel rooms. (There is only one salesper-son shown in the figure.) In this VPN, whenever two hosts within headquarters sendIP datagrams to each other or whenever two hosts within the branch office want tocommunicate, they use good-old vanilla IPv4 (that is, without IPsec services). How-ever, when two of the institutions hosts communicate over a path that traverses thepublic Internet, the traffic is encrypted before it enters the Internet.To get a feel for how a VPN works, lets walk through a simple example in thecontext of Figure 8.27. When a host in headquarters sends an IP datagram to a sales-person in a hotel, the gateway router in headquarters converts the vanilla IPv4 data-gram into an IPsec datagram and then forwards this IPsec datagram into the Internet.This IPsec datagram actually has a traditional IPv4 header, so that the routers in thepublic Internet process the datagram as if it were an ordinary IPv4 datagramtothem, the datagram is a perfectly ordinary datagram. But, as shown Figure 8.27, thepayload of the IPsec datagram includes an IPsec header, which is used for IPsec proc-essing; furthermore, the payload of the IPsec datagram is encrypted. When the IPsecdatagram arrives at the salespersons laptop, the OS in the laptop decrypts the pay-load (and provides other security services, such as verifying data integrity) and passesthe unencrypted payload to the upper-layer protocol (for example, to TCP or UDP).8.7 NETWORK-LAYER SECURITY: IPSEC AND VIRTUAL PRIVATE NETWORKS 719Figure 8.27 Virtual Private Network (VPN)IPheaderIPsecheaderSecurepayloadIPheaderIPsecheaderSecurepayloadIPheaderIPsecheaderSecurepayloadIPheaderPayloadIPheaderPayloadLaptop w/IPsecRouterw/IPv4 andIPsecRouterw/IPv4 andIPsecBranch OfficeHeadquartersSalespersonin HotelPublicInternetWe have just given a high-level overview of how an institution can employIPsec to create a VPN. To see the forest through the trees, we have brushed asidemany important details. Lets now take a closer look.8.7.2 The AH and ESP ProtocolsIPsec is a rather complex animalit is defined in more than a dozen RFCs. Twoimportant RFCs are RFC 4301, which describes the overall IP security architecture,and RFC 6071, which provides an overview of the IPsec protocol suite. Our goal inthis textbook, as usual, is not simply to re-hash the dry and arcane RFCs, but insteadtake a more operational and pedagogic approach to describing the protocols.In the IPsec protocol suite, there are two principal protocols: the AuthenticationHeader (AH) protocol and the Encapsulation Security Payload (ESP) protocol.When a source IPsec entity (typically a host or a router) sends secure datagrams to adestination entity (also a host or a router), it does so with either the AH protocol or theESP protocol. The AH protocol provides source authentication and data integrity butdoes not provide confidentiality. The ESP protocol provides source authentication,data integrity, and confidentiality. Because confidentiality is often critical for VPNsand other IPsec applications, the ESP protocol is much more widely used than the AHprotocol. In order to de-mystify IPsec and avoid much of its complication, we willhenceforth focus exclusively on the ESP protocol. Readers wanting to learn also aboutthe AH protocol are encouraged to explore the RFCs and other online resources.8.7.3 Security AssociationsIPsec datagrams are sent between pairs of network entities, such as between two hosts,between two routers, or between a host and router. Before sending IPsec datagramsfrom source entity to destination entity, the source and destination entities create a net-work-layer logical connection. This logical connection is called a security association(SA). An SA is a simplex logical connection; that is, it is unidirectional from source todestination. If both entities want to send secure datagrams to each other, then two SAs(that is, two logical connections) need to be established, one in each direction.For example, consider once again the institutional VPN in Figure 8.27. This insti-tution consists of a headquarters office, a branch office and, say, n traveling salesper-sons. For the sake of example, lets suppose that there is bi-directional IPsec trafficbetween headquarters and the branch office and bi-directional IPsec traffic betweenheadquarters and the salespersons. In this VPN, how many SAs are there? To answerthis question, note that there are two SAs between the headquarters gateway router andthe branch-office gateway router (one in each direction); for each salespersons laptop,there are two SAs between the headquarters gateway router and the laptop (again, onein each direction). So, in total, there are (2 + 2n) SAs. Keep in mind, however, that notall traffic sent into the Internet by the gateway routers or by the laptops will be IPsecsecured. For example, a host in headquarters may want to access a Web server (such asAmazon or Google) in the public Internet. Thus, the gateway router (and the laptops)will emit into the Internet both vanilla IPv4 datagrams and secured IPsec datagrams.720 CHAPTER 8 SECURITY IN COMPUTER NETWORKSLets now take a look inside an SA. To make the discussion tangible and con-crete, lets do this in the context of an SA from router R1 to router R2 in Figure 8.28.(You can think of Router R1 as the headquarters gateway router and Router R2 asthe branch office gateway router from Figure 8.27.) Router R1 will maintain stateinformation about this SA, which will include: A 32-bit identifier for the SA, called the Security Parameter Index (SPI) The origin interface of the SA (in this case 200.168.1.100) and the destinationinterface of the SA (in this case 193.68.2.23) The type of encryption to be used (for example, 3DES with CBC) The encryption key The type of integrity check (for example, HMAC with MD5) The authentication keyWhenever router R1 needs to construct an IPsec datagram for forwarding overthis SA, it accesses this state information to determine how it should authenticateand encrypt the datagram. Similarly, router R2 will maintain the same state infor-mation for this SA and will use this information to authenticate and decrypt anyIPsec datagram that arrives from the SA.An IPsec entity (router or host) often maintains state information for many SAs.For example, in the VPN example in Figure 8.27 with n salespersons, the headquar-ters gateway router maintains state information for (2 + 2n) SAs. An IPsec entitystores the state information for all of its SAs in its Security Association Database(SAD), which is a data structure in the entitys OS kernel.8.7.4 The IPsec DatagramHaving now described SAs, we can now describe the actual IPsec datagram. IPsechas two different packet forms, one for the so-called tunnel mode and the other forthe so-called transport mode. The tunnel mode, being more appropriate for VPNs,is more widely deployed than the transport mode. In order to further de-mystify8.7 NETWORK-LAYER SECURITY: IPSEC AND VIRTUAL PRIVATE NETWORKS 721Figure 8.28 Security Association (SA) from R1 to R2InternetSAR1172.16.1/24Headquarters Branch Office200.168.1.100 193.68.2.23172.16.2/24R2IPsec and avoid much of its complication, we henceforth focus exclusively on thetunnel mode. Once you have a solid grip on the tunnel mode, you should be able toeasily learn about the transport mode on your own.The packet format of the IPsec datagram is shown in Figure 8.29. You mightthink that packet formats are boring and insipid, but we will soon see that the IPsecdatagram actually looks and tastes like a popular Tex-Mex delicacy! Lets examinethe IPsec fields in the context of Figure 8.28. Suppose router R1 receives an ordi-nary IPv4 datagram from host 172.16.1.17 (in the headquarters network) which isdestined to host 172.16.2.48 (in the branch-office network). Router R1 uses the fol-lowing recipe to convert this original IPv4 datagram into an IPsec datagram: Appends to the back of the original IPv4 datagram (which includes the originalheader fields!) an ESP trailer field Encrypts the result using the algorithm and key specified by the SA Appends to the front of this encrypted quantity a field called ESP header; theresulting package is called the enchilada Creates an authentication MAC over the whole enchilada using the algorithmand key specified in the SA Appends the MAC to the back of the enchilada forming the payload Finally, creates a brand new IP header with all the classic IPv4 header fields(together normally 20 bytes long), which it appends before the payloadNote that the resulting IPsec datagram is a bona fide IPv4 datagram, with thetraditional IPv4 header fields followed by a payload. But in this case, the payloadcontains an ESP header, the original IP datagram, an ESP trailer, and an ESP authen-tication field (with the original datagram and ESP trailer encrypted). The original IPdatagram has 172.16.1.17 for the source IP address and 172.16.2.48 for the destina-tion IP address. Because the IPsec datagram includes the original IP datagram, theseaddresses are included (and encrypted) as part of the payload of the IPsec packet.But what about the source and destination IP addresses that are in the new IP header,that is, in the left-most header of the IPsec datagram? As you might expect, they areset to the source and destination router interfaces at the two ends of the tunnels,namely, 200.168.1.100 and 193.68.2.23. Also, the protocol number in this new IPv4header field is not set to that of TCP, UDP, or SMTP, but instead to 50, designatingthat this is an IPsec datagram using the ESP protocol.After R1 sends the IPsec datagram into the public Internet, it will pass throughmany routers before reaching R2. Each of these routers will process the datagram asif it were an ordinary datagramthey are completely oblivious to the fact that thedatagram is carrying IPsec-encrypted data. For these public Internet routers, becausethe destination IP address in the outer header is R2, the ultimate destination of thedatagram is R2.Having walked through an example of how an IPsec datagram is constructed, letsnow take a closer look at the ingredients in the enchilada. We see in Figure 8.29722 CHAPTER 8 SECURITY IN COMPUTER NETWORKSthat the ESP trailer consists of three fields: padding; pad length; and next header.Recall that block ciphers require the message to be encrypted to be an integer multi-ple of the block length. Padding (consisting of meaningless bytes) is used so thatwhen added to the original datagram (along with the pad length and next headerfields), the resulting message is an integer number of blocks. The pad-length fieldindicates to the receiving entity how much padding was inserted (and thus needs tobe removed). The next header identifies the type (e.g., UDP) of data contained in thepayload-data field. The payload data (typically the original IP datagram) and theESP trailer are concatenated and then encrypted.Appended to the front of this encrypted unit is the ESP header, which is sent inthe clear and consists of two fields: the SPI and the sequence number field. The SPIindicates to the receiving entity the SA to which the datagram belongs; the receivingentity can then index its SAD with the SPI to determine the appropriate authentica-tion/decryption algorithms and keys. The sequence number field is used to defendagainst replay attacks.The sending entity also appends an authentication MAC. As stated earlier, thesending entity calculates a MAC over the whole enchilada (consisting of the ESPheader, the original IP datagram, and the ESP trailerwith the datagram and trailerbeing encrypted). Recall that to calculate a MAC, the sender appends a secret MACkey to the enchilada and then calculates a fixed-length hash of the result.When R2 receives the IPsec datagram, R2 observes that the destination IPaddress of the datagram is R2 itself. R2 therefore processes the datagram. Becausethe protocol field (in the left-most IP header) is 50, R2 sees that it should applyIPsec ESP processing to the datagram. First, peering into the enchilada, R2 uses theSPI to determine to which SA the datagram belongs. Second, it calculates the MACof the enchilada and verifies that the MAC is consistent with the value in the ESPMAC field. If it is, it knows that the enchilada comes from R1 and has not been tam-pered with. Third, it checks the sequence-number field to verify that the datagram isfresh (and not a replayed datagram). Fourth, it decrypts the encrypted unit using the8.7 NETWORK-LAYER SECURITY: IPSEC AND VIRTUAL PRIVATE NETWORKS 723Figure 8.29 IPsec datagram formatNew IPheaderESPheaderESPtrailerESPMACOriginalIP headerOriginal IPdatagram payloadEncryptedEnchilada authenticatedPadlengthPaddingNextheaderSPI Seq #decryption algorithm and key associated with the SA. Fifth, it removes padding andextracts the original, vanilla IP datagram. And finally, sixth, it forwards the originaldatagram into the branch office network towards its ultimate destination. Whew,what a complicated recipe, huh? Well no one ever said that preparing and unravel-ing an enchilada was easy!There is actually another important subtlety that needs to be addressed. It cen-ters on the following question: When R1 receives an (unsecured) datagram from ahost in the headquarters network, and that datagram is destined to some destina-tion IP address outside of headquarters, how does R1 know whether it should beconverted to an IPsec datagram? And if it is to be processed by IPsec, how doesR1 know which SA (of many SAs in its SAD) should be used to construct theIPsec datagram? The problem is solved as follows. Along with a SAD, the IPsecentity also maintains another data structure called the Security Policy Database(SPD). The SPD indicates what types of datagrams (as a function of source IPaddress, destination IP address, and protocol type) are to be IPsec processed; andfor those that are to be IPsec processed, which SA should be used. In a sense, theinformation in a SPD indicates what to do with an arriving datagram; the infor-mation in the SAD indicates how to do it.Summary of IPsec ServicesSo what services does IPsec provide, exactly? Let us examine these services fromthe perspective of an attacker, say Trudy, who is a woman-in-the-middle, sittingsomewhere on the path between R1 and R2 in Figure 8.28. Assume throughout thisdiscussion that Trudy does not know the authentication and encryption keys used bythe SA. What can and cannot Trudy do? First, Trudy cannot see the original data-gram. If fact, not only is the data in the original datagram hidden from Trudy, but sois the protocol number, the source IP address, and the destination IP address. Fordatagrams sent over the SA, Trudy only knows that the datagram originated fromsome host in 172.16.1.0/24 and is destined to some host in 172.16.2.0/24. She doesnot know if it is carrying TCP, UDP, or ICMP data; she does not know if it is carry-ing HTTP, SMTP, or some other type of application data. This confidentiality thusgoes a lot farther than SSL. Second, suppose Trudy tries to tamper with a datagramin the SA by flipping some of its bits. When this tampered datagram arrives at R2, itwill fail the integrity check (using the MAC), thwarting Trudys vicious attemptsonce again. Third, suppose Trudy tries to masquerade as R1, creating a IPsec data-gram with source 200.168.1.100 and destination 193.68.2.23. Trudys attack will befutile, as this datagram will again fail the integrity check at R2. Finally, becauseIPsec includes sequence numbers, Trudy will not be able create a successful replayattack. In summary, as claimed at the beginning of this section, IPsec providesbetween any pair of devices that process packets through the network layerconfidentiality, source authentication, data integrity, and replay-attack prevention.724 CHAPTER 8 SECURITY IN COMPUTER NETWORKS8.7.5 IKE: Key Management in IPsecWhen a VPN has a small number of end points (for example, just two routers as inFigure 8.28), the network administrator can manually enter the SA information(encryption/authentication algorithms and keys, and the SPIs) into the SADs of theendpoints. Such manual keying is clearly impractical for a large VPN, which mayconsist of hundreds or even thousands of IPsec routers and hosts. Large, geographi-cally distributed deployments require an automated mechanism for creating theSAs. IPsec does this with the Internet Key Exchange (IKE) protocol, specified inRFC 5996.IKE has some similarities with the handshake in SSL (see Section 8.6). EachIPsec entity has a certificate, which includes the entitys public key. As with SSL, theIKE protocol has the two entities exchange certificates, negotiate authentication andencryption algorithms, and securely exchange key material for creating session keysin the IPsec SAs. Unlike SSL, IKE employs two phases to carry out these tasks.Lets investigate these two phases in the context of two routers, R1 and R2, inFigure 8.28. The first phase consists of two exchanges of message pairs between R1and R2: During the first exchange of messages, the two sides use Diffie-Hellman (seeHomework Problems) to create a bi-directional IKE SA between the routers. Tokeep us all confused, this bi-directional IKE SA is entirely different from theIPsec SAs discussed in Sections 8.6.3 and 8.6.4. The IKE SA provides an authen-ticated and encrypted channel between the two routers. During this first mes-sage-pair exchange, keys are established for encryption and authentication forthe IKE SA. Also established is a master secret that will be used to computeIPSec SA keys later in phase 2. Observe that during this first step, RSA publicand private keys are not used. In particular, neither R1 nor R2 reveals its identityby signing a message with its private key. During the second exchange of messages, both sides reveal their identity to eachother by signing their messages. However, the identities are not revealed to apassive sniffer, since the messages are sent over the secured IKE SA channel.Also during this phase, the two sides negotiate the IPsec encryption and authen-tication algorithms to be employed by the IPsec SAs.In phase 2 of IKE, the two sides create an SA in each direction. At the end ofphase 2, the encryption and authentication session keys are established on both sidesfor the two SAs. The two sides can then use the SAs to send secured datagrams, asdescribed in Sections 8.7.3 and 8.7.4. The primary motivation for having two phasesin IKE is computational costsince the second phase doesnt involve any public-key cryptography, IKE can generate a large number of SAs between the two IPsecentities with relatively little computational cost.8.7 NETWORK-LAYER SECURITY: IPSEC AND VIRTUAL PRIVATE NETWORKS 7258.8 Securing Wireless LANsSecurity is a particularly important concern in wireless networks, where radio wavescarrying frames can propagate far beyond the building containing the wireless basestation and hosts. In this section we present a brief introduction to wireless security.For a more in-depth treatment, see the highly readable book by Edney and Arbaugh[Edney 2003].The issue of security in 802.11 has attracted considerable attention in both techni-cal circles and in the media. While there has been considerable discussion, there hasbeen little debatethere seems to be universal agreement that the original 802.11specification contains a number of serious security flaws. Indeed, public domain soft-ware can now be downloaded that exploits these holes, making those who use thevanilla 802.11 security mechanisms as open to security attacks as users who use nosecurity features at all.In the following section, we discuss the security mechanisms initially standard-ized in the 802.11 specification, known collectively as Wired Equivalent Privacy(WEP). As the name suggests, WEP is meant to provide a level of security similarto that found in wired networks. Well then discuss a few of the security holes inWEP and discuss the 802.11i standard, a fundamentally more secure version of802.11 adopted in 2004.8.8.1 Wired Equivalent Privacy (WEP)The IEEE 802.11 WEP protocol was designed in 1999 to provide authentication anddata encryption between a host and a wireless access point (that is, base station)using a symmetric shared key approach. WEP does not specify a key managementalgorithm, so it is assumed that the host and wireless access point have somehowagreed on the key via an out-of-band method. Authentication is carried out as follows:1. A wireless host requests authentication by an access point.2. The access point responds to the authentication request with a 128-byte noncevalue.3. The wireless host encrypts the nonce using the symmetric key that it shareswith the access point.4. The access point decrypts the host-encrypted nonce.If the decrypted nonce matches the nonce value originally sent to the host, then thehost is authenticated by the access point.The WEP data encryption algorithm is illustrated in Figure 8.30. A secret 40-bitsymmetric key, KS, is assumed to be known by both a host and the access point. Inaddition, a 24-bit Initialization Vector (IV) is appended to the 40-bit key to create a64-bit key that will be used to encrypt a single frame. The IV will change from one726 CHAPTER 8 SECURITY IN COMPUTER NETWORKSframe to another, and hence each frame will be encrypted with a different 64-bit key.Encryption is performed as follows. First a 4-byte CRC value (see Section 5.2) iscomputed for the data payload. The payload and the four CRC bytes are thenencrypted using the RC4 stream cipher. We will not cover the details of RC4 here(see [Schneier 1995] and [Edney 2003] for details). For our purposes, it is enough toknow that when presented with a key value (in this case, the 64-bit (KS, IV) key), theRC4 algorithm produces a stream of key values, k1IV, k2IV, k3IV, . . . that are used toencrypt the data and CRC value in a frame. For practical purposes, we can think ofthese operations being performed a byte at a time. Encryption is performed byXOR-ing the ith byte of data, di, with the ith key, kiIV, in the stream of key valuesgenerated by the (KS,IV) pair to produce the ith byte of ciphertext, ci:ci = di kiIVThe IV value changes from one frame to the next and is included in plaintext inthe header of each WEP-encrypted 802.11 frame, as shown in Figure 8.30. Thereceiver takes the secret 40-bit symmetric key that it shares with the sender, appendsthe IV, and uses the resulting 64-bit key (which is identical to the key used by thesender to perform encryption) to decrypt the frame:di = ci kiIVProper use of the RC4 algorithm requires that the same 64-bit key value neverbe used more than once. Recall that the WEP key changes on a frame-by-framebasis. For a given KS (which changes rarely, if ever), this means that there are only224 unique keys. If these keys are chosen randomly, we can show [Walker 2000;Edney 2003] that the probability of having chosen the same IV value (and henceused the same 64-bit key) is more than 99 percent after only 12,000 frames. With 1Kbyte frame sizes and a data transmission rate of 11 Mbps, only a few seconds are8.8 SECURING WIRELESS LANS 727Figure 8.30 802.11 WEP protocolKey sequence generator(for given Ks, IV)k1IVd1c1k2IV k3IV kNIV IVkN+1IVkN+4Ks: 40-bit secret symmetricPlaintext frame data plus CRCIV (per frame)802.11headerIVWEP-encrypted dataplus CRCd2c2d3c3dNcNCRC1cN+1 cN+4CRC4needed before 12,000 frames are transmitted. Furthermore, since the IV is transmit-ted in plaintext in the frame, an eavesdropper will know whenever a duplicate IVvalue is used.To see one of the several problems that occur when a duplicate key is used, con-sider the following chosen-plaintext attack taken by Trudy against Alice. Supposethat Trudy (possibly using IP spoofing) sends a request (for example, an HTTP orFTP request) to Alice to transmit a file with known content, d1, d2, d3, d4,. . . . Trudyalso observes the encrypted data c1, c2, c3, c4. . . . Since di = ci kiIV, if we XOR ciwith each side of this equality we havedi ci = kiIVWith this relationship, Trudy can use the known values of di and ci to compute kiIV.The next time Trudy sees the same value of IV being used, she will know the keysequence k1IV, k2IV, k3IV, . . . and will thus be able to decrypt the encrypted message.There are several additional security concerns with WEP as well. [Fluhrer2001] described an attack exploiting a known weakness in RC4 when certain weakkeys are chosen. [Stubblefield 2002] discusses efficient ways to implement andexploit this attack. Another concern with WEP involves the CRC bits shown in Fig-ure 8.30 and transmitted in the 802.11 frame to detect altered bits in the payload.However, an attacker who changes the encrypted content (e.g., substituting gibber-ish for the original encrypted data), computes a CRC over the substituted gibberish,and places the CRC into a WEP frame can produce an 802.11 frame that will beaccepted by the receiver. What is needed here are message integrity techniques suchas those we studied in Section 8.3 to detect content tampering or substitution. Formore details of WEP security, see [Edney 2003; Walker 2000; Weatherspoon 2000]and the references therein.8.8.2 IEEE 802.11iSoon after the 1999 release of IEEE 802.11, work began on developing a new andimproved version of 802.11 with stronger security mechanisms. The new standard,known as 802.11i, underwent final ratification in 2004. As well see, while WEPprovided relatively weak encryption, only a single way to perform authentication,and no key distribution mechanisms, IEEE 802.11i provides for much strongerforms of encryption, an extensible set of authentication mechanisms, and a key dis-tribution mechanism. In the following, we present an overview of 802.11i; an excel-lent (streaming audio) technical overview of 802.11i is [TechOnline 2012].Figure 8.31 overviews the 802.11i framework. In addition to the wireless clientand access point, 802.11i defines an authentication server with which the AP cancommunicate. Separating the authentication server from the AP allows one authenti-cation server to serve many APs, centralizing the (often sensitive) decisions728 CHAPTER 8 SECURITY IN COMPUTER NETWORKSregarding authentication and access within the single server, and keeping AP costsand complexity low. 802.11i operates in four phases:1. Discovery. In the discovery phase, the AP advertises its presence and the formsof authentication and encryption that can be provided to the wireless clientnode. The client then requests the specific forms of authentication and encryp-tion that it desires. Although the client and AP are already exchanging mes-sages, the client has not yet been authenticated nor does it have an encryptionkey, and so several more steps will be required before the client can communi-cate with an arbitrary remote host over the wireless channel.2. Mutual authentication and Master Key (MK) generation. Authentication takesplace between the wireless client and the authentication server. In this phase,the access point acts essentially as a relay, forwarding messages between theclient and the authentication server. The Extensible Authentication Protocol(EAP) [RFC 3748] defines the end-to-end message formats used in a simplerequest/response mode of interaction between the client and authenticationserver. As shown in Figure 8.32 EAP messages are encapsulated usingEAPoL (EAP over LAN, [IEEE 802.1X]) and sent over the 802.11 wirelesslink. These EAP messages are then decapsulated at the access point, and then8.8 SECURING WIRELESS LANS 729Figure 8.31 802.11i: four phases of operationSTA:client stationAP:access pointWirednetworkAS:authenticationserver1Discovery ofsecurity capabilities4STA, AP use PMK to deriveTemporal Key (TK) used formessage encryption, integrityAS derives same PMK,sends to APSTA derives Pairwise Master Key (PMK)233STA and AS mutually authenticate, together generateMaster Key (MK). AP serves as pass throughre-encapsulated using the RADIUS protocol for transmission over UDP/IP tothe authentication server. While the RADIUS server and protocol [RFC 2865]are not required by the 802.11i protocol, they are de facto standard compo-nents for 802.11i. The recently standardized DIAMETER protocol [RFC3588] is likely to replace RADIUS in the near future.With EAP, the authentication server can choose one of a number of ways toperform authentication. While 802.11i does not mandate a particular authenti-cation method, the EAP-TLS authentication scheme [RFC 5216] is often used.EAP-TLS uses public key techniques (including nonce encryption and messagedigests) similar to those we studied in Section 8.3 to allow the client and theauthentication server to mutually authenticate each other, and to derive a Master Key (MK) that is known to both parties.3. Pairwise Master Key (PMK) generation. The MK is a shared secret knownonly to the client and the authentication server, which they each use to generatea second key, the Pairwise Master Key (PMK). The authentication server thensends the PMK to the AP. This is where we wanted to be! The client and APnow have a shared key (recall that in WEP, the problem of key distribution wasnot addressed at all) and have mutually authenticated each other. Theyre justabout ready to get down to business.4. Temporal Key (TK) generation. With the PMK, the wireless client and AP cannow generate additional keys that will be used for communication. Of particularinterest is the Temporal Key (TK), which will be used to perform the link-levelencryption of data sent over the wireless link and to an arbitrary remote host.730 CHAPTER 8 SECURITY IN COMPUTER NETWORKSFigure 8.32 EAP is an end-to-end protocol. EAP messages are encapsu-lated using EAPoL over the wireless link between the clientand the access point, and using RADIUS over UDP/IPbetween the access point and the authentication serverSTA:client stationAP:access pointWirednetworkAS:authenticationserverEAP TLSEAPEAP over LAN (EAPoL) RADIUSIEEE 802.11 UDP/IP802.11i provides several forms of encryption, including an AES-based encryptionscheme and a strengthened version of WEP encryption.8.9 Operational Security: Firewalls and Intrusion Detection SystemsWeve seen throughout this chapter that the Internet is not a very safe placebadguys are out there, wreaking all sorts of havoc. Given the hostile nature of theInternet, lets now consider an organizations network and the network administra-tor who administers it. From a network administrators point of view, the worlddivides quite neatly into two campsthe good guys (who belong to the organiza-tions network, and who should be able to access resources inside the organi-zations network in a relatively unconstrained manner) and the bad guys (everyoneelse, whose access to network resources must be carefully scrutinized). In manyorganizations, ranging from medieval castles to modern corporate office buildings,there is a single point of entry/exit where both good guys and bad guys enteringand leaving the organization are security-checked. In a castle, this was done at agate at one end of the drawbridge; in a corporate building, this is done at the secu-rity desk. In a computer network, when traffic entering/leaving a network is secu-rity-checked, logged, dropped, or forwarded, it is done by operational devicesknown as firewalls, intrusion detection systems (IDSs), and intrusion preventionsystems (IPSs).8.9.1 FirewallsA firewall is a combination of hardware and software that isolates an organizationsinternal network from the Internet at large, allowing some packets to pass and block-ing others. A firewall allows a network administrator to control access between theoutside world and resources within the administered network by managing the traf-fic flow to and from these resources. A firewall has three goals: All traffic from outside to inside, and vice versa, passes through the firewall.Figure 8.33 shows a firewall, sitting squarely at the boundary between the admin-istered network and the rest of the Internet. While large organizations may usemultiple levels of firewalls or distributed firewalls [Skoudis 2006], locating afirewall at a single access point to the network, as shown in Figure 8.33, makesit easier to manage and enforce a security-access policy. Only authorized traffic, as defined by the local security policy, will be allowed topass. With all traffic entering and leaving the institutional network passingthrough the firewall, the firewall can restrict access to authorized traffic.8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 731 The firewall itself is immune to penetration. The firewall itself is a device con-nected to the network. If not designed or installed properly, it can be compro-mised, in which case it provides only a false sense of security (which is worsethan no firewall at all!).Cisco and Check Point are two of the leading firewall vendors today. You can alsoeasily create a firewall (packet filter) from a Linux box using iptables (public-domain software that is normally shipped with Linux).Firewalls can be classified in three categories: traditional packet filters,stateful filters, and application gateways. Well cover each of these in turn in thefollowing subsections.Traditional Packet FiltersAs shown in Figure 8.33, an organization typically has a gateway router connectingits internal network to its ISP (and hence to the larger public Internet). All traffic leav-ing and entering the internal network passes through this router, and it is at this routerwhere packet filtering occurs. A packet filter examines each datagram in isolation,determining whether the datagram should be allowed to pass or should be droppedbased on administrator-specific rules. Filtering decisions are typically based on:732 CHAPTER 8 SECURITY IN COMPUTER NETWORKSFigure 8.33 Firewall placement between the administered network andthe outside worldAdministerednetworkFirewallPublicInternet IP source or destination address Protocol type in IP datagram field: TCP, UDP, ICMP, OSPF, and so on TCP or UDP source and destination port TCP flag bits: SYN, ACK, and so on ICMP message type Different rules for datagrams leaving and entering the network Different rules for the different router interfacesA network administrator configures the firewall based on the policy of the organ-ization. The policy may take user productivity and bandwidth usage into account aswell as the security concerns of an organization. Table 8.5 lists a number of possiblepolices an organization may have, and how they would be addressed with a packetfilter. For example, if the organization doesnt want any incoming TCP connectionsexcept those for its public Web server, it can block all incoming TCP SYN segmentsexcept TCP SYN segments with destination port 80 and the destination IP addresscorresponding to the Web server. If the organization doesnt want its users to monop-olize access bandwidth with Internet radio applications, it can block all not-criticalUDP traffic (since Internet radio is often sent over UDP). If the organization doesntwant its internal network to be mapped (tracerouted) by an outsider, it can block allICMP TTL expired messages leaving the organizations network.A filtering policy can be based on a combination of addresses and port numbers.For example, a filtering router could forward all Telnet datagrams (those with a port number of 23) except those going to and coming from a list of specific IPaddresses. This policy permits Telnet connections to and from hosts on the allowed8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 733Policy Firewall SettingNo outside Web access. Drop all outgoing packets to any IP address, port 80Table 8.5 Policies and corresponding filtering rules for an organizationsnetwork 130.27/16 with Web server at 130.207.244.203No incoming TCP connections, except those for organizations public Web server only.Drop all incoming TCP SYN packets to any IP except130.207.244.203, port 80Prevent Web-radios from eating up the available bandwidth.Drop all incoming UDP packetsexcept DNS packets.Prevent your network from being used for a smurf DoS attack.Drop all ICMP ping packets going to a broadcastaddress (eg 130.207.255.255).Prevent your network from being tracerouted Drop all outgoing ICMP TTL expired trafficlist. Unfortunately, basing the policy on external addresses provides no protectionagainst datagrams that have had their source addresses spoofed.Filtering can also be based on whether or not the TCP ACK bit is set. This trickis quite useful if an organization wants to let its internal clients connect to externalservers but wants to prevent external clients from connecting to internal servers.Recall from Section 3.5 that the first segment in every TCP connection has the ACKbit set to 0, whereas all the other segments in the connection have the ACK bit set to1. Thus, if an organization wants to prevent external clients from initiating connec-tions to internal servers, it simply filters all incoming segments with the ACK bit setto 0. This policy kills all TCP connections originating from the outside, but permitsconnections originating internally.Firewall rules are implemented in routers with access control lists, with eachrouter interface having its own list. An example of an access control list for anorganization 222.22/16 is shown in Table 8.6. This access control list is for aninterface that connects the router to the organizations external ISPs. Rules areapplied to each datagram that passes through the interface from top to bottom. Thefirst two rules together allow internal users to surf the Web: The first rule allows anyTCP packet with destination port 80 to leave the organizations network; the sec-ond rule allows any TCP packet with source port 80 and the ACK bit set to enterthe organizations network. Note that if an external source attempts to establish aTCP connection with an internal host, the connection will be blocked, even if thesource or destination port is 80. The second two rules together allow DNS packetsto enter and leave the organizations network. In summary, this rather restrictiveaccess control list blocks all traffic except Web traffic initiated from within theorganization and DNS traffic. [CERT Filtering 2012] provides a list of recom-mended port/protocol packet filterings to avoid a number of well-known securityholes in existing network applications.734 CHAPTER 8 SECURITY IN COMPUTER NETWORKSaction source address dest address protocol source port dest port flag bitTable 8.6 An access control list for a router interfaceallow 222.22/16 outside of222.22/16TCP > 1023 80 any
allow outside of
222.22/16
222.22/16 TCP 80 > 1023 ACK
allow 222.22/16 outside of
222.22/16
UDP > 1023 53
allow outside of
222.22/16
222.22/16 UDP 53 > 1023
deny all all all all all all
Stateful Packet Filters
In a traditional packet filter, filtering decisions are made on each packet in isolation.
Stateful filters actually track TCP connections, and use this knowledge to make fil-
tering decisions.
To understand stateful filters, lets reexamine the access control list in Table 8.6.
Although rather restrictive, the access control list in Table 8.6 nevertheless allows
any packet arriving from the outside with ACK = 1 and source port 80 to get through
the filter. Such packets could be used by attackers in attempts to crash internal sys-
tems with malformed packets, carry out denial-of-service attacks, or map the inter-
nal network. The naive solution is to block TCP ACK packets as well, but such an
approach would prevent the organizations internal users from surfing the Web.
Stateful filters solve this problem by tracking all ongoing TCP connections in a
connection table. This is possible because the firewall can observe the beginning of
a new connection by observing a three-way handshake (SYN, SYNACK, and
ACK); and it can observe the end of a connection when it sees a FIN packet for the
connection. The firewall can also (conservatively) assume that the connection is
over when it hasnt seen any activity over the connection for, say, 60 seconds. An
example connection table for a firewall is shown in Table 8.7. This connection table
indicates that there are currently three ongoing TCP connections, all of which have
been initiated from within the organization. Additionally, the stateful filter includes
a new column, check connection, in its access control list, as shown in Table 8.8.
Note that Table 8.8 is identical to the access control list in Table 8.6, except now it
indicates that the connection should be checked for two of the rules.
Lets walk through some examples to see how the connection table and the
extended access control list work hand-in-hand. Suppose an attacker attempts to
send a malformed packet into the organizations network by sending a datagram
with TCP source port 80 and with the ACK flag set. Further suppose that this packet
has source port number 12543 and source IP address 150.23.23.155. When this
packet reaches the firewall, the firewall checks the access control list in Table 8.7,
which indicates that the connection table must also be checked before permitting
this packet to enter the organizations network. The firewall duly checks the connec-
tion table, sees that this packet is not part of an ongoing TCP connection, and rejects
8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 735
source address dest address source port dest port
222.22.1.7 37.96.87.123 12699 80
222.22.93.2 199.1.205.23 37654 80
222.22.65.143 203.77.240.43 48712 80
Table 8.7 Connection table for stateful filter
the packet. As a second example, suppose that an internal user wants to surf an
external Web site. Because this user first sends a TCP SYN segment, the users TCP
connection gets recorded in the connection table. When the Web server sends back
packets (with the ACK bit necessarily set), the firewall checks the table and sees that
a corresponding connection is in progress. The firewall will thus let these packets
pass, thereby not interfering with the internal users Web surfing activity.
Application Gateway
In the examples above, we have seen that packet-level filtering allows an organiza-
tion to perform coarse-grain filtering on the basis of the contents of IP and
TCP/UDP headers, including IP addresses, port numbers, and acknowledgment bits.
But what if an organization wants to provide a Telnet service to a restricted set of
internal users (as opposed to IP addresses)? And what if the organization wants such
privileged users to authenticate themselves first before being allowed to create Telnet
sessions to the outside world? Such tasks are beyond the capabilities of traditional
and stateful filters. Indeed, information about the identity of the internal users is
application-layer data and is not included in the IP/TCP/UDP headers.
To have finer-level security, firewalls must combine packet filters with applica-
tion gateways. Application gateways look beyond the IP/TCP/UDP headers and
make policy decisions based on application data. An application gateway is an
application-specific server through which all application data (inbound and out-
bound) must pass. Multiple application gateways can run on the same host, but each
gateway is a separate server with its own processes.
736 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
Table 8.8 Access control list for stateful filter
action source
address
dest
address
protocol source port dest port flag bit
allow 222.22/16 outside of
222.22/16
TCP >1023 80 any
allow outside of
222.22/16
222.22/16 TCP 80 >1023 ACK
allow 222.22/16 outside of
222.22/16
UDP >1023 53
allow outside of
222.22/16
222.22/16 UDP 53 >1023
deny all all all all all all
check
conxion
X
X
To get some insight into application gateways, lets design a firewall that allows
only a restricted set of internal users to Telnet outside and prevents all external
clients from Telneting inside. Such a policy can be accomplished by implementing a
combination of a packet filter (in a router) and a Telnet application gateway, as
shown in Figure 8.34. The routers filter is configured to block all Telnet connec-
tions except those that originate from the IP address of the application gateway.
Such a filter configuration forces all outbound Telnet connections to pass through
the application gateway. Consider now an internal user who wants to Telnet to the
outside world. The user must first set up a Telnet session with the application gate-
way. An application running in the gateway, which listens for incoming Telnet ses-
sions, prompts the user for a user ID and password. When the user supplies this
information, the application gateway checks to see if the user has permission to
Telnet to the outside world. If not, the Telnet connection from the internal user to the
gateway is terminated by the gateway. If the user has permission, then the gateway
(1) prompts the user for the host name of the external host to which the user wants
to connect, (2) sets up a Telnet session between the gateway and the external host,
and (3) relays to the external host all data arriving from the user, and relays to the
user all data arriving from the external host. Thus, the Telnet application gateway
not only performs user authorization but also acts as a Telnet server and a Telnet
client, relaying information between the user and the remote Telnet server. Note that
8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 737
Figure 8.34 Firewall consisting of an application gateway and a filter
Application
gateway
Host-to-gateway
Telnet session
Gateway-to-remote
host Telnet session
Router
and filter
the filter will permit step 2 because the gateway initiates the Telnet connection to
the outside world.
738 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
ANONYMITY AND PRIVACY
Suppose you want to visit a controversial Web site (for example, a political activist
site) and you (1) dont want to reveal your IP address to the Web site, (2) dont want
your local ISP (which may be your home or office ISP) to know that you are visiting
the site, and (3) you dont want your local ISP to see the data you are exchanging
with the site. If you use the traditional approach of connecting directly to the Web
site without any encryption, you fail on all three counts. Even if you use SSL, you fail
on the first two counts: Your source IP address is presented to the Web site in every
datagram you send; and the destination address of every packet you send can easily
be sniffed by your local ISP.
To obtain privacy and anonymity, you can instead use a combination of a trusted
proxy server and SSL, as shown in Figure 8.35. With this approach, you first make
an SSL connection to the trusted proxy. You then send, into this SSL connection, an
HTTP request for a page at the desired site. When the proxy receives the SSL-encrypt-
ed HTTP request, it decrypts the request and forwards the cleartext HTTP request to
the Web site. The Web site then responds to the proxy, which in turn forwards the
response to you over SSL. Because the Web site only sees the IP address of the
proxy, and not of your clients address, you are indeed obtaining anonymous access
to the Web site. And because all traffic between you and the proxy is encrypted,
your local ISP cannot invade your privacy by logging the site you visited or recording
the data you are exchanging. Many companies today (such as proxify.com) make
available such proxy services.
Of course, in this solution, your proxy knows everything: It knows your IP address
and the IP address of the site youre surfing; and it can see all the traffic in cleartext
exchanged between you and the Web site. Such a solution, therefore, is only as
good as the trustworthiness of the proxy. A more robust approach, taken by the TOR
anonymizing and privacy service, is to route your traffic through a series of non-
colluding proxy servers [TOR 2012]. In particular, TOR allows independent individu-
als to contribute proxies to its proxy pool. When a user connects to a server using
TOR, TOR randomly chooses (from its proxy pool) a chain of three proxies and
routes all traffic between client and server over the chain. In this manner, assuming
the proxies do not collude, no one knows that communication took place between
your IP address and the target Web site. Furthermore, although cleartext is sent
between the last proxy and the server, the last proxy doesnt know what IP address is
sending and receiving the cleartext.
CASE HISTORY
Internal networks often have multiple application gateways, for example, gate-
ways for Telnet, HTTP, FTP, and e-mail. In fact, an organizations mail server (see
Section 2.4) and Web cache are application gateways.
Application gateways do not come without their disadvantages. First, a differ-
ent application gateway is needed for each application. Second, there is a perform-
ance penalty to be paid, since all data will be relayed via the gateway. This becomes
a concern particularly when multiple users or applications are using the same gate-
way machine. Finally, the client software must know how to contact the gateway
when the user makes a request, and must know how to tell the application gateway
what external server to connect to.
8.9.2 Intrusion Detection Systems
Weve just seen that a packet filter (traditional and stateful) inspects IP, TCP, UDP,
and ICMP header fields when deciding which packets to let pass through the firewall.
However, to detect many attack types, we need to perform deep packet inspection,
that is, look beyond the header fields and into the actual application data that the
packets carry. As we saw in Section 8.9.1, application gateways often do deep packet
inspection. But an application gateway only does this for a specific application.
Clearly, there is a niche for yet another devicea device that not only examines
the headers of all packets passing through it (like a packet filter), but also performs
deep packet inspection (unlike a packet filter). When such a device observes a
suspicious packet, or a suspicious series of packets, it could prevent those packets
from entering the organizational network. Or, because the activity is only deemed as
suspicious, the device could let the packets pass, but send alerts to a network
administrator, who can then take a closer look at the traffic and take appropriate
8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 739
Figure 8.35 Providing anonymity and privacy with a proxy
Alice
Anonymizing
Proxy
SSL
Cle
art
ex
t
actions. A device that generates alerts when it observes potentially malicious traffic
is called an intrusion detection system (IDS). A device that filters out suspicious
traffic is called an intrusion prevention system (IPS). In this section we study both
systemsIDS and IPStogether, since the most interesting technical aspect of
these systems is how they detect suspicious traffic (and not whether they send alerts or
drop packets). We will henceforth collectively refer to IDS systems and IPS systems
as IDS systems.
An IDS can be used to detect a wide range of attacks, including network map-
ping (emanating, for example, from nmap), port scans, TCP stack scans, DoS band-
width-flooding attacks, worms and viruses, OS vulnerability attacks, and application
vulnerability attacks. (See Section 1.6 for a survey of network attacks.) Today, thou-
sands of organizations employ IDS systems. Many of these deployed systems are
proprietary, marketed by Cisco, Check Point, and other security equipment vendors.
But many of the deployed IDS systems are public-domain systems, such as the
immensely popular Snort IDS system (which well discuss shortly).
740 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
Figure 8.36 An organization deploying a filter, an application gateway,
and IDS sensors
Internet
Web
server
FTP
server
DNS
server
Internal
network
Application
gateway
Demilitarized zone
Filter
Key:
= IDS sensors
An organization may deploy one or more IDS sensors in its organizational net-
work. Figure 8.36 shows an organization that has three IDS sensors. When multiple
sensors are deployed, they typically work in concert, sending information about sus-
picious traffic activity to a central IDS processor, which collects and integrates the
information and sends alarms to network administrators when deemed appropriate.
In Figure 8.36, the organization has partitioned its network into two regions: a high-
security region, protected by a packet filter and an application gateway and moni-
tored by IDS sensors; and a lower-security regionreferred to as the demilitarized
zone (DMZ)which is protected only by the packet filter, but also monitored by
IDS sensors. Note that the DMZ includes the organizations servers that need to com-
municate with the outside world, such as its public Web server and its authoritative
DNS server.
You may be wondering at this stage, why multiple IDS sensors? Why not just
place one IDS sensor just behind the packet filter (or even integrated with the packet
filter) in Figure 8.36? We will soon see that an IDS not only needs to do deep packet
inspection, but must also compare each passing packet with tens of thousands of
signatures; this can be a significant amount of processing, particularly if the
organization receives gigabits/sec of traffic from the Internet. By placing the IDS
sensors further downstream, each sensor sees only a fraction of the organizations
traffic, and can more easily keep up. Nevertheless, high-performance IDS and IPS
systems are available today, and many organizations can actually get by with just
one sensor located near its access router.
IDS systems are broadly classified as either signature-based systems or
anomaly-based systems. A signature-based IDS maintains an extensive database
of attack signatures. Each signature is a set of rules pertaining to an intrusion activ-
ity. A signature may simply be a list of characteristics about a single packet (e.g.,
source and destination port numbers, protocol type, and a specific string of bits in
the packet payload), or may relate to a series of packets. The signatures are nor-
mally created by skilled network security engineers who research known attacks.
An organizations network administrator can customize the signatures or add its
own to the database.
Operationally, a signature-based IDS sniffs every packet passing by it, compar-
ing each sniffed packet with the signatures in its database. If a packet (or series of
packets) matches a signature in the database, the IDS generates an alert. The alert
could be sent to the network administrator in an e-mail message, could be sent to the
network management system, or could simply be logged for future inspection.
Signature-based IDS systems, although widely deployed, have a number of limi-
tations. Most importantly, they require previous knowledge of the attack to generate
an accurate signature. In other words, a signature-based IDS is completely blind to
new attacks that have yet to be recorded. Another disadvantage is that even if a sig-
nature is matched, it may not be the result of an attack, so that a false alarm is gener-
ated. Finally, because every packet must be compared with an extensive collection of
signatures, the IDS can become overwhelmed with processing and actually fail to
detect many malicious packets.
8.9 OPERATIONAL SECURITY: FIREWALLS AND INTRUSION DETECTION SYSTEMS 741
An anomaly-based IDS creates a traffic profile as it observes traffic in normal
operation. It then looks for packet streams that are statistically unusual, for exam-
ple, an inordinate percentage of ICMP packets or a sudden exponential growth in
port scans and ping sweeps. The great thing about anomaly-based IDS systems is
that they dont rely on previous knowledge about existing attacksthat is, they can
potentially detect new, undocumented attacks. On the other hand, it is an extremely
challenging problem to distinguish between normal traffic and statistically unusual
traffic. To date, most IDS deployments are primarily signature-based, although
some include some anomaly-based features.
Snort
Snort is a public-domain, open source IDS with hundreds of thousands of existing
deployments [Snort 2012; Koziol 2003]. It can run on Linux, UNIX, and Windows
platforms. It uses the generic sniffing interface libpcap, which is also used by
Wireshark and many other packet sniffers. It can easily handle 100 Mbps of traffic;
for installations with gibabit/sec traffic rates, multiple Snort sensors may be needed.
To gain some insight into Snort, lets take a look at an example of a Snort signature:
alert icmp $EXTERNAL_NET any -> $HOME_NET any
(msg:ICMP PING NMAP; dsize: 0; itype: 8;)
This signature is matched by any ICMP packet that enters the organizations net-
work ($HOME_NET) from the outside ($EXTERNAL_NET), is of type 8 (ICMP
ping), and has an empty payload (dsize = 0). Since nmap (see Section 1.6) generates
ping packets with these specific characteristics, this signature is designed to detect
nmap ping sweeps. When a packet matches this signature, Snort generates an alert
that includes the message ICMP PING NMAP.
Perhaps what is most impressive about Snort is the vast community of users and
security experts that maintain its signature database. Typically within a few hours of
a new attack, the Snort community writes and releases an attack signature, which is
then downloaded by the hundreds of thousands of Snort deployments distributed
around the world. Moreover, using the Snort signature syntax, network administra-
tors can tailor the signatures to their own organizations needs by either modifying
existing signatures or creating entirely new ones.
8.10 Summary
In this chapter, weve examined the various mechanisms that our secret lovers, Bob
and Alice, can use to communicate securely. Weve seen that Bob and Alice are
interested in confidentiality (so they alone are able to understand the contents of a
transmitted message), end-point authentication (so they are sure that they are talking
742 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
with each other), and message integrity (so they are sure that their messages are
not altered in transit). Of course, the need for secure communication is not confined
to secret lovers. Indeed, we saw in Sections 8.5 through 8.8 that security can be used
in various layers in a network architecture to protect against bad guys who have a
large arsenal of possible attacks at hand.
The first part of this chapter presented various principles underlying secure
communication. In Section 8.2, we covered cryptographic techniques for encrypting
and decrypting data, including symmetric key cryptography and public key cryptog-
raphy. DES and RSA were examined as specific case studies of these two major
classes of cryptographic techniques in use in todays networks.
In Section 8.3, we examined two approaches for providing message integrity:
message authentication codes (MACs) and digital signatures. The two approaches
have a number of parallels. Both use cryptographic hash functions and both tech-
niques enable us to verify the source of the message as well as the integrity of the
message itself. One important difference is that MACs do not rely on encryption
whereas digital signatures require a public key infrastructure. Both techniques are
extensively used in practice, as we saw in Sections 8.5 through 8.8. Furthermore, dig-
ital signatures are used to create digital certificates, which are important for verifying
the validity of public keys. In Section 8.4, we examined endpoint authentication and
introduced nonces to defend against the replay attack.
In Sections 8.5 through 8.8 we examined several security networking proto-
cols that enjoy extensive use in practice. We saw that symmetric key cryptogra-
phy is at the core of PGP, SSL, IPsec, and wireless security. We saw that public
key cryptography is crucial for both PGP and SSL. We saw that PGP uses digital
signatures for message integrity, whereas SSL and IPsec use MACs. Having now
an understanding of the basic principles of cryptography, and having studied how
these principles are actually used, you are now in position to design your own
secure network protocols!
Armed with the techniques covered in Sections 8.2 through 8.8, Bob and Alice
can communicate securely. (One can only hope that they are networking students
who have learned this material and can thus avoid having their tryst uncovered by
Trudy!) But confidentiality is only a small part of the network security picture.
As we learned in Section 8.9, increasingly, the focus in network security has been
on securing the network infrastructure against a potential onslaught by the bad guys.
In the latter part of this chapter, we thus covered firewalls and IDS systems which
inspect packets entering and leaving an organizations network.
This chapter has covered a lot of ground, while focusing on the most impor-
tant topics in modern network security. Readers who desire to dig deeper are
encouraged to investigate the references cited in this chapter. In particular, we rec-
ommend [Skoudis 2006] for attacks and operational security, [Kaufman 1995] for
cryptography and how it applies to network security, [Rescorla 2001] for an in-
depth but readable treatment of SSL, and [Edney 2003] for a thorough discussion
of 802.11 security, including an insightful investigation into WEP and its flaws.
8.10 SUMMARY 743
Homework Problems and Questions
Chapter 8 Review Problems
SECTION 8.1
R1. What are the differences between message confidentiality and message
integrity? Can you have confidentiality without integrity? Can you have
integrity without confidentiality? Justify your answer.
R2. Internet entities (routers, switches, DNS servers, Web servers, user end sys-
tems, and so on) often need to communicate securely. Give three specific
example pairs of Internet entities that may want secure communication.
SECTION 8.2
R3. From a service perspective, what is an important difference between a
symmetric-key system and a public-key system?
R4. Suppose that an intruder has an encrypted message as well as the decrypted
version of that message. Can the intruder mount a ciphertext-only attack, a
known-plaintext attack, or a chosen-plaintext attack?
R5. Consider an 8-block cipher. How many possible input blocks does this cipher
have? How many possible mappings are there? If we view each mapping as a
key, then how many possible keys does this cipher have?
R6. Suppose N people want to communicate with each of N 1 other people
using symmetric key encryption. All communication between any two peo-
ple, i and j, is visible to all other people in this group of N, and no other per-
son in this group should be able to decode their communication. How many
keys are required in the system as a whole? Now suppose that public key
encryption is used. How many keys are required in this case?
R7. Suppose n = 10,000, a = 10,023, and b = 10,004. Use an identity of modular
arithmetic to calculate in your head (a b) mod n.
R8. Suppose you want to encrypt the message 10101111 by encrypting the deci-
mal number that corresponds to the message. What is the decimal number?
SECTIONS 8.38.4
R9. In what way does a hash provide a better message integrity check than a
checksum (such as the Internet checksum)?
R10. Can you decrypt a hash of a message to get the original message? Explain
your answer.
R11. Consider a variation of the MAC algorithm (Figure 8.9) where the sender
sends (m, H(m) + s), where H(m) + s is the concatenation of H(m) and s. Is
this variation flawed? Why or why not?
744 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
R12. What does it mean for a signed document to be verifiable and non-forgeable?
R13. In what way does the public-key encrypted message hash provide a better
digital signature than the public-key encrypted message?
R14. Suppose certifier.com creates a certificate for foo.com. Typically, the entire
certificate would be encrypted with certifier.coms public key. True or False?
R15. Suppose Alice has a message that she is ready to send to anyone who asks.
Thousands of people want to obtain Alices message, but each wants to be sure
of the integrity of the message. In this context, do you think a MAC-based or a
digital-signature-based integrity scheme is more suitable? Why?
R16. What is the purpose of a nonce in an end-point authentication protocol?
R17. What does it mean to say that a nonce is a once-in-a-lifetime value? In whose
lifetime?
R18. Is the message integrity scheme based on HMAC susceptible to playback
attacks? If so, how can a nonce be incorporated into the scheme to remove
this susceptibility?
SECTIONS 8.58.8
R19. Suppose that Bob receives a PGP message from Alice. How does Bob know
for sure that Alice created the message (rather than, say, Trudy)? Does PGP
use a MAC for message integrity?
R20. In the SSL record, there is a field for SSL sequence numbers. True or False?
R21. What is the purpose of the random nonces in the SSL handshake?
R22. Suppose an SSL session employs a block cipher with CBC. True or False:
The server sends to the client the IV in the clear?
R23. Suppose Bob initiates a TCP connection to Trudy who is pretending to be
Alice. During the handshake, Trudy sends Bob Alices certificate. In what
step of the SSL handshake algorithm will Bob discover that he is not commu-
nicating with Alice?
R24. Consider sending a stream of packets from Host A to Host B using IPsec.
Typically, a new SA will be established for each packet sent in the stream.
True or False?
R25. Suppose that TCP is being run over IPsec between headquarters and the
branch office in Figure 8.28. If TCP retransmits the same packet, then the two
corresponding packets sent by R1 packets will have the same sequence num-
ber in the ESP header. True or False?
R26. An IKE SA and an IPsec SA are the same thing. True or False?
R27. Consider WEP for 802.11. Suppose that the data is 10101100 and the
keystream is 1111000. What is the resulting ciphertext?
R28. In WEP, an IV is sent in the clear in every frame. True or False?
HOMEWORK PROBLEMS AND QUESTIONS 745
SECTION 8.9
R29. Stateful packet filters maintain two data structures. Name them and briefly
describe what they do.
R30. Consider a traditional (stateless) packet filter. This packet filter may filter
packets based on TCP flag bits as well as other header fields. True or False?
R31. In a traditional packet filter, each interface can have its own access control
list. True or False?
R32. Why must an application gateway work in conjunction with a router filter to
be effective?
R33. Signature-based IDSs and IPSs inspect into the payloads of TCP and UDP
segments. True or False?
Problems
P1. Using the monoalphabetic cipher in Figure 8.3, encode the message This is
an easy problem. Decode the message rmiju uamu xyj.
P2. Show that Trudys known-plaintext attack, in which she knows the
(ciphertext, plaintext) translation pairs for seven letters, reduces the num-
ber of possible substitutions to be checked in the example in Section 8.2.1
by approximately 109.
P3. Consider the polyalphabetic system shown in Figure 8.4. Will a chosen-
plaintext attack that is able to get the plaintext encoding of the message The
quick brown fox jumps over the lazy dog. be sufficient to decode all mes-
sages? Why or why not?
P4. Consider the block cipher in Figure 8.5. Suppose that each block cipher Ti
simply reverses the order of the eight input bits (so that, for example,
11110000 becomes 00001111). Further suppose that the 64-bit scrambler
does not modify any bits (so that the output value of the mth bit is equal to
the input value of the mth bit). (a) With n = 3 and the original 64-bit input
equal to 10100000 repeated eight times, what is the value of the output?
(b) Repeat part (a) but now change the last bit of the original 64-bit input
from a 0 to a 1. (c) Repeat parts (a) and (b) but now suppose that the 64-bit
scrambler inverses the order of the 64 bits.
P5. Consider the block cipher in Figure 8.5. For a given key Alice and Bob
would need to keep eight tables, each 8 bits by 8 bits. For Alice (or Bob)
to store all eight tables, how many bits of storage are necessary? How does
this number compare with the number of bits required for a full-table 64-
bit block cipher?
P6. Consider the 3-bit block cipher in Table 8.1. Suppose the plaintext is
100100100. (a) Initially assume that CBC is not used. What is the resulting
746 CHAPTER 8 SECURITY IN COMPUTER NETWORKS
ciphertext? (b) Suppose Trudy sniffs the ciphertext. Assuming she knows that
a 3-bit block cipher without CBC is being employed (but doesnt know the
specific cipher), what can she surmise? (c) Now suppose that CBC is used
with IV = 111. What is the resulting ciphertext?
P7. (a) Using RSA, choose p = 3 and q = 11, and encode the word dog by
encrypting each letter separately. Apply the decryption algorithm to the
encrypted version to recover the original plaintext message. (b) Repeat
part (a) but now encrypt dog as one message m.
P8. Consider RSA with p = 5 and q = 11.
a. What are n and z?
b. Let e be 3. Why is this an acceptable choice for e?
c. Find d such that de = 1 (mod z) and d < 160.d. Encrypt the message m = 8 using the key (n, e). Let c denote the corre-sponding ciphertext. Show all work. Hint: To simplify the calculations,use the fact:[(a mod n) (b mod n)] mod n = (a b) mod nP9. In this problem, we explore the Diffie-Hellman (DH) public-key encryptionalgorithm, which allows two entities to agree on a shared key. The DH algorithm makes use of a large prime number p and another large numberg less than p. Both p and g are made public (so that an attacker would knowthem). In DH, Alice and Bob each independently choose secret keys, SA andSB, respectively. Alice then computes her public key, TA, by raising g to SA andthen taking mod p. Bob similarly computes his own public key TB by raising gto SB and then taking mod p. Alice and Bob then exchange their public keysover the Internet. Alice then calculates the shared secret key S by raising TB toSA and then taking mod p. Similarly, Bob calculates the shared key S by rais-ing TA to SB and then taking mod p.a. Prove that, in general, Alice and Bob obtain the same symmetric key, thatis, prove S = S.b. With p = 11 and g = 2, suppose Alice and Bob choose private keys SA = 5and SB = 12, respectively. Calculate Alices and Bobs public keys, TA andTB . Show all work.c. Following up on part (b), now calculate S as the shared symmetric key. Showall work.d. Provide a timing diagram that shows how Diffie-Hellman can be attacked bya man-in-the-middle. The timing diagram should have three vertical lines, onefor Alice, one for Bob, and one for the attacker Trudy.P10. Suppose Alice wants to communicate with Bob using symmetric key cryp-tography using a session key KS. In Section 8.2, we learned how public-keyPROBLEMS 747cryptography can be used to distribute the session key from Alice to Bob. Inthis problem, we explore how the session key can be distributedwithoutpublic key cryptographyusing a key distribution center (KDC). The KDCis a server that shares a unique secret symmetric key with each registereduser. For Alice and Bob, denote these keys by KA-KDC and KB-KDC. Design ascheme that uses the KDC to distribute KS to Alice and Bob. Your schemeshould use three messages to distribute the session key: a message from Aliceto the KDC; a message from the KDC to Alice; and finally a message fromAlice to Bob. The first message is KA-KDC (A, B). Using the notation, KA-KDC,KB-KDC, S, A, and B answer the following questions.a. What is the second message?b. What is the third message?P11. Compute a third message, different from the two messages in Figure 8.8, thathas the same checksum as the messages in Figure 8.8.P12. Suppose Alice and Bob share two secret keys: an authentication key S1 and asymmetric encryption key S2. Augment Figure 8.9 so that both integrity andconfidentiality are provided.P13. In the BitTorrent P2P file distribution protocol (see Chapter 2), the seedbreaks the file into blocks, and the peers redistribute the blocks to each other.Without any protection, an attacker can easily wreak havoc in a torrent bymasquerading as a benevolent peer and sending bogus blocks to a small sub-set of peers in the torrent. These unsuspecting peers then redistribute thebogus blocks to other peers, which in turn redistribute the bogus blocks toeven more peers. Thus, it is critical for BitTorrent to have a mechanism thatallows a peer to verify the integrity of a block, so that it doesnt redistributebogus blocks. Assume that when a peer joins a torrent, it initially gets a.torrent file from a fully trusted source. Describe a simple scheme thatallows peers to verify the integrity of blocks.P14. The OSPF routing protocol uses a MAC rather than digital signatures toprovide message integrity. Why do you think a MAC was chosen over dig-ital signatures?P15. Consider our authentication protocol in Figure 8.18 in which Alice authenti-cates herself to Bob, which we saw works well (i.e., we found no flaws in it).Now suppose that while Alice is authenticating herself to Bob, Bob mustauthenticate himself to Alice. Give a scenario by which Trudy, pretending to beAlice, can now authenticate herself to Bob as Alice. (Hint: Consider that thesequence of operations of the protocol, one with Trudy initiating and one withBob initiating, can be arbitrarily interleaved. Pay particular attention to the factthat both Bob and Alice will use a nonce, and that if care is not taken, the samenonce can be used maliciously.)P16. A natural question is whether we can use a nonce and public key cryptogra-phy to solve the end-point authentication problem in Section 8.4. Consider748 CHAPTER 8 SECURITY IN COMPUTER NETWORKSthe following natural protocol: (1)Alice sends the message I amAlice to Bob. (2) Bob chooses a nonce, R, and sends it to Alice. (3) Aliceuses her private key to encrypt the nonce and sends the resulting value toBob. (4) Bob applies Alice’s public key to the received message. Thus, Bobcomputes R and authenticates Alice. a. Diagram this protocol, using the notation for public and private keysemployed in the textbook.b. Suppose that certificates are not used. Describe how Trudy can become awoman-in-the-middle by intercepting Alices messages and then pre-tending to be Alice to Bob.P17. Figure 8.19 shows the operations that Alice must perform with PGP to pro-vide confidentiality, authentication, and integrity. Diagram the correspondingoperations that Bob must perform on the package received from Alice.P18. Suppose Alice wants to send an e-mail to Bob. Bob has a public-private keypair (KB+,KB), and Alice has Bobs certificate. But Alice does not have apublic, private key pair. Alice and Bob (and the entire world) share the samehash function H().a. In this situation, is it possible to design a scheme so that Bob can verifythat Alice created the message? If so, show how with a block diagram forAlice and Bob.b. Is it possible to design a scheme that provides confidentiality for sendingthe message from Alice to Bob? If so, show how with a block diagram forAlice and Bob.P19. Consider the Wireshark output below for a portion of an SSL session.a. Is Wireshark packet 112 sent by the client or server?b. What is the servers IP address and port number?c. Assuming no loss and no retransmissions, what will be the sequence num-ber of the next TCP segment sent by the client?d. How many SSL records does Wireshark packet 112 contain?e. Does packet 112 contain a Master Secret or an Encrypted Master Secret orneither?f. Assuming that the handshake type field is 1 byte and each length field is3 bytes, what are the values of the first and last bytes of the Master Secret(or Encrypted Master Secret)?g. The client encrypted handshake message takes into account how manySSL records?h. The server encrypted handshake message takes into account how manySSL records?P20. In Section 8.6.1, it is shown that without sequence numbers, Trudy (a woman-in-the middle) can wreak havoc in an SSL session by interchanging TCPPROBLEMS 749segments. Can Trudy do something similar by deleting a TCP segment? Whatdoes she need to do to succeed at the deletion attack? What effect will it have?P21. Suppose Alice and Bob are communicating over an SSL session. Suppose anattacker, who does not have any of the shared keys, inserts a bogus TCP segmentinto a packet stream with correct TCP checksum and sequence numbers (and cor-rect IP addresses and port numbers). Will SSL at the receiving side accept thebogus packet and pass the payload to the receiving application? Why or why not?P22. The following True/False questions pertain to Figure 8.28.a. When a host in 172.16.1/24 sends a datagram to an Amazon.com server,the router R1 will encrypt the datagram using IPsec.b. When a host in 172.16.1/24 sends a datagram to a host in 172.16.2/24, therouter R1 will change the source and destination address of the IP datagram.c. Suppose a host in 172.16.1/24 initiates a TCP connection to a Web serverin 172.16.2/24. As part of this connection, all datagrams sent by R1 willhave protocol number 50 in the left-most IPv4 header field.750 CHAPTER 8 SECURITY IN COMPUTER NETWORKS(Wireshark screenshot reprinted by permission of the Wireshark Foundation.)d. Consider sending a TCP segment from a host in 172.16.1/24 to a host in172.16.2/24. Suppose the acknowledgment for this segment gets lost, sothat TCP resends the segment. Because IPsec uses sequence numbers, R1will not resend the TCP segment.P23. Consider the example in Figure 8.28. Suppose Trudy is a woman-in-the-middle, who can insert datagrams into the stream of datagrams going fromR1 and R2. As part of a replay attack, Trudy sends a duplicate copy of one ofthe datagrams sent from R1 to R2. Will R2 decrypt the duplicate datagramand forward it into the branch-office network? If not, describe in detail howR2 detects the duplicate datagram.P24. Consider the following pseudo-WEP protocol. The key is 4 bits and the IVis 2 bits. The IV is appended to the end of the key when generating thekeystream. Suppose that the shared secret key is 1010. The keystreams forthe four possible inputs are as follows:101000: 0010101101010101001011010100100 . . .101001: 1010011011001010110100100101101 . . .101010: 0001101000111100010100101001111 . . .101011: 1111101010000000101010100010111 . . .Suppose all messages are 8-bits long. Suppose the ICV (integrity check) is 4-bits long, and is calculated by XOR-ing the first 4 bits of data with the last4 bits of data. Suppose the pseudo-WEP packet consists of three fields: firstthe IV field, then the message field, and last the ICV field, with some of these fields encrypted.a. We want to send the message m = 10100000 using the IV = 11 and usingWEP. What will be the values in the three WEP fields?b. Show that when the receiver decrypts the WEP packet, it recovers themessage and the ICV.c. Suppose Trudy intercepts a WEP packet (not necessarily with the IV = 11)and wants to modify it before forwarding it to the receiver. Suppose Trudyflips the first ICV bit. Assuming that Trudy does not know the keystreamsfor any of the IVs, what other bit(s) must Trudy also flip so that thereceived packet passes the ICV check?d. Justify your answer by modifying the bits in the WEP packet in part (a),decrypting the resulting packet, and verifying the integrity check.P25. Provide a filter table and a connection table for a stateful firewall that is asrestrictive as possible but accomplishes the following:a. Allows all internal users to establish Telnet sessions with external hosts.b. Allows external users to surf the company Web site at 222.22.0.12.c. But otherwise blocks all inbound and outbound traffic.PROBLEMS 751The internal network is 222.22/16. In your solution, suppose that the connec-tion table is currently caching three connections, all from inside to outside.Youll need to invent appropriate IP addresses and port numbers.P26. Suppose Alice wants to visit the Web site activist.com using a TOR-like service. This service uses two non-colluding proxy servers, Proxy1 andProxy2. Alice first obtains the certificates (each containing a public key) forProxy1 and Proxy2 from some central server. Denote K1+( ), K2+( ), K1( ), andK2( ) for the encryption/decryption with public and private RSA keys.a. Using a timing diagram, provide a protocol (as simple as possible) thatenables Alice to establish a shared session key S1 with Proxy1. DenoteS1(m) for encryption/decryption of data m with the shared key S1.b. Using a timing diagram, provide a protocol (as simple as possible) thatallows Alice to establish a shared session key S2 with Proxy2 withoutrevealing her IP address to Proxy2.c. Assume now that shared keys S1 and S2 are now established. Using a tim-ing diagram, provide a protocol (as simple as possible and not using public-key cryptography) that allows Alice to request an html page fromactivist.com without revealing her IP address to Proxy2 and withoutrevealing to Proxy1 which site she is visiting. Your diagram should endwith an HTTP request arriving at activist.com.Wireshark LabIn this lab (available from the companion Web site), we investigate the Secure SocketsLayer (SSL) protocol. Recall from Section 8.6 that SSL is used for securing a TCPconnection, and that it is extensively used in practice for secure Internet transactions.In this lab, we will focus on the SSL records sent over the TCP connection. We willattempt to delineate and classify each of the records, with a goal of understandingthe why and how for each record. We investigate the various SSL record types aswell as the fields in the SSL messages. We do so by analyzing a trace of the SSLrecords sent between your host and an e-commerce server.IPsec LabIn this lab (available from the companion Web site), we will explore how to createIPsec SAs between linux boxes. You can do the first part of the lab with two ordi-nary linux boxes, each with one Ethernet adapter. But for the second part of the lab,you will need four linux boxes, two of which having two Ethernet adapters. In thesecond half of the lab, you will create IPsec SAs using the ESP protocol in the tun-nel mode. You will do this by first manually creating the SAs, and then by havingIKE create the SAs.752 CHAPTER 8 SECURITY IN COMPUTER NETWORKSWhat led you to specialize in the networking security area?This is going to sound odd, but the answer is simple: It was fun. My background was in sys-tems programming and systems administration, which leads fairly naturally to security. AndIve always been interested in communications, ranging back to part-time systems program-ming jobs when I was in college.My work on security continues to be motivated by two thingsa desire to keep com-puters useful, which means that their function cant be corrupted by attackers, and a desireto protect privacy.What was your vision for Usenet at the time that you were developing it? And now?We originally viewed it as a way to talk about computer science and computer programmingaround the country, with a lot of local use for administrative matters, for-sale ads, and so on.In fact, my original prediction was one to two messages per day, from 50100 sites at themostever. But the real growth was in people-related topics, includingbut not limitedtohuman interactions with computers. My favorite newsgroups, over the years, have beenthings like rec.woodworking, as well as sci.crypt.To some extent, netnews has been displaced by the Web. Were I to start designing ittoday, it would look very different. But it still excels as a way to reach a very broad audi-ence that is interested in the topic, without having to rely on particular Web sites.Has anyone inspired you professionally? In what ways?Professor Fred Brooksthe founder and original chair of the computer science departmentat the University of North Carolina at Chapel Hill, the manager of the team that developedthe IBM S/360 and OS/360, and the author of The Mythical Man-Monthwas a tremendousinfluence on my career. More than anything else, he taught outlook and trade-offshow to753Steven M. Bellovin joined the faculty at Columbia University aftermany years at the Network Services Research Lab at AT&T LabsResearch in Florham Park, New Jersey. His focus is on networks,security, and why the two are incompatible. In 1995, he wasawarded the Usenix Lifetime Achievement Award for his work in thecreation of Usenet, the first newsgroup exchange network that linkedtwo or more computers and allowed users to share information andjoin in discussions. Steve is also an elected member of the NationalAcademy of Engineering. He received his BA from ColumbiaUniversity and his PhD from the University of North Carolina atChapel Hill.AN INTERVIEW WITH . . .Steven M. Bellovinlook at problems in the context of the real world (and how much messier the real world isthan a theorist would like), and how to balance competing interests in designing a solution.Most computer work is engineeringthe art of making the right trade-offs to satisfy manycontradictory objectives.What is your vision for the future of networking and security?Thus far, much of the security we have has come from isolation. A firewall, for example,works by cutting off access to certain machines and services. But were in an era of increas-ing connectivityits gotten harder to isolate things. Worse yet, our production systemsrequire far more separate pieces, interconnected by networks. Securing all that is one of ourbiggest challenges.What would you say have been the greatest advances in security? How much further dowe have to go?At least scientifically, we know how to do cryptography. Thats been a big help. But mostsecurity problems are due to buggy code, and thats a much harder problem. In fact, its theoldest unsolved problem in computer science, and I think it will remain that way. The chal-lenge is figuring out how to secure systems when we have to build them out of insecurecomponents. We can already do that for reliability in the face of hardware failures; can wedo the same for security?Do you have any advice for students about the Internet and networking security?Learning the mechanisms is the easy part. Learning how to think paranoid is harder. Youhave to remember that probability distributions dont applythe attackers can and will findimprobable conditions. And the details mattera lot.754CHAPTER 9NetworkManagement755Having made our way through the first eight chapters of this text, were now wellaware that a network consists of many complex, interacting pieces of hardware andsoftwarefrom the links, switches, routers, hosts, and other devices that comprisethe physical components of the network to the many protocols (in both hardwareand software) that control and coordinate these devices. When hundreds or thou-sands of such components are cobbled together by an organization to form a net-work, it is not surprising that components will occasionally malfunction, thatnetwork elements will be misconfigured, that network resources will be overuti-lized, or that network components will simply break (for example, a cable will becut or a can of soda will be spilled on top of a router). The network administrator,whose job it is to keep the network up and running, must be able to respond to(and better yet, avoid) such mishaps. With potentially thousands of network compo-nents spread out over a wide area, the network administrator in a network operationscenter (NOC) clearly needs tools to help monitor, manage, and control the network.In this chapter, well examine the architecture, protocols, and information base usedby a network administrator in this task.9.1 What Is Network Management?Before diving in to network management itself, lets first consider a few illustrativereal-world non-networking scenarios in which a complex system with many inter-acting components must be monitored, managed, and controlled by an administra-tor. Electrical power-generation plants have a control room where dials, gauges, andlights monitor the status (temperature, pressure, flow) of remote valves, pipes, ves-sels, and other plant components. These devices allow the operator to monitor theplants many components, and may alert the operator (with the famous flashing redwarning light) when trouble is imminent. Actions are taken by the plant operator tocontrol these components. Similarly, an airplane cockpit is instrumented to allow apilot to monitor and control the many components that make up an airplane. In thesetwo examples, the administrator monitors remote devices and analyzes their datato ensure that they are operational and operating within prescribed limits (for exam-ple, that a core meltdown of a nuclear power plant is not imminent, or that the planeis not about to run out of fuel), reactively controls the system by making adjustmentsin response to the changes within the system or its environment, and proactivelymanages the system (for example, by detecting trends or anomalous behavior,allowing action to be taken before serious problems arise). In a similar sense, thenetwork administrator will actively monitor, manage, and control the system withwhich she or he is entrusted.In the early days of networking, when computer networks were research arti-facts rather than a critical infrastructure used by hundreds of millions of people aday, network management was unheard of. If one encountered a network problem,one might run a few pings to locate the source of the problem and then modify sys-tem settings, reboot hardware or software, or call a remote colleague to do so. (Avery readable discussion of the first major crash of the ARPAnet on October 27,1980, long before network management tools were available, and the efforts takento recover from and understand the crash is [RFC 789].) As the public Internet andprivate intranets have grown from small networks into a large global infrastructure,the need to manage the huge number of hardware and software components withinthese networks more systematically has grown more important as well.In order to motivate our study of network management, lets begin with a sim-ple example. Figure 9.1 illustrates a small network consisting of three routers and anumber of hosts and servers. Even in such a simple network, there are many scenar-ios in which a network administrator might benefit tremendously from havingappropriate network management tools: Detecting failure of an interface card at a host or a router. With appropriate net-work management tools, a network entity (for example, router A) may report tothe network administrator that one of its interfaces has gone down. (This iscertainly preferable to a phone call to the NOC from an irate user who says thenetwork connection is down!) A network administrator who actively monitors756 CHAPTER 9 NETWORK MANAGEMENTand analyzes network traffic may be able to really impress the would-be irateuser by detecting problems in the interface ahead of time and replacing the inter-face card before it fails. This might be done, for example, if the administratornoted an increase in checksum errors in frames being sent by the soon-to-dieinterface. Host monitoring. Here, the network administrator might periodically check tosee if all network hosts are up and operational. Once again, the network adminis-trator may really be able to impress a network user by proactively responding toa problem (host down) before it is reported by a user. Monitoring traffic to aid in resource deployment. A network administrator mightmonitor source-to-destination traffic patterns and notice, for example, that byswitching servers between LAN segments, the amount of traffic that crossesmultiple LANs could be significantly decreased. Imagine the happiness allaround when better performance is achieved with no new equipment costs. Simi-larly, by monitoring link utilization, a network administrator might determinethat a LAN segment or the external link to the outside world is overloaded and9.1 WHAT IS NETWORK MANAGEMENT? 757HostH1ABCHostLink toexternalnetworkServerFigure 9.1 A simple scenario illustrating the uses of network managementthat a higher-bandwidth link should thus be provisioned (alas, at an increasedcost). The network administrator might also want to be notified automaticallywhen congestion levels on a link exceed a given threshold value, in order to pro-vision a higher-bandwidth link before congestion becomes serious. Detecting rapid changes in routing tables. Route flappingfrequent changes in therouting tablesmay indicate instabilities in the routing or a misconfigured router.Certainly, the network administrator who has improperly configured a router wouldprefer to discover the error him- or herself, before the network goes down. Monitoring for SLAs. Service Level Agreements (SLAs) are contracts thatdefine specific performance metrics and acceptable levels of network-providerperformance with respect to these metrics [Huston 1999a]. Verizon and Sprintare just two of the many network providers that guarantee SLAs [AT&T SLA2012; Verizon SLA 2012] to their customers. These SLAs include service avail-ability (outage), latency, throughput, and outage notification requirements.Clearly, if performance criteria are to be part of a service agreement between anetwork provider and its users, then measuring and managing performance willbe of great importance to the network administrator. Intrusion detection. A network administrator may want to be notified when net-work traffic arrives from, or is destined for, a suspicious source (for example,host or port number). Similarly, a network administrator may want to detect (andin many cases filter) the existence of certain types of traffic (for example, source-routed packets, or a large number of SYN packets directed to a given host) thatare known to be characteristic of the types of security attacks that we consideredin Chapter 8.The International Organization for Standardization (ISO) has created a networkmanagement model that is useful for placing the anecdotal scenarios above in amore structured framework. Five areas of network management are defined: Performance management. The goal of performance management is to quan-tify, measure, report, analyze, and control the performance (for example, uti-lization and throughput) of different network components. These componentsinclude individual devices (for example, links, routers, and hosts) as well asend-to-end abstractions such as a path through the network. We will seeshortly that protocol standards such as the Simple Network Management Pro-tocol (SNMP) [RFC 3410] play a central role in Internet performance management. Fault management. The goal of fault management is to log, detect, and respondto fault conditions in the network. The line between fault management and per-formance management is rather blurred. We can think of fault management asthe immediate handling of transient network failures (for example, link, host,or router hardware or software outages), while performance management takes758 CHAPTER 9 NETWORK MANAGEMENTthe longer-term view of providing acceptable levels of performance in the faceof varying traffic demands and occasional network device failures. As withperformance management, the SNMP protocol plays a central role in faultmanagement. Configuration management. Configuration management allows a network man-ager to track which devices are on the managed network and the hardware andsoftware configurations of these devices. An overview of configuration manage-ment and requirements for IP-based networks can be found in [RFC 3139]. Accounting management. Accounting management allows the network managerto specify, log, and control user and device access to network resources. Usagequotas, usage-based charging, and the allocation of resource-access privileges allfall under accounting management. Security management. The goal of security management is to control access tonetwork resources according to some well-defined policy. The key distributioncenters that we studied in Section 8.3 are components of security management.The use of firewalls to monitor and control external access points to ones net-work, a topic we studied in Section 8.9, is another crucial component.In this chapter, well cover only the rudiments of network management. Ourfocus will be purposefully narrowwell examine only the infrastructure fornetwork managementthe overall architecture, network management protocols,and information base through which a network administrator keeps the networkup and running. Well not cover the decision-making processes of the networkadministrator, who must plan, analyze, and respond to the management informa-tion that is conveyed to the NOC. In this area, topics such as fault identificationand management [Katzela 1995; Medhi 1997; Labovitz 1997; Steinder 2002;Feamster 2005; Wu 2005; Teixeira 2006], anomaly detection [Lakhina 2004;Lakhina 2005; Barford 2009], and more come into consideration. Nor will wecover the broader topic of service management [Saydam 1996; RFC 3052]theprovisioning of resources such as bandwidth, server capacity, and the other com-putational/communication resources needed to meet the mission-specific servicerequirements of an enterprise.An often-asked question is What is network management? Our discussionabove has motivated the need for, and illustrated a few of the uses of, network man-agement. Well conclude this section with a single-sentence (albeit a rather long run-on sentence) definition of network management from [Saydam 1996]:Network management includes the deployment, integration, and coordinationof the hardware, software, and human elements to monitor, test, poll, configure,analyze, evaluate, and control the network and element resources to meet thereal-time, operational performance, and Quality of Service requirements at areasonable cost.9.1 WHAT IS NETWORK MANAGEMENT? 759Its a mouthful, but its a good workable definition. In the following sections, welladd some meat to this rather bare-bones definition of network management.9.2 The Infrastructure for Network ManagementWeve seen in the preceding section that network management requires the abilityto monitor, test, poll, configure, . . . and control the hardware and software com-ponents in a network. Because the network devices are distributed, this will, at aminimum, require that the network administrator be able to gather data (for exam-ple, for monitoring purposes) from a remote entity and effect changes at that remoteentity (for example, control it). A human analogy will prove useful here for under-standing the infrastructure needed for network management.Imagine that youre the head of a large organization that has branch officesaround the world. Its your job to make sure that the pieces of your organization areoperating smoothly. How will you do so? At a minimum, youll periodically gatherdata from your branch offices in the form of reports and various quantitative meas-ures of activity, productivity, and budget. Youll occasionally (but not always) beexplicitly notified when theres a problem in one of the branch offices; the branchmanager who wants to climb the corporate ladder (perhaps to get your job) maysend you unsolicited reports indicating how smoothly things are running at his orher branch. Youll sift through the reports you receive, hoping to find smooth opera-tions everywhere but no doubt finding problems in need of your attention. Youmight initiate a one-on-one dialogue with one of your problem branch offices,gather more data in order to understand the problem, and then pass down an execu-tive order (Make this change!) to the branch office manager.Implicit in this very common human scenario is an infrastructure for control-ling the organizationthe boss (you), the remote sites being controlled (the branchoffices), your remote agents (the branch office managers), communication protocols(for transmitting standard reports and data, and for one-on-one dialogues), and data(the report contents and the quantitative measures of activity, productivity, andbudget). Each of these components in human organizational management has acounterpart in network management.The architecture of a network management system is conceptually identical tothis simple human organizational analogy. The network management field has itsown specific terminology for the various components of a network managementarchitecture, and so we adopt that terminology here. As shown in Figure 9.2, thereare three principal components of a network management architecture: a managingentity (the boss in our analogy aboveyou), the managed devices (the branchoffice), and a network management protocol.760 CHAPTER 9 NETWORK MANAGEMENTThe managing entity is an application, typically with a human in the loop,running in a centralized network management station in the NOC. The managingentity is the locus of activity for network management; it controls the collection,processing, analysis, and/or display of network management information. It ishere that actions are initiated to control network behavior and here that thehuman network administrator interacts with the network devices.A managed device is a piece of network equipment (including its software)that resides on a managed network. This is the branch office in our human anal-ogy. A managed device might be a host, router, bridge, hub, printer, or modem.Within a managed device, there may be several so-called managed objects. Thesemanaged objects are the actual pieces of hardware within the managed device (forexample, a network interface card), and the sets of configuration parameters forthe pieces of hardware and software (for example, an intradomain routing proto-col such as RIP). In our human analogy, the managed objects might be the depart-ments within the branch office. These managed objects have pieces of informationassociated with them that are collected into a Management Information Base9.2 THE INFRASTRUCTURE FOR NETWORK MANAGEMENT 761Figure 9.2 Principal components of a network management architectureAgent Data Agent DataAgent DataAgent DataManageddeviceManageddeviceManageddeviceManageddeviceAgent DataManagingentityData(MIB); well see that the values of these pieces of information are available to(and in many cases able to be set by) the managing entity. In our human analogy,the MIB corresponds to quantitative data (measures of activity, productivity, andbudget, with the latter being settable by the managing entity!) exchanged betweenthe branch office and the main office. Well study MIBs in detail in Section 9.3.Finally, also resident in each managed device is a network management agent, aprocess running in the managed device that communicates with the managingentity, taking local actions at the managed device under the command and controlof the managing entity. The network management agent is the branch manager inour human analogy.The third piece of a network management architecture is the network manage-ment protocol. The protocol runs between the managing entity and the manageddevices, allowing the managing entity to query the status of managed devices andindirectly take actions at these devices via its agents. Agents can use the networkmanagement protocol to inform the managing entity of exceptional events (forexample, component failures or violation of performance thresholds). Its importantto note that the network management protocol does not itself manage the network.Instead, it provides capabilities that a network administrator can use to manage(monitor, test, poll, configure, analyze, evaluate, and control) the network. This isa subtle, but important, distinction.Although the infrastructure for network management is conceptually simple,one can often get bogged down with the network-management-speak vocabulary ofmanaging entity, managed device, managing agent, and Management Infor-mation Base. For example, in network-management-speak, in our simple host-monitoring scenario, managing agents located at managed devices areperiodically queried by the managing entitya simple idea, but a linguisticmouthful! With any luck, keeping in mind the human organizational analogy and itsobvious parallels with network management will be of help as we continue throughthis chapter.Our discussion of network management architecture above has been generic,and broadly applies to a number of the network management standards andefforts that have been proposed over the years. Network management standardsbegan maturing in the late 1980s, with OSI CMISE/CMIP (the Common Man-agement Information Services Element/Common Management InformationProtocol) [Piscatello 1993; Stallings 1993; Glitho 1998] and the Internet SNMP(Simple Network Management Protocol) [RFC 3410; Stallings 1999; Rose1996] emerging as the two most important standards [Subramanian 2000]. Bothare designed to be independent of vendor-specific products or networks. BecauseSNMP was quickly designed and deployed at a time when the need for networkmanagement was becoming painfully clear, SNMP found widespread use andacceptance. Today, SNMP has emerged as the most widely used and deployednetwork management framework. Well cover SNMP in detail in the followingsection.762 CHAPTER 9 NETWORK MANAGEMENT9.2 THE INFRASTRUCTURE FOR NETWORK MANAGEMENT 763COMCASTS NETWORK OPERATIONS CENTERComcasts world-class fiber-based IP network delivers converged products and services to49 million combined video, data and voice customers. Comcasts network includes morethan 618,000 plant route miles, 138,000 fiber route miles, 30,000 backbone miles,122,000 optical nodes, and massive storage for the Comcast Content Delivery Network,which delivers a Video on Demand product of more than 134 Terabytes. Each part ofComcasts network, up to and including the customers homes or businesses, is monitoredby one of the companys Operations Centers.Comcast operates two National Network Operations Centers that manage the nationalbackbone, regional area networks, national applications and specific platforms support-ing voice, data and video infrastructure for residential, commercial and wholesale cus-tomers. In addition, Comcast has three Divisional Operations Centers that manage thelocal infrastructure that supports all of their customers. Both the National and DivisionalOperations Centers are accountable for proactively monitoring all aspects of their networkand product performance on a 7 x 24 x 365 basis, utilizing common processes and systems. For example, various network events at the national and local levels have common pre-defined severity levels, recovery processes, and expected Mean Time toRestore objectives. The national and divisional centers can back up each other if a localissue impacts a sites operation. In addition, the National and Divisional OperationsCenters have an extensive Virtual Private Network that allows engineers to securely accessthe network to remotely perform proactive or reactive network management activities.Comcasts approach to network management involves five key areas: PerformanceManagement, Fault Management, Configuration Management, Accounting Managementand Security Management. Performance Management is focused on understandingThese screens show tools supporting correlation, threshold management, tick-eting used by Comcast technicians (Courtesy of Comcast.)(continues)PRINCIPLES IN PRACTICEhow the network/systems and applications (collectively referred to as the ecosystem) areperforming with respect to pre-defined measures specific to time of day, day of week, orspecial events (e.g., storm surges or pay events, such as a boxing match). These pre-defined performance measures exist throughout the service path, from the customers resi-dence or business through the entire network, as well as the interface points to partnersand peers. In addition, synthetic transactions are run to ensure the health of the ecosystemon a continual basis. Fault Management is defined as the ability to detect, log andunderstand anomalies that may impact customers. Comcast utilizes correlation engines toproperly determine an events severity and act appropriately, eliminating or remediatingpotential issues before they affect customers. Configuration Management makes sureappropriate versions of hardware and software are in place across all elements of theecosystem. Keeping these elements at their peak golden levels helps them avoid unin-tended consequences. Accounting Management ensures that the operations centershave a clear understanding of the provisioning and utilization of the ecosystem. This isespecially important to ensure that at all times the operations centers have the ability to re-route traffic effectively. Security Management ensures that the proper controls exist toensure the ecosystem is effectively protected against inappropriate access.Network Operations Centers and the ecosystem they support are not static. Engineeringand Operations personnel are constantly re-evaluating the pre-defined performance measures and tools to ensure that the customers expectations for operational excellenceare met.9.3 The Internet-Standard ManagementFrameworkContrary to what the name SNMP (Simple Network Management Protocol) mightsuggest, network management in the Internet is much more than just a protocol formoving management data between a management entity and its agents, and hasgrown to be much more complex than the word simple might suggest. The currentInternet-Standard Management Framework traces its roots back to the Simple Gate-way Monitoring Protocol, SGMP [RFC 1028]. SGMP was designed by a group ofuniversity network researchers, users, and managers, whose experience with SGMPallowed them to design, implement, and deploy SNMP in just a few months [Lynch1993]a far cry from todays rather drawn-out standardization process. Since then,SNMP has evolved from SNMPv1 through SNMPv2 to the most recent version,SNMPv3 [RFC 3410], released in April 1999 and updated in December 2002.When describing any framework for network management, certain questionsmust inevitably be addressed:764 CHAPTER 9 NETWORK MANAGEMENT What (from a semantic viewpoint) is being monitored? And what form of controlcan be exercised by the network administrator? What is the specific form of the information that will be reported and/or exchanged? What is the communication protocol for exchanging this information?Recall our human organizational analogy from the previous section. The bossand the branch managers will need to agree on the measures of activity, productiv-ity, and budget used to report the branch offices status. Similarly, theyll need toagree on the actions the boss can take (for example, cut the budget, order the branchmanager to change some aspect of the offices operation, or fire the staff and shutdown the branch office). At a lower level of detail, theyll need to agree on the formin which this data is reported. For example, in what currency (dollars, euros?) willthe budget be reported? In what units will productivity be measured? While thesemay seem like trivial details, they must be agreed upon, nonetheless. Finally, themanner in which information is conveyed between the main office and the branchoffices (that is, their communication protocol) must be specified.The Internet-Standard Management Framework addresses the questions posedabove. The framework consists of four parts: Definitions of network management objects, known as MIB objects. In the Inter-net-Standard Management Framework, management information is representedas a collection of managed objects that together form a virtual information store,known as the Management Information Base (MIB). An MIB object might be acounter, such as the number of IP datagrams discarded at a router due to errors inan IP datagram header, or the number of carrier sense errors in an Ethernet inter-face card; descriptive information such as the version of the software running ona DNS server; status information such as whether a particular device is function-ing correctly; or protocol-specific information such as a routing path to adestination. MIB objects thus define the management information maintained bya managed device. Related MIB objects are gathered into MIB modules. In ourhuman organizational analogy, the MIB defines the information conveyedbetween the branch office and the main office. A data definition language, known as SMI (Structure of Management Informa-tion). SMI defines the data types, an object model, and rules for writing andrevising management information. MIB objects are specified in this data defini-tion language. In our human organizational analogy, the SMI is used to definethe details of the format of the information to be exchanged. A protocol, SNMP. SNMP is used for conveying information and commandsbetween a managing entity and an agent executing on behalf of that entity withina managed network device. Security and administration capabilities. The addition of these capabilities rep-resents the major enhancement in SNMPv3 over SNMPv2.9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 765The Internet network management architecture is thus modular by design, with aprotocol-independent data definition language and MIB, and an MIB-independentprotocol. Interestingly, this modular architecture was first put in place to ease the tran-sition from an SNMP-based network management to a network management frame-work being developed by ISO, the competing network management architecture whenSNMP was first conceiveda transition that never occurred. Over time, however,SNMPs design modularity has allowed it to evolve through three major revisions,with each of the four major parts of SNMP discussed above evolving independently.Clearly, the right decision about modularity was made, even if for the wrong reason!In the following subsections, we cover the four major components of the Inter-net-Standard Management Framework in more detail.9.3.1 Structure of Management Information: SMIThe Structure of Management Information, SMI (a rather oddly named componentof the network management framework whose name gives no hint of its functional-ity), is the language used to define the management information residing in a man-aged-network entity. Such a definition language is needed to ensure that the syntaxand semantics of the network management data are well defined and unambiguous.Note that the SMI does not define a specific instance of the data in a managed-networkentity, but rather the language in which such information is specified. The documentsdescribing the SMI for SNMPv3 (which rather confusingly, is called SMIv2) are [RFC2578; RFC 2579; RFC 2580]. Lets examine the SMI in a bottom-up manner, startingwith the base data types in the SMI. Well then look at how managed objects aredescribed in SMI, then how related managed objects are grouped into modules.SMI Base Data TypesRFC 2578 specifies the basic data types in the SMI MIB module-definition language.Although the SMI is based on the ASN.1 (Abstract Syntax Notation One) [ISO X.6802002] object-definition language (see Section 9.4), enough SMI-specific data typeshave been added that SMI should be considered a data definition language in its ownright. The 11 basic data types defined in RFC 2578 are shown in Table 9.1. In additionto these scalar objects, it is also possible to impose a tabular structure on an orderedcollection of MIB objects using the SEQUENCE OF construct; see RFC 2578 fordetails. Most of the data types in Table 9.1 will be familiar (or self-explanatory) tomost readers. The one data type we will discuss in more detail shortly is the OBJECTIDENTIFIER data type, which is used to name an object.SMI Higher-Level ConstructsIn addition to the basic data types, the SMI data definition language also provideshigher-level language constructs.766 CHAPTER 9 NETWORK MANAGEMENTThe OBJECT-TYPE construct is used to specify the data type, status, andsemantics of a managed object. Collectively, these managed objects contain themanagement data that lies at the heart of network management. There are more than10,000 defined objects in various Internet RFCs [RFC 3410]. The OBJECT-TYPEconstruct has four clauses. The SYNTAX clause of an OBJECT-TYPE definitionspecifies the basic data type associated with the object. The MAX-ACCESS clausespecifies whether the managed object can be read, be written, be created, or have itsvalue included in a notification. The STATUS clause indicates whether the objectdefinition is current and valid, obsolete (in which case it should not be implemented,as its definition is included for historical purposes only), or deprecated (obsolete,but implementable for interoperability with older implementations). The DESCRIP-TION clause contains a human-readable textual definition of the object; this docu-ments the purpose of the managed object and should provide all the semanticinformation needed to implement the managed object.As an example of the OBJECT-TYPE construct, consider the ipSystem-StatsInDelivers object-type definition from [RFC 4293]. This object definesa 32-bit counter that keeps track of the number of IP datagrams that were receivedat the managed device and were successfully delivered to an upper-layer protocol.9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 767Data Type DescriptionINTEGER 32-bit integer, as defined in ASN.1, with a value between 231 and 231 1inclusive, or a value from a list of possible named constant values.Integer32 32-bit integer with a value between 231 and 231 1 inclusive.Unsigned32 Unsigned 32-bit integer in the range 0 to 232 1 inclusive.OCTET STRING ASN.1-format byte string representing arbitrary binary or textual data, up to 65,535bytes long.OBJECT IDENTIFIER ASN.1-format administratively assigned (structured name); see Section 9.3.2.IPaddress 32-bit Internet address, in network-byte order.Counter32 32-bit counter that increases from 0 to 232 1 and then wraps around to 0.Counter64 64-bit counter.Gauge32 32-bit integer that will not count above 232 1 nor decrease beyond 0 whenincreased or decreased.TimeTicks Time, measured in 1/100ths of a second since some event.Opaque Uninterpreted ASN.1 string, needed for backward compatibility.Table 9.1 Basic data types of the SMIThe final line of this definition is concerned with the name of this object, a topicwell consider in the following subsection.ipSystemStatsInDelivers OBJECT-TYPESYNTAX Counter32MAX-ACCESS read-onlySTATUS currentDESCRIPTION”The total number of datagrams successfullydelivered to IPuser-protocols (including ICMP).When tracking interface statistics, the counterof the interface to which these datagrams wereaddressed is incremented. This interface mightnot be the same as the input interface forsome of the datagrams.Discontinuities in the value of this counter canoccur at re-initialization of the managementsystem, and at other times as indicated by thevalue of ipSystemStatsDiscontinuityTime.”::= { ipSystemStatsEntry 18 }The MODULE-IDENTITY construct allows related objects to be groupedtogether within a module. For example, [RFC 4293] specifies the MIB module thatdefines managed objects (including ipSystemStatsInDelivers) for manag-ing implementations of the Internet Protocol (IP) and its associated Internet ControlMessage Protocol (ICMP). [RFC 4022] specifies the MIB module for TCP, and [RFC4113] specifies the MIB module for UDP. [RFC 4502] defines the MIB module forRMON remote monitoring. In addition to containing the OBJECT-TYPE definitionsof the managed objects within the module, the MODULE-IDENTITY constructcontains clauses to document contact information of the author of the module, thedate of the last update, a revision history, and a textual description of the module. Asan example, consider the module definition for management of the IP protocol:ipMIB MODULE-IDENTITYLAST-UPDATED “200602020000Z”ORGANIZATION “IETF IPv6 MIB Revision Team”CONTACT-INFO”Editor:Shawn A. RouthierInterworking Labs108 Whispering Pines Dr. Suite 235768 CHAPTER 9 NETWORK MANAGEMENTScotts Valley, CA 95066USAEMail:
DESCRIPTION
The MIB module for managing IP and ICMP
implementations, but excluding their
management of IP routes.
Copyright (C) The Internet Society (2006).
This version of this MIB module is part of
RFC 4293; see the RFC itself for full legal
notices.
REVISION200602020000Z
DESCRIPTION
The IP version neutral revision with added
IPv6 objects for ND, default routers, and
router advertisements. As well as being the
successor to RFC 2011, this MIB is also the
successor to RFCs 2465 and 2466. Published
as RFC 4293.
REVISION199411010000Z
DESCRIPTION
A separate MIB module (IP-MIB) for IP and
ICMP management objects. Published as RFC
2011.
REVISION199103310000Z
DESCRIPTION
The initial revision of this MIB module was
part of MIB-II, which was published as RFC
1213.
::= { mib-2 48}
The NOTIFICATION-TYPE construct is used to specify information regarding
SNMPv2-Trap and InformationRequest messages generated by an agent, or a man-
aging entity; see Section 9.3.3. This information includes a textual DESCRIPTION
of when such messages are to be sent, as well as a list of values to be included in the
message generated; see [RFC 2578] for details. The MODULE-COMPLIANCE
construct defines the set of managed objects within a module that an agent must
implement. The AGENT-CAPABILITIES construct specifies the capabilities of
agents with respect to object- and event-notification definitions.
9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 769
770 CHAPTER 9 NETWORK MANAGEMENT
9.3.2 Management Information Base: MIB
As noted previously, the Management Information Base, MIB, can be thought of
as a virtual information store, holding managed objects whose values collectively
reflect the current state of the network. These values may be queried and/or set by
a managing entity by sending SNMP messages to the agent that is executing in a
managed device on behalf of the managing entity. Managed objects are specified
using the OBJECT-TYPE SMI construct discussed above and gathered into MIB
modules using the MODULE-IDENTITY construct.
The IETF has been busy standardizing the MIB modules associated with routers,
hosts, and other network equipment. This includes basic identification data about a
particular piece of hardware, and management information about the devices net-
work interfaces and protocols. As of 2006 there were more than 200 standards-based
MIB modules and an even larger number of vendor-specific (private) MIB modules.
With all of these standards, the IETF needed a way to identify and name the standard-
ized modules as well as the specific managed objects within a module. Rather than
start from scratch, the IETF adopted a standardized object identification (naming)
framework that had already been put in place by the International Organization for
Standardization (ISO). As is the case with many standards bodies, the ISO had
grand plans for its standardized object identification frameworkto identify every
possible standardized object (for example, data format, protocol, or piece of informa-
tion) in any network, regardless of the network standards organization (for example,
Internet IETF, ISO, IEEE, or ANSI), equipment manufacturer, or network owner. A
lofty goal indeed! The object identification framework adopted by ISO is part of the
ASN.1 (Abstract Syntax Notation One) [ISO X.680 2002] object definition language
that well discuss in Section 9.4. Standardized MIB modules have their own cozy
corner in this all-encompassing naming framework, as discussed below.
As shown in Figure 9.3, objects are named in the ISO naming framework in a
hierarchical manner. Note that each branch point in the tree has both a name and a
number (shown in parentheses); any point in the tree is thus identifiable by the
sequence of names or numbers that specify the path from the root to that point in the
identifier tree. A fun, but incomplete and unofficial, Web-based utility for traversing
part of the object identifier tree (using branch information contributed by volun-
teers) may be found in [OID Repository 2012].
At the top of the hierarchy are the ISO and the Telecommunication Standard-
ization Sector of the International Telecommunication Union (ITU-T), the two
main standards organizations dealing with ASN.1, as well as a branch for joint
efforts by these two organizations. Under the ISO branch of the tree, we find
entries for all ISO standards (1.0) and for standards issued by standards bodies of
various ISO-member countries (1.2). Although not shown in Figure 9.3, under (ISO
member body, a.k.a. 1.2) we would find USA (1.2.840), under which we would
find a number of IEEE, ANSI, and company-specific standards. These include RSA
(1.2.840.11359) and Microsoft (1.2.840.113556), under which we find the
Microsoft File Formats (1.2.840.113556.4) for various Microsoft products, such as
Word (1.2.840.113556.4.2). But we are interested here in networking (not
Microsoft Word files), so let us turn our attention to the branch labeled 1.3, the
standards issued by bodies recognized by the ISO. These include the U.S. Depart-
ment of Defense (6) (under which we will find the Internet standards), the Open
Software Foundation (22), the airline association SITA (69), NATO-identified bod-
ies (57), as well as many other organizations.
Under the Internet branch of the tree (1.3.6.1), there are seven categories.
Under the private (1.3.6.1.4) branch, we find a list [IANA 2009b] of the names
and private enterprise codes of many thousands of private companies that have reg-
istered with the Internet Assigned Numbers Authority (IANA) [IANA 2009a]. Under
the management (1.3.6.1.2) and MIB-2 branches (1.3.6.1.2.1) of the object iden-
tifier tree, we find the definitions of the standardized MIB modules. Whewits a
long journey down to our corner of the ISO name space!
Standardized MIB Modules
The lowest level of the tree in Figure 9.3 shows some of the important hardware-
oriented MIB modules (system and interface) as well as modules associated
9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 771
ITU-T (0) Joint ISO/ITU-T (2)ISO (1)
Standard (0) ISO member
body (2)
ISO identified
organization (3)
NATO
identified (57)
Open Software
Foundation (22)
US
DoD (6)
Internet (1)
directory
(1)
experimental
(3)
security
(5)
mail
(7)
private
(4)
snmpv2
(6)
management
(2)
MIB-2 (1)
system
(1)
address
translation
(3)
icmp
(5)
udp
(7)
cmot
(9)
interface
(2)
ip
(4)
tcp
(6)
egp
(8)
transmission
(10)
rmon
(16)
snmp
(11)
Figure 9.3 ASN.1 object identifier tree
772 CHAPTER 9 NETWORK MANAGEMENT
with some of the most important Internet protocols. [RFC 5000] lists all of the stan-
dardized MIB modules as of 2008. While MIB-related RFCs make for rather tedious
and dry reading, it is instructive (that is, like eating vegetables, it is good for you)
to consider a few MIB module definitions to get a flavor for the type of information
in a module.
The managed objects falling under system contain general information about
the device being managed; all managed devices must support the system MIB
objects. Table 9.2 defines the objects in the system group, as defined in [RFC 1213].
Table 9.3 defines the managed objects in the MIB module for the UDP protocol at a
managed entity.
9.3.3 SNMP Protocol Operations and Transport Mappings
The Simple Network Management Protocol version 2 (SNMPv2) [RFC 3416] is used
to convey MIB information among managing entities and agents executing on behalf
of managing entities. The most common usage of SNMP is in a request-response
mode in which an SNMPv2 managing entity sends a request to an SNMPv2 agent,
who receives the request, performs some action, and sends a reply to the request. Typ-
ically, a request will be used to query (retrieve) or modify (set) MIB object values
Object Identifier Name Type Description (from RFC 1213)
1.3.6.1.2.1.1.1 sysDescr OCTET STRING Full name and version identification of the systems hardware
type, software operating-system, and networking software.
1.3.6.1.2.1.1.2 sysObjectID OBJECT IDENTIFIER Vendor-assigned object ID that provides an easy and
unambiguous means for determining what kind of box is
being managed.
1.3.6.1.2.1.1.3 sysUpTime TimeTicks The time (in hundredths of a second) since the network
management portion of the system was last re-initialized.
1.3.6.1.2.1.1.4 sysContact OCTET STRING The contact person for this managed node, together with infor-
mation on how to contact this person.
1.3.6.1.2.1.1.5 sysName OCTET STRING An administratively assigned name for this managed node. By
convention, this is the nodes fully qualified domain name.
1.3.6.1.2.1.1.6 sysLocation OCTET STRING The physical location of this node.
1.3.6.1.2.1.1.7 sysServices Integer32 A coded value that indicates the set of services available
at this node: physical (for example, a repeater), data
link/subnet (for example, bridge), Internet (for example,
IP gateway), end-to-end (for example, host), applications.
Table 9.2 Managed objects in the MIB-2 system group
associated with a managed device. A second common usage of SNMP is for an agent
to send an unsolicited message, known as a trap message, to a managing entity. Trap
messages are used to notify a managing entity of an exceptional situation that has
resulted in changes to MIB object values. We saw earlier in Section 9.1 that the net-
work administrator might want to receive a trap message, for example, when an inter-
face goes down, congestion reaches a predefined level on a link, or some other
noteworthy event occurs. Note that there are a number of important trade-offs between
polling (request-response interaction) and trapping; see the homework problems.
SNMPv2 defines seven types of messages, known generically as protocol data
unitsPDUsas shown in Table 9.4 and described next. The format of the PDU is
shown in Figure 9.4.
The GetRequest, GetNextRequest, and GetBulkRequest PDUs are
all sent from a managing entity to an agent to request the value of one or more
MIB objects at the agents managed device. The object identifiers of the MIB
objects whose values are being requested are specified in the variable binding
portion of the PDU. GetRequest, GetNextRequest, and GetBulkRe-
quest differ in the granularity of their data requests. GetRequest can request
an arbitrary set of MIB values; multiple GetNextRequests can be used to
sequence through a list or table of MIB objects; GetBulkRequest allows a
large block of data to be returned, avoiding the overhead incurred if multiple
GetRequest or GetNextRequest messages were to be sent. In all three
cases, the agent responds with a Response PDU containing the object identi-
fiers and their associated values.
The SetRequest PDU is used by a managing entity to set the value of one or
more MIB objects in a managed device. An agent replies with a Response PDU
with the noError error status to confirm that the value has indeed been set.
Object Identifier Name Type Description (from RFC 4113)
1.3.6.1.2.1.7.1 udpInDatagrams Counter32 total number of UDP datagrams delivered to UDP users
1.3.6.1.2.1.7.2 udpNoPorts Counter32 total number of received UDP datagrams for which there
was no application at the destination port
1.3.6.1.2.1.7.3 udpInErrors Counter32 number of received UDP datagrams that could not be
delivered for reasons other than the lack of an application
at the destination port
1.3.6.1.2.1.7.4 udpOutDatagrams Counter32 total number of UDP datagrams sent from this entity
Table 9.3 Selected managed objects in the MIB-2 UDP module
9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 773
774 CHAPTER 9 NETWORK MANAGEMENT
PDU
type
(03)
Request
Id
Error
Status
(05)
Error
Index
Name
ValueName
Name Value
PDU
Type
(4)
Enterprise Agent
Addr
Trap
Type
(07)
Specific
code
Time
stamp
Value
Get/set header
Trap header Trap information
SNMP PDU
Variables to get/set
Figure 9.4 SNMP PDU format
SNMPv2 PDU Type Sender-receiver Description
GetRequest manager-to-agent get value of one or more MIB object instances
GetNextRequest manager-to-agent get value of next MIB object instance in list or table
GetBulkRequest manager-to-agent get values in large block of data, for example, values in a
large table
InformRequest manager-to-manager inform remote managing entity of MIB values
remote to its access
SetRequest manager-to-agent set value of one or more MIB object instances
Response agent-to-manager or generated in response to
manager-to-manager GetRequest,
GetNextRequest,
GetBulkRequest,
SetRequest PDU, or
InformRequest
SNMPv2-Trap agent-to-manager inform manager of an exceptional event
Table 9.4 SNMPv2 PDU types
The InformRequest PDU is used by a managing entity to notify another
managing entity of MIB information that is remote to the receiving entity. The
receiving entity replies with a Response PDU with the noError error status
to acknowledge receipt of the InformRequest PDU.
The final type of SNMPv2 PDU is the trap message. Trap messages are generated
asynchronously; that is, they are not generated in response to a received request but
rather in response to an event for which the managing entity requires notification.
RFC 3418 defines well-known trap types that include a cold or warm start by a
device, a link going up or down, the loss of a neighbor, or an authentication failure
event. A received trap request has no required response from a managing entity.
Given the request-response nature of SNMPv2, it is worth noting here that
although SNMP PDUs can be carried via many different transport protocols, the
SNMP PDU is typically carried in the payload of a UDP datagram. Indeed, RFC
3417 states that UDP is the preferred transport mapping. Since UDP is an unreli-
able transport protocol, there is no guarantee that a request, or its response, will be
received at the intended destination. The request ID field of the PDU is used by the
managing entity to number its requests to an agent; an agents response takes its
request ID from that of the received request. Thus, the request ID field can be used
by the managing entity to detect lost requests or replies. It is up to the managing
entity to decide whether to retransmit a request if no corresponding response is
received after a given amount of time. In particular, the SNMP standard does not
mandate any particular procedure for retransmission, or even if retransmission is to
be done in the first place. It only requires that the managing entity needs to act
responsibly in respect to the frequency and duration of retransmissions. This, of
course, leads one to wonder how a responsible protocol should act!
9.3.4 Security and Administration
The designers of SNMPv3 have said that SNMPv3 can be thought of as SNMPv2
with additional security and administration capabilities [RFC 3410]. Certainly,
there are changes in SNMPv3 over SNMPv2, but nowhere are those changes more
evident than in the area of administration and security. The central role of security
in SNMPv3 was particularly important, since the lack of adequate security resulted
in SNMP being used primarily for monitoring rather than control (for example,
SetRequest is rarely used in SNMPv1).
As SNMP has matured through three versions, its functionality has grown but
so too, alas, has the number of SNMP-related standards documents. This is evi-
denced by the fact that there is even now an RFC [RFC 3411] that describes an
architecture for describing SNMP Management Frameworks! While the notion of
an architecture for describing a framework might be a bit much to wrap ones
mind around, the goal of RFC 3411 is an admirable oneto introduce a common
language for describing the functionality and actions taken by an SNMPv3 agent or
9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 775
776 CHAPTER 9 NETWORK MANAGEMENT
Command
generator
SNMP
applications
SNMP
engine
Notification
receiver
Proxy
forwarder
Dispatching
Message-
processing
system
Timeliness,
authentication,
privacy
Security
Transport layer
Access
control
Command
generator
Notification
originator
Other
PDU
Security/message header PDU
Figure 9.5 SNMPv3 engine and applications
managing entity. The architecture of an SNMPv3 entity is straightforward, and a
tour through the architecture will serve to solidify our understanding of SNMP.
So-called SNMP applications consist of a command generator, notification
receiver, and proxy forwarder (all of which are typically found in a managing
entity); a command responder and notification originator (both of which are typi-
cally found in an agent); and the possibility of other applications. The command
generator generates the GetRequest, GetNextRequest, GetBulkRequest,
and SetRequest PDUs that we examined in Section 9.3.3 and handles the
received responses to these PDUs. The command responder executes in an agent
and receives, processes, and replies (using the Response message) to received
GetRequest, GetNextRequest, GetBulkRequest, and SetRequest
PDUs. The notification originator application in an agent generates Trap PDUs;
these PDUs are eventually received and processed in a notification receiver applica-
tion at a managing entity. The proxy forwarder application forwards request, notifi-
cation, and response PDUs.
A PDU sent by an SNMP application next passes through the SNMP engine
before it is sent via the appropriate transport protocol. Figure 9.5 shows how a PDU
generated by the command generator application first enters the dispatch module,
where the SNMP version is determined. The PDU is then processed in the message-
processing system, where the PDU is wrapped in a message header containing the
SNMP version number, a message ID, and message size information. If encryption or
authentication is needed, the appropriate header fields for this information are
included as well; see [RFC 3411] for details. Finally, the SNMP message (the applica-
tion-generated PDU plus the message header information) is passed to the appropriate
transport protocol. The preferred transport protocol for carrying SNMP messages is
UDP (that is, SNMP messages are carried as the payload in a UDP datagram), and the
preferred port number for the SNMP is port 161. Port 162 is used for trap messages.
We have seen above that SNMP messages are used not just to monitor, but also
to control (for example, through the SetRequest command) network elements.
Clearly, an intruder that could intercept SNMP messages and/or generate its own
SNMP packets into the management infrastructure could wreak havoc in the net-
work. Thus, it is crucial that SNMP messages be transmitted securely. Surprisingly,
it is only in the most recent version of SNMP that security has received the attention
that it deserves. SNMPv3 security is known as user-based security [RFC 3414] in
that there is the traditional concept of a user, identified by a username, with which
security information such as a password, key value, or access privileges are associ-
ated. SNMPv3 provides for encryption, authentication, protection against playback
attacks (see Section 8.3), and access control.
Encryption. SNMP PDUs can be encrypted using the Data Encryption Standard
(DES) in Cipher Block Chaining (CBC) mode. Note that since DES is a shared-
key system, the secret key of the user encrypting data must be known by the
receiving entity that must decrypt the data.
Authentication. SNMP uses the Message Authentication Code (MAC) technique
that we studied in Section 8.3.1 to provide both authentication and protection
against tampering [RFC 4301]. Recall that a MAC requires the sender and
receiver both to know a common secret key.
Protection against playback. Recall from our discussion in Chapter 8 that nonces
can be used to guard against playback attacks. SNMPv3 adopts a related
approach. In order to ensure that a received message is not a replay of some
earlier message, the receiver requires that the sender include a value in each mes-
sage that is based on a counter in the receiver. This counter, which functions as a
nonce, reflects the amount of time since the last reboot of the receivers network
management software and the total number of reboots since the receivers net-
work management software was last configured. As long as the counter in a
received message is within some margin of error of the receivers actual value,
the message is accepted as a nonreplay message, at which point it may be authen-
ticated and/or decrypted. See [RFC 3414] for details.
Access control. SNMPv3 provides a view-based access control [RFC 3415] that
controls which network management information can be queried and/or set
by which users. An SNMP entity retains information about access rights and
9.3 THE INTERNET-STANDARD MANAGEMENT FRAMEWORK 777
778 CHAPTER 9 NETWORK MANAGEMENT
policies in a Local Configuration Datastore (LCD). Portions of the LCD are
themselves accessible as managed objects, defined in the View-Based Access
Control Model Configuration MIB [RFC 3415], and thus can be managed and
manipulated remotely via SNMP.
9.4 ASN.1
In this book, we have covered a number of interesting topics in computer networking.
This section on ASN.1, however, may not make the top-ten list of interesting topics.
Like vegetables, knowledge about ASN.1 and the broader issue of presentation serv-
ices is something that is good for you. ASN.1 is an ISO-originated standard that is
used in a number of Internet-related protocols, particularly in the area of network man-
agement. For example, we saw in Section 9.3 that MIB variables in SNMP were inex-
tricably tied to ASN.1. So while the material on ASN.1 in this section may be rather
dry, we hope the reader will take it on faith that the material is important.
In order to motivate our discussion here, consider the following thought experi-
ment. Suppose one could reliably copy data from one computers memory directly
into a remote computers memory. If one could do this, would the communication
problem be solved? The answer to the question depends on ones definition of
the communication problem. Certainly, a perfect memory-to-memory copy would
exactly communicate the bits and bytes from one machine to another. But does such
an exact copy of the bits and bytes mean that when software running on the receiv-
ing computer accesses this data, it will see the same values that were stored into the
sending computers memory? The answer to this question is not necessarily! The
crux of the problem is that different computer architectures, different operating sys-
tems, and different compilers have different conventions for storing and represent-
ing data. If data is to be communicated and stored among multiple computers (as it
is in every communication network), this problem of data representation must
clearly be solved.
As an example of this problem, consider the simple C code fragment below.
How might this structure be laid out in memory?
struct {
char code;
int x;
} test;
test.x = 259;
test.code = a;
The left side of Figure 9.6 shows a possible layout of this data on one hypo-
thetical architecture: there is a single byte of memory containing the character a,
followed by a 16-bit word containing the integer value 259, stored with the most
significant byte first. The layout in memory on another computer is shown in the
right half of Figure 9.6. The character a is followed by the integer value stored
with the least significant byte stored first and with the 16-bit integer aligned to
start on a 16-bit word boundary. Certainly, if one were to perform a verbatim
copy between these two computers memories and use the same structure defini-
tion to access the stored values, one would see very different results on the two
computers!
The fact that different architectures have different internal data formats is a real
and pervasive problem. The particular problem of integer storage in different for-
mats is so common that it has a name. Big-endian order for storing integers has
the most significant bytes of the integer stored first (at the lowest storage address).
Little-endian order stores the least significant bytes first. Sun SPARC and
Motorola processors are big-endian, while Intel processors are little-endian. As an
aside, the terms big-endian and little-endian come from the book, Gullivers
Travels, by Jonathan Swift, in which two groups of people dogmatically insist
on doing a simple thing in two different ways (hopefully, the analogy to the com-
puter architecture community is clear). One group in the land of Lilliput insists on
breaking their eggs at the larger end (the big-endians), while the other insists
on breaking them at the smaller end. The difference was the cause of great civil
strife and rebellion.
Given that different computers store and represent data in different ways,
how should networking protocols deal with this? For example, if an SNMP agent
is about to send a Response message containing the integer count of the number
of received UDP datagrams, how should it represent the integer value to be sent
to the managing entityin big-endian or little-endian order? One option would
be for the agent to send the bytes of the integer in the same order in which they
would be stored in the managing entity. Another option would be for the agent to
send in its own storage order and have the receiving entity reorder the bytes, as
needed. Either option would require the sender or receiver to learn the others
format for integer representation.
9.4 ASN.1 779
a
00000001
00000011
test.code
test.x
test.code
test.x
a
00000011
00000001
Figure 9.6 Two different data layouts on two different architectures
780 CHAPTER 9 NETWORK MANAGEMENT
Aging
60s hippie
Groovy
Grandma 2012 Teenager
Groo
vy
Figure 9.7 The presentation problem
A third option is to have a machine-independent, OS-independent, language-
independent method for describing integers and other data types (that is, a data-
definition language) and rules that state the manner in which each of the data types
is to be transmitted over the network. When data of a given type is received, it is
received in a known format and can then be stored in whatever machine-specific
format is required. Both the SMI that we studied in Section 9.3 and ASN.1 adopt
this third option. In ISO parlance, these two standards describe a presentation
servicethe service of transmitting and translating information from one machine-
specific format to another. Figure 9.7 illustrates a real-world presentation problem;
neither receiver understands the essential idea being communicatedthat the
speaker likes something. As shown in Figure 9.8, a presentation service can solve
this problem by translating the idea into a commonly understood (by the presenta-
tion service), person-independent language, sending that information to the receiver,
and then translating into a language understood by the receiver.
Table 9.5 shows a few of the ASN.1-defined data types. Recall that we
encountered the INTEGER, OCTET STRING, and OBJECT IDENTIFIER data
types in our earlier study of the SMI. Since our goal here is (mercifully) not to
provide a complete introduction to ASN.1, we refer the reader to the standards or
to the printed and online book [Larmouth 1996] for a description of ASN.1 types
and constructors, such as SEQUENCE and SET, that allow for the definition of
structured data types.
In addition to providing a data definition language, ASN.1 also provides Basic
Encoding Rules (BER) that specify how instances of objects that have been defined
using the ASN.1 data definition language are to be sent over the network. The BER
adopts a so-called TLV (Type, Length, Value) approach to encoding data for
transmission. For each data item to be sent, the data type, the length of the data item,
and then the actual value of the data item are sent, in that order. With this simple
convention, the received data is essentially self-identifying.
Figure 9.9 shows how the two data items in a simple example would be sent.
In this example, the sender wants to send the character string smith followed by
the value 259 decimal (which equals 00000001 00000011 in binary, or a byte value
of 1 followed by a byte value of 3), assuming big-endian order. The first byte in the
9.4 ASN.1 781
Aging
60s hippie
Presentation
service
Grandma 2012 Teenager
Presentation
service
It is pleasing
Cats pajamas Awesome
Groovy
It is pleasing Presentation
service
Figure 9.8 The presentation problem solved
Tag Type Description
1 BOOLEAN value is true or false
2 INTEGER can be arbitrarily large
3 BITSTRING list of one or more bits
4 OCTET STRING list of one or more bytes
5 NULL no value
6 OBJECT IDENTIFIER name, in the ASN.1 standard naming
tree; see Section 9.2.2
9 REAL floating point
Table 9.5 Selected ASN.1 data types
782 CHAPTER 9 NETWORK MANAGEMENT
lastname ::= OCTET STRING
weight ::= INTEGER
{weight, 259}
{lastname, smith}
Module of data type
declarations written
in ASN.1
Instances of data type
specified in module
Basic Encoding Rules
(BER)
Transmitted
byte stream
3
1
2
2
h
t
i
m
s
5
4
Figure 9.9 BER encoding example
transmitted stream has the value 4, indicating that the type of the following data
item is an OCTET STRING; this is the T in the TLV encoding. The second byte
in the stream contains the length of the OCTET STRING, in this case 5. The third
byte in the transmitted stream begins the OCTET STRING of length 5; it contains
the ASCII representation of the letter s. The T, L, and V values of the next data item
are 2 (the INTEGER type tag value), 2 (that is, an integer of length 2 bytes), and
the 2-byte big-endian representation of the value 259 decimal.
In our previous discussion, we have only touched on a small and simple subset
of ASN.1. Resources for learning more about ASN.1 include the ASN.1 standards
document [ISO X.680 2002], the online OSI-related book [Larmouth 2012], and the
ASN.1-related Web sites, [OSS 2012] and [OID Repository 2012].
9.5 Conclusion
Our study of network management, and indeed of all of networking, is now complete!
In this final chapter on network management, we began by motivating the need
for providing appropriate tools for the network administratorthe person whose job
it is to keep the network up and runningfor monitoring, testing, polling,
configuring, analyzing, evaluating, and controlling the operation of the network.
Our analogies with the management of complex systems such as power plants,
airplanes, and human organization helped motivate this need. We saw that the
architecture of network management systems revolves around five key components:
(1) a network manager, (2) a set of managed remote (from the network manager)
devices, (3) the Management Information Bases (MIBs) at these devices, containing
data about the devices status and operation, (4) remote agents that report MIB infor-
mation and take action under the control of the network manager, and (5) a protocol
for communication between the network manager and the remote devices.
We then delved into the details of the Internet-Standard Management Frame-
work, and the SNMP protocol in particular. We saw how SNMP instantiates the five
key components of a network management architecture, and we spent considerable
time examining MIB objects, the SMIthe data definition language for specifying
MIBs, and the SNMP protocol itself. Noting that the SMI and ASN.1 are inextrica-
bly tied together, and that ASN.1 plays a key role in the presentation layer in the
ISO/OSI seven-layer reference model, we then briefly examined ASN.1. Perhaps
more important than the details of ASN.1 itself was the noted need to provide for
translation between machine-specific data formats in a network. While some net-
work architectures explicitly acknowledge the importance of this service by having
a presentation layer, this layer is absent in the Internet protocol stack.
It is also worth noting that there are many topics in network management that
we chose not to covertopics such as fault identification and management, proac-
tive anomaly detection, alarm correlation, and the larger issues of service manage-
ment (for example, as opposed to network management). While important, these
topics would form a text in their own right, and we refer the reader to the references
noted in Section 9.1.
Homework Problems and Questions
Chapter 9 Review Questions
SECTION 9.1
R1. Why would a network manager benefit from having network management
tools? Describe five scenarios.
R2. What are the five areas of network management defined by the ISO?
HOMEWORK PROBLEMS AND QUESTIONS 783
784 CHAPTER 9 NETWORK MANAGEMENT
R3. What is the difference between network management and service
management?
SECTION 9.2
R4. Define the following terms: managing entity, managed device, management
agent, MIB, network management protocol.
SECTION 9.3
R5. What is the role of the SMI in network management?
R6. What is an important difference between a request-response message and a
trap message in SNMP?
R7. What are the seven message types used in SNMP?
R8. What is meant by an SNMP engine?
SECTION 9.4
R9. What is the purpose of the ASN.1 object identifier tree?
R10. What is the role of ASN.1 in the ISO/OSI reference models presentation layer?
R11. Does the Internet have a presentation layer? If not, how are concerns about
differences in machine architecturesfor example, the different representa-
tion of integers on different machinesaddressed?
R12. What is meant by TLV encoding?
Problems
P1. Consider the two ways in which communication occurs between a managing
entity and a managed device: request-response mode and trapping. What
are the pros and cons of these two approaches, in terms of (1) overhead,
(2) notification time when exceptional events occur, and (3) robustness
with respect to lost messages between the managing entity and the
device?
P2. In Section 9.3 we saw that it was preferable to transport SNMP messages in
unreliable UDP datagrams. Why do you think the designers of SNMP chose
UDP rather than TCP as the transport protocol of choice for SNMP?
P3. What is the ASN.1 object identifier for the ICMP protocol (see Figure 9.3)?
P4. Suppose you worked for a US-based company that wanted to develop its
own MIB for managing a product line. Where in the object identifier tree
(Figure 9.3) would it be registered? (Hint: Youll have to do some digging
through RFCs or other documents to answer this question.)
P5. Recall from Section 9.3.2 that a private company (enterprise) can create its
own MIB variables under the private branch 1.3.6.4. Suppose that IBM
wanted to create a MIB for its Web server software. What would be the next
OID qualifier after 1.3.6.1.4? (In order to answer this question, you will need
to consult [IANA 2009b]). Search the Web and see if you can find out
whether such a MIB exists for an IBM server.
P6. Why do you think the length precedes the value in a TLV encoding (rather
than the length following the value)?
P7. Consider Figure 9.9. What would be the BER encoding of {weight, 165}
{lastname, Michael}?
P8. Consider Figure 9.9. What would be the BER encoding of {weight, 145}
{lastname, Sridhar}?
PROBLEMS 785
Please describe one or two of the most exciting projects you have worked on during your
career. What were the biggest challenges?
When I was a researcher at AT&T, a group of us designed a new way to manage routing in
Internet Service Provider backbone networks. Traditionally, network operators configure
each router individually, and these routers run distributed protocols to compute paths
through the network. We believed that network management would be simpler and more
flexible if network operators could exercise direct control over how routers forward traffic
based on a network-wide view of the topology and traffic. The Routing Control Platform
(RCP) we designed and built could compute the routes for all of AT&Ts backbone on a
single commodity computer, and could control legacy routers without modification. To me,
this project was exciting because we had a provocative idea, a working system, and
ultimately a real deployment in an operational network.
What changes and innovations do you see happening in network management in the
future?
Rather than simply bolting on network management on top of existing networks,
researchers and practitioners alike are starting to design networks that are fundamentally
easier to manage. Like our early work on the RCP, the main idea in so-called Software
Defined Networking (SDN) is to run a controller that can install low-level packet-handling
rules in the underlying switches using a standard protocol. This controller can run various
786
Jennifer Rexford
Jennifer Rexford is a Professor in the Computer Science department at
Princeton University. Her research has the broad goal of making com-
puter networks easier to design and manage, with particular emphasis
on routing protocols. From 19962004, she was a member of the
Network Management and Performance department at AT&T Labs
Research. While at AT&T, she designed techniques and tools for net-
work measurement, traffic engineering, and router configuration that were deployed in AT&Ts
backbone network. Jennifer is co-author of the book Web Protocols and Practice:
Networking Protocols, Caching, and Traffic Measurement, published by Addison-Wesley in
May 2001. She served as the chair of ACM SIGCOMM from 2003 to 2007. She received
her BSE degree in electrical engineering from Princeton University in 1991, and her MSE
and PhD degrees in electrical engineering and computer science from the University of
Michigan in 1993 and 1996, respectively. In 2004, Jennifer was the winner of ACMs
Grace Murray Hopper Award for outstanding young computer professional and appeared on
the MIT TR-100 list of top innovators under the age of 35.
AN INTERVIEW WITH
network-management applications, such as dynamic access control, seamless user mobility,
traffic engineering, server load balancing, energy-efficient networking, and so on. I believe
SDN is a great opportunity to get network management right, by rethinking the relationship
between the network devices and the software that manages them.
Where do you see the future of networking and the Internet?
Networking is an exciting field because the applications and the underlying technologies
change all the time. We are always reinventing ourselves! Who would have predicted even
five or ten years ago the dominance of smart phones, allowing mobile users to access
existing applications as well as new location-based services? The emergence of cloud com-
puting is fundamentally changing the relationship between users and the applications they
run, and networked sensors are enabling a wealth of new applications. The pace of innova-
tion is truly inspiring.
The underlying network is a crucial component in all of these innovations. Yet, the
network is notoriously in the waylimiting performance, compromising reliability,
constraining applications, and complicating the deployment and management of services.
We should strive to make the network of the future as invisible as the air we breathe, so it
never stands in the way of new ideas and valuable services. To do this, we need to raise the
level of abstraction above individual network devices and protocols (and their attendant
acronyms!), so we can reason about the network as a whole.
What people inspired you professionally?
Ive long been inspired by Sally Floyd at the International Computer Science Institute.
Her research is always purposeful, focusing on the important challenges facing the Internet.
She digs deeply into hard questions until she understands the problem and the space of
solutions completely, and she devotes serious energy into making things happen, such as
pushing her ideas into protocol standards and network equipment. Also, she gives back to
the community, through professional service in numerous standards and research organiza-
tions and by creating tools (such as the widely used ns-2 and ns-3 simulators) that enable
other researchers to succeed. She retired in 2009 but her influence on the field will be felt
for years to come.
What are your recommendations for students who want careers in computer science and
networking?
Networking is an inherently interdisciplinary field. Applying techniques from other disciplines
to networking problems is a great way to move the field forward. Weve seen tremendous
787
breakthroughs in networking come from such diverse areas as queuing theory, game
theory, control theory, distributed systems, network optimization, programming languages,
machine learning, algorithms, data structures, and so on. I think that becoming conversant
in a related field, or collaborating closely with experts in those fields, is a wonderful
way to put networking on a stronger foundation, so we can learn how to build networks
that are worthy of societys trust. Beyond the theoretical disciplines, networking is exciting
because we create real artifacts that real people use. Mastering how to design and build
systemsby gaining experience in operating systems, computer architecture, and so
onis another fantastic way to amplify your knowledge of networking to help change
the world.
788
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789
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[RFC 001] S. Crocker, Host Software, RFC 001 (the very first RFC!).
[RFC 768] J. Postel, User Datagram Protocol, RFC 768, Aug. 1980.
[RFC 789] E. Rosen, Vulnerabilities of Network Control Protocols, RFC 789.
[RFC 791] J. Postel, Internet Protocol: DARPA Internet Program Protocol Specification,
RFC 791, Sept. 1981.
[RFC 792] J. Postel, Internet Control Message Protocol, RFC 792, Sept. 1981.
[RFC 793] J. Postel, Transmission Control Protocol, RFC 793, Sept. 1981.
[RFC 801] J. Postel, NCP/TCP Transition Plan, RFC 801, Nov. 1981.
[RFC 826] D. C. Plummer, An Ethernet Address Resolution ProtocolorConverting
Network Protocol Addresses to 48 bit Ethernet Address for Transmission on Ethernet
Hardware, RFC 826, Nov. 1982.
[RFC 829] V. Cerf, Packet Satellite Technology Reference Sources, RFC 829, Nov.
1982.
[RFC 854] J. Postel, J. Reynolds, TELNET Protocol Specification, RFC 854, May 1993.
[RFC 950] J. Mogul, J. Postel, Internet Standard Subnetting Procedure, RFC 950, Aug.
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[RFC 959] J. Postel and J. Reynolds, File Transfer Protocol (FTP), RFC 959, Oct. 1985.
[RFC 977] B. Kantor, P. Lapsley, Network News Transfer Protocol, RFC 977, Feb. 1986.
[RFC 1028] J. Davin, J.D. Case, M. Fedor, M. Schoffstall, A Simple Gateway Monitoring
Protocol, RFC 1028, Nov. 1987.
[RFC 1034] P. V. Mockapetris, Domain NamesConcepts and Facilities, RFC 1034,
Nov. 1987.
[RFC 1035] P. Mockapetris, Domain NamesImplementation and Specification, RFC
1035, Nov. 1987.
[RFC 1058] C. L. Hendrick, Routing Information Protocol, RFC 1058, June 1988.
810 REFERENCES
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[RFC 1071] R. Braden, D. Borman, and C. Partridge, Computing The Internet Checksum,
RFC 1071, Sept. 1988.
[RFC 1075] D. Waitzman, C. Partridge, S. Deering, Distance Vector Multicast Routing
Protocol, RFC 1075, Nov. 1988.
[RFC 1112] S. Deering, Host Extension for IP Multicasting, RFC 1112, Aug. 1989.
[RFC 1122] R. Braden, Requirements for Internet HostsCommunication Layers, RFC
1122, Oct. 1989.
[RFC 1123] R. Braden, ed., Requirements for Internet HostsApplication and Support,
RFC-1123, Oct. 1989.
[RFC 1142] D. Oran, OSI IS-IS Intra-Domain Routing Protocol, RFC 1142, Feb. 1990.
[RFC 1190] C. Topolcic, Experimental Internet Stream Protocol: Version 2 (ST-II), RFC
1190, Oct. 1990.
[RFC 1191] J. Mogul, S. Deering, Path MTU Discovery, RFC 1191, Nov. 1990.
[RFC 1213] K. McCloghrie, M. T. Rose, Management Information Base for Network
Management of TCP/IP-based internets: MIB-II, RFC 1213, Mar. 1991.
[RFC 1256] S. Deering, ICMP Router Discovery Messages, RFC 1256, Sept. 1991.
[RFC 1320] R. Rivest, The MD4 Message-Digest Algorithm, RFC 1320, Apr. 1992.
[RFC 1321] R. Rivest, The MD5 Message-Digest Algorithm, RFC 1321, Apr. 1992.
[RFC 1323] V. Jacobson, S. Braden, D. Borman, TCP Extensions for High Performance,
RFC 1323, May 1992.
[RFC 1422] S. Kent, Privacy Enhancement for Internet Electronic Mail: Part II: Certificate-
Based Key Management, RFC 1422.
[RFC 1546] C. Partridge, T. Mendez, W. Milliken, Host Anycasting Service, RFC 1546,
1993.
[RFC 1547] D. Perkins, Requirements for an Internet Standard Point-to-Point Protocol,
RFC 1547, Dec. 1993.
[RFC 1584] J. Moy, Multicast Extensions to OSPF, RFC 1584, Mar. 1994.
[RFC 1633] R. Braden, D. Clark, S. Shenker, Integrated Services in the Internet
Architecture: an Overview, RFC 1633, June 1994.
[RFC 1636] R. Braden, D. Clark, S. Crocker, C. Huitema, Report of IAB Workshop on
Security in the Internet Architecture, RFC 1636, Nov. 1994.
[RFC 1661] W. Simpson (ed.), The Point-to-Point Protocol (PPP), RFC 1661, July 1994.
[RFC 1662] W. Simpson (ed.), PPP in HDLC-Like Framing, RFC 1662, July 1994.
[RFC 1700] J. Reynolds and J. Postel, Assigned Numbers, RFC 1700, Oct. 1994.
[RFC 1752] S. Bradner, A. Mankin, The Recommendations for the IP Next Generation
Protocol, RFC 1752, Jan. 1995.
[RFC 1918] Y. Rekhter, B. Moskowitz, D. Karrenberg, G. J. de Groot, E. Lear, Address
Allocation for Private Internets, RFC 1918, Feb. 1996.
[RFC 1930] J. Hawkinson, T. Bates, Guidelines for Creation, Selection, and Registration of
an Autonomous System (AS), RFC 1930, Mar. 1996.
[RFC 1938] N. Haller, C. Metz, A One-Time Password System, RFC 1938, May 1996.
[RFC 1939] J. Myers and M. Rose, Post Office ProtocolVersion 3, RFC 1939, May 1996.
REFERENCES 811
[RFC 1945] T. Berners-Lee, R. Fielding, H. Frystyk, Hypertext Transfer Protocol
HTTP/1.0, RFC 1945, May 1996.
[RFC 2003] C. Perkins, IP Encapsulation within IP, RFC 2003, Oct. 1996.
[RFC 2004] C. Perkins, Minimal Encapsulation within IP, RFC 2004, Oct. 1996.
[RFC 2018] M. Mathis, J. Mahdavi, S. Floyd, A. Romanow, TCP Selective
Acknowledgment Options, RFC 2018, Oct. 1996.
[RFC 2050] K. Hubbard, M. Kosters, D. Conrad, D. Karrenberg, J. Postel, Internet Registry
IP Allocation Guidelines, RFC 2050, Nov. 1996.
[RFC 2104] H. Krawczyk, M. Bellare, R. Canetti, HMAC: Keyed-Hashing for Message
Authentication, RFC 2104, Feb. 1997.
[RFC 2131] R. Droms, Dynamic Host Configuration Protocol, RFC 2131, Mar. 1997.
[RFC 2136] P. Vixie, S. Thomson, Y. Rekhter, J. Bound, Dynamic Updates in the Domain
Name System, RFC 2136, Apr. 1997.
[RFC 2153] W. Simpson, PPP Vendor Extensions, RFC 2153, May 1997.
[RFC 2205] R. Braden, Ed., L. Zhang, S. Berson, S. Herzog, S. Jamin, Resource
ReSerVation Protocol (RSVP)Version 1 Functional Specification, RFC 2205, Sept. 1997.
[RFC 2210] J. Wroclawski, The Use of RSVP with IETF Integrated Services, RFC 2210,
Sept. 1997.
[RFC 2211] J. Wroclawski, Specification of the Controlled-Load Network Element
Service, RFC 2211, Sept. 1997.
[RFC 2215] S. Shenker, J. Wroclawski, General Characterization Parameters for Integrated
Service Network Elements, RFC 2215, Sept. 1997.
[RFC 2326] H. Schulzrinne, A. Rao, R. Lanphier, Real Time Streaming Protocol (RTSP),
RFC 2326, Apr. 1998.
[RFC 2328] J. Moy, OSPF Version 2, RFC 2328, Apr. 1998.
[RFC 2420] H. Kummert, The PPP Triple-DES Encryption Protocol (3DESE), RFC 2420,
Sept. 1998.
[RFC 2453] G. Malkin, RIP Version 2, RFC 2453, Nov. 1998.
[RFC 2460] S. Deering, R. Hinden, Internet Protocol, Version 6 (IPv6) Specification, RFC
2460, Dec. 1998.
[RFC 2475] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, W. Weiss, An
Architecture for Differentiated Services, RFC 2475, Dec. 1998.
[RFC 2578] K. McCloghrie, D. Perkins, J. Schoenwaelder, Structure of Management
Information Version 2 (SMIv2), RFC 2578, Apr. 1999.
[RFC 2579] K. McCloghrie, D. Perkins, J. Schoenwaelder, Textual Conventions for
SMIv2, RFC 2579, Apr. 1999.
[RFC 2580] K. McCloghrie, D. Perkins, J. Schoenwaelder, Conformance Statements for
SMIv2, RFC 2580, Apr. 1999.
[RFC 2597] J. Heinanen, F. Baker, W. Weiss, J. Wroclawski, Assured Forwarding PHB
Group, RFC 2597, June 1999.
[RFC 2616] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, L. Masinter, P. Leach, T. Berners-
Lee, R. Fielding, Hypertext Transfer ProtocolHTTP/1.1, RFC 2616, June 1999.
812 REFERENCES
[RFC 2663] P. Srisuresh, M. Holdrege, IP Network Address Translator (NAT) Terminology
and Considerations, RFC 2663.
[RFC 2702] D. Awduche, J. Malcolm, J. Agogbua, M. ODell, J. McManus, Requirements
for Traffic Engineering Over MPLS, RFC 2702, Sept. 1999.
[RFC 2827] P. Ferguson, D. Senie, Network Ingress Filtering: Defeating Denial of Service
Attacks which Employ IP Source Address Spoofing, RFC 2827, May 2000.
[RFC 2865] C. Rigney, S. Willens, A. Rubens, W. Simpson, Remote Authentication Dial In
User Service (RADIUS), RFC 2865, June 2000.
[RFC 2961] L. Berger, D. Gan, G. Swallow, P. Pan, F. Tommasi, S. Molendini, RSVP
Refresh Overhead Reduction Extensions, RFC 2961, Apr. 2001.
[RFC 3007] B. Wellington, Secure Domain Name System (DNS) Dynamic Update, RFC
3007, Nov. 2000.
[RFC 3022] P. Srisuresh, K. Egevang, Traditional IP Network Address Translator
(Traditional NAT), RFC 3022, Jan. 2001.
[RFC 3022] P. Srisuresh, K. Egevang, Traditional IP Network Address Translator
(Traditional NAT), RFC 3022, Jan. 2001.
[RFC 3031] E. Rosen, A. Viswanathan, R. Callon, Multiprotocol Label Switching
Architecture, RFC 3031, Jan. 2001.
[RFC 3032] E. Rosen, D. Tappan, G. Fedorkow, Y. Rekhter, D. Farinacci, T. Li, A. Conta,
MPLS Label Stack Encoding, RFC 3032, Jan. 2001.
[RFC 3052] M. Eder, S. Nag, Service Management Architectures Issues and Review, RFC
3052, Jan. 2001.
[RFC 3139] L. Sanchez, K. McCloghrie, J. Saperia, Requirements for Configuration
Management of IP-Based Networks, RFC 3139, June 2001.
[RFC 3168] K. Ramakrishnan, S. Floyd, D. Black, The Addition of Explicit Congestion
Notification (ECN) to IP, RFC 3168, Sept. 2001.
[RFC 3209] D. Awduche, L. Berger, D. Gan, T. Li, V. Srinivasan, G. Swallow, RSVP-TE:
Extensions to RSVP for LSP Tunnels, RFC 3209, Dec. 2001.
[RFC 3221] G. Huston, Commentary on Inter-Domain Routing in the Internet, RFC 3221,
Dec. 2001.
[RFC 3232] J. Reynolds, Assigned Numbers: RFC 1700 is Replaced by an On-line
Database, RFC 3232, Jan. 2002.
[RFC 3246] B. Davie, A. Charny, J.C.R. Bennet, K. Benson, J.Y. Le Boudec, W. Courtney,
S. Davari, V. Firoiu, D. Stiliadis, An Expedited Forwarding PHB (Per-Hop Behavior),
RFC 3246, Mar. 2002.
[RFC 3260] D. Grossman, New Terminology and Clarifications for Diffserv, RFC 3260,
Apr. 2002.
[RFC 3261] J. Rosenberg, H. Schulzrinne, G. Carmarillo, A. Johnston, J. Peterson, R.
Sparks, M. Handley, E. Schooler, SIP: Session Initiation Protocol, RFC 3261, July 2002.
[RFC 3272] J. Boyle, V. Gill, A. Hannan, D. Cooper, D. Awduche, B. Christian, W.S. Lai,
Overview and Principles of Internet Traffic Engineering, RFC 3272, May 2002.
[RFC 3286] L. Ong, J. Yoakum, An Introduction to the Stream Control Transmission
Protocol (SCTP), RFC 3286, May 2002.
REFERENCES 813
[RFC 3346] J. Boyle, V. Gill, A. Hannan, D. Cooper, D. Awduche, B. Christian, W. S. Lai,
Applicability Statement for Traffic Engineering with MPLS, RFC 3346, Aug. 2002.
[RFC 3376] B. Cain, S. Deering, I. Kouvelas, B. Fenner, A. Thyagarajan, Internet Group
Management Protocol, Version 3, RFC 3376, Oct. 2002.
[RFC 3390] M. Allman, S. Floyd, C. Partridge, Increasing TCPs Initial Window, RFC
3390, Oct. 2002.
[RFC 3410] J. Case, R. Mundy, D. Partain, Introduction and Applicability Statements for
Internet Standard Management Framework, RFC 3410, Dec. 2002.
[RFC 3411] D. Harrington, R. Presuhn, B. Wijnen, An Architecture for Describing
Simple Network Management Protocol (SNMP) Management Frameworks, RFC 3411,
Dec. 2002.
[RFC 3414] U. Blumenthal and B. Wijnen, User-based Security Model (USM) for
Version 3 of the Simple Network Management Protocol (SNMPv3), RFC 3414,
December 2002.
[RFC 3415] B. Wijnen, R. Presuhn, K. McCloghrie, View-based Access Control Model
(VACM) for the Simple Network Management Protocol (SNMP), RFC 3415, Dec. 2002.
[RFC 3416] R. Presuhn, J. Case, K. McCloghrie, M. Rose, S. Waldbusser, Version 2 of the
Protocol Operations for the Simple Network Management Protocol (SNMP), Dec. 2002.
[RFC 3439] R. Bush and D. Meyer, Some internet architectural guidelines and philosophy,
RFC 3439, Dec. 2003.
[RFC 3447] J. Jonsson, B. Kaliski, Public-Key Cryptography Standards (PKCS) #1: RSA
Cryptography Specifications Version 2.1, RFC 3447, Feb. 2003.
[RFC 3468] L. Andersson, G. Swallow, The Multiprotocol Label Switching (MPLS)
Working Group Decision on MPLS Signaling Protocols, RFC 3468, Feb. 2003.
[RFC 3469] V. Sharma, Ed., F. Hellstrand, Ed, Framework for Multi-Protocol Label
Switching (MPLS)-based Recovery, RFC 3469, Feb. 2003. ftp://ftp.rfc-editor.org/in-
notes/rfc3469.txt
[RFC 3501] M. Crispin, Internet Message Access ProtocolVersion 4rev1, RFC 3501,
Mar. 2003.
[RFC 3550] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, RTP: A Transport
Protocol for Real-Time Applications, RFC 3550, July 2003.
[RFC 3569] S. Bhattacharyya (ed.), An Overview of Source-Specific Multicast (SSM),
RFC 3569, July 2003.
[RFC 3588] P. Calhoun, J. Loughney, E. Guttman, G. Zorn, J. Arkko, Diameter Base
Protocol, RFC 3588, Sept. 2003.
[RFC 3618] B. Fenner, D. Meyer, Ed., Multicast Source Discovery Protocol (MSDP),
RFC 3618, Oct. 2003.
[RFC 3649] S. Floyd, High Speed TCP for Large Congestion Windows, RFC 3649, Dec. 2003.
[RFC 3748] B. Aboba, L. Blunk, J. Vollbrecht, J. Carlson, H. Levkowetz, Ed., Extensible
Authentication Protocol (EAP), RFC 3748, June 2004.
[RFC 3782] S. Floyd, T. Henderson, A. Gurtov, The NewReno Modification to TCPs Fast
Recovery Algorithm, RFC 3782, Apr. 2004.
814 REFERENCES
[RFC 3973] A. Adams, J. Nicholas, W. Siadak, Protocol Independent MulticastDense
Mode (PIM-DM): Protocol Specification (Revised), RFC 3973, Jan. 2005.
[RFC 4022] R. Raghunarayan, Ed., Management Information Base for the Transmission
Control Protocol (TCP), RFC 4022, Mar. 2005.
[RFC 4113] B. Fenner, J. Flick, Management Information Base for the User Datagram
Protocol (UDP), RFC 4113, June 2005.
[RFC 4213] E. Nordmark, R. Gilligan, Basic Transition Mechanisms for IPv6 Hosts and
Routers, RFC 4213, Oct. 2005.
[RFC 4271] Y. Rekhter, T. Li, S. Hares, Ed., A Border Gateway Protocol 4 (BGP-4), RFC
4271, Jan. 2006.
[RFC 4272] S. Murphy, BGP Security Vulnerabilities Analysis, RFC 4274, Jan. 2006.
[RFC 4274] Meyer, D. and K. Patel, BGP-4 Protocol Analysis, RFC 4274, January
2006.
[RFC 4291] R. Hinden, S. Deering, IP Version 6 Addressing Architecture, RFC 4291,
February 2006.
[RFC 4293] S. Routhier, Ed. Management Information Base for the Internet Protocol (IP),
RFC 4293, Apr. 2006.
[RFC 4301] S. Kent, K. Seo, Security Architecture for the Internet Protocol, RFC 4301,
Dec. 2005.
[RFC 4302] S. Kent, IP Authentication Header, RFC 4302, Dec. 2005.
[RFC 4303] S. Kent, IP Encapsulating Security Payload (ESP), RFC 4303, Dec. 2005.
[RFC 4305] D. Eastlake, Cryptographic Algorithm Implementation Requirements for
Encapsulating Security Payload (ESP) and Authentication Header (AH), RFC 4305,
Dec. 2005.
[RFC 4340] E. Kohler, M. Handley, S. Floyd, Datagram Congestion Control Protocol
(DCCP), RFC 4340, Mar. 2006.
[RFC 4443] A. Conta, S. Deering, M. Gupta, Ed., Internet Control Message Protocol
(ICMPv6) for the Internet Protocol Version 6 (IPv6) Specification, RFC 4443, Mar.
2006.
[RFC 4346] T. Dierks, E. Rescorla, The Transport Layer Security (TLS) Protocol Version
1.1, RFC 4346, Apr. 2006.
[RFC 4502] S. Waldbusser, Remote Network Monitoring Management Information Base
Version 2, RFC 4502, May 2006.
[RFC 4514] K. Zeilenga, Ed., Lightweight Directory Access Protocol (LDAP): String
Representation of Distinguished Names, RFC 4514, June 2006.
[RFC 4601] B. Fenner, M. Handley, H. Holbrook, I. Kouvelas, Protocol Independent
MulticastSparse Mode (PIM-SM): Protocol Specification (Revised), RFC 4601, Aug. 2006.
[RFC 4607] H. Holbrook, B. Cain, Source-Specific Multicast for IP, RFC 4607, Aug. 2006.
[RFC 4611] M. McBride, J. Meylor, D. Meyer, Multicast Source Discovery Protocol
(MSDP) Deployment Scenarios, RFC 4611, Aug. 2006.
[RFC 4632] V. Fuller, T. Li, Classless Inter-domain Routing (CIDR): The Internet Address
Assignment and Aggregation Plan, RFC 4632, Aug. 2006.
REFERENCES 815
[RFC 4960] R. Stewart, ed., Stream Control Transmission Protocol, RFC 4960, Sept. 2007.
[RFC 4987] W. Eddy, TCP SYN Flooding Attacks and Common Mitigations, RFC 4987,
Aug. 2007.
[RFC 5000] RFC editor, Internet Official Protocol Standards, RFC 5000, May 2008.
[RFC 5109] A. Li (ed.), RTP Payload Format for Generic Forward Error Correction, RFC
5109, Dec. 2007.
[RFC 5110] P. Savola, Overview of the Internet Multicast Routing Architecture, RFC
5110, Jan. 2008.
[RFC 5216] D. Simon, B. Aboba, R. Hurst, The EAP-TLS Authentication Protocol, RFC
5216, Mar. 2008.
[RFC 5218] D. Thaler, B. Aboba, What Makes for a Successful Protocol?, RFC 5218, July
2008.
[RFC 5321] J. Klensin, Simple Mail Transfer Protocol, RFC 5321, Oct. 2008.
[RFC 5322] P. Resnick, Ed., Internet Message Format, RFC 5322, Oct. 2008.
[RFC 5348] S. Floyd, M. Handley, J. Padhye, J.Widmer, TCP Friendly Rate Control
(TFRC): Protocol Specification, RFC 5348, Sept. 2008.
[RFC 5411] J Rosenberg, A Hitchhikers Guide to the Session Initiation Protocol (SIP),
RFC 5411, Feb. 2009.
[RFC 5681] M. Allman, V. Paxson, E. Blanton, TCP Congestion Control, RFC 5681,
Sept. 2009.
[RFC 5944] C. Perkins, Ed., IP Mobility Support for IPv4, Revised, RFC 5944, November
2010.
[RFC 5996] C. Kaufman, P. Hoffman, Y. Nir, P. Eronen, Internet Key Exchange Protocol
Version 2 (IKEv2), RFC 5996, Sept. 2010.
[RFC 6071] S. Frankel, S. Krishnan, IP Security (IPsec) and Internet Key Exchange (IKE)
Document Roadmap, RFC 6071, Feb. 2011.
[RFC 6265] A Barth, HTTP State Management Mechanism, RFC 6265, Apr. 2011.
[RFC 6298] V. Paxson, M. Allman, J. Chu, M. Sargent, Computing TCPs Retransmission
Timer, RFC 6298, June 2011.
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Index
A
AAC (Advanced Audio Coding), 590
Abramson, Norman, 62, 454, 473
ABR ATM network service, 313
ABR (available bit-rate), 259
spare available bandwidth advantage,
267
Abstract Syntax Notation One. See ASN.1
access control and SNMPv3, 777778
access control lists, 734736
access delay, 113
access ISP, 3233
access networks, 1218
cable Internet access, 1415, 460461
dial-up access, 16
DSL (digital subscriber line), 13
enterprises, 1617
Ethernet, 1617
FTTH (fiber to the home), 15
home access, 13
link-layer switches, 4
LTE (Long-Term Evolution), 18, 533
optical distribution network, 15
satellite links, 16
2G, 3G, 4G (generation) wide-area
wireless networks, 18, 547554
wide-area wireless access, 18
wireless LANs, 17
access points. See AP
access routers, 492
Accounting Management, 759, 764
acknowledgments, 210, 234237
piggybacked, 237
TCP (Transmission Control Protocol),
235236, 244
Telnet, 237238
ACK (positive acknowledgments), 207,
240, 208, 212, 217, 235, 257
ACK receipt, 242
active optical networks. See AONs
active queue management algorithms. See
AQM algorithms
adapters, 463465
MAC addresses, 471
routers, 468
adaptive congestion control, 201
adaptive playout delay, 616618
adaptive streaming, 600601
adaptive HTTP streaming, 593
additive-increase, multiplicative-decrease.
See AIMD
address aggregation, 342
address indirection, 406
addressing
Internet, 331363
processes, 90
Address Resolution Protocol. See ARP
Address Supporting Organization of
ICANN, 345
ad hoc networks, 528
802.15.1 networks, 544
wireless hosts, 517
Adleman, Leonard, 684
Advanced Audio Coding. See AAC
Advanced Encryption Standard. See AES
Advanced Research Projects Agency. See
ARPA
AES (Advanced Encryption Standard),
680
agent advertisement, 566567
agent discovery, 565567
agent solicitation, 567
aging time, 478
AH (Authentication Header) protocol, 720
AIMD (additive-increase, multiplicative-
decrease) algorithm, 277278, 280
Akamai, 114, 133, 609, 603604,
35, 273
823
alias hostname, 140
ALOHAnet, 6263, 454, 473
ALOHA protocol, 6263, 452, 473, 511,
453, 63, 453455
alternating-bit protocols, 214
Amazon cloud, 608610
analog audio, 590
anchor foreign agent, 563564
anchor MSC, 574
Andreessen, Marc, 64
anomaly-based IDSs (intrusion detection
systems), 742
anonymity, 738
anycast, 356
AONs (active optical networks), 15
AP (access points), 517, 528530,
538540
Apache Web server, 99, 156
API (Application Programming Interface),
6, 89
application architecture, 8688
application gateways, 732, 736738
deep packet inspection, 739740
application-layer messages, 51, 5455,
186
application-layer protocols, 4950
DNS (domain name system), 131, 132
electronic mail, 98
FTP (File Transfer Protocol), 51
HTTP (HyperText Transfer Protocol),
51, 97
Internet e-mail application, 97
proprietary, 97
public domain, 97
security, 705
SMTP (Simple Mail Transfer
Protocol), 51, 97, 121
Telnet, 237
Application Programming Interface. See
API
AQM (active queue management) algo-
rithms, 329
area border routers, 389
ARPA (Advanced Research Projects
Agency), 61, 511
ARP (Address Resolution Protocol),
465468, 498
ARPAnet, 454, 473, 511
ARP messages 467, 498
ARP tables, 466467, 481
ARQ (Automatic Repeat reQuest) proto-
cols, 207208, 576577
ASN.1 (Abstract Syntax Notation One),
766, 770, 778782
ASN (autonomous system number),
394
AS-PATH attribute, 394
ASs (autonomous systems), 380383
association, 529531
assured forwarding PHB, 651
asynchronous transfer mode. See ATM
ATM ABR (available bit-rate) congestion
control, 266269, 313
ATM (asynchronous transfer mode), 259,
512
complexity and cost, 470
services, 312313
multiple service models, 312
Q2931b protocol, 654
audio
AAC (Advanced Audio Coding), 590
glitches, 591
human speech, 590
MP3 (MPEG 1 layer 3), 590
PCM (pulse code modulation), 590
properties, 590591
quantization, 590
removing jitter at receiver, 614618
authentication
cryptographic techniques, 675
end-point, 700705
802.11i, 729730
MD5, 389
networks, 700705
secret password, 701703
SNMPv3, 777
WEP (Wired Equivalent Privacy), 726
802.11 wireless LANs, 530
wireless station, 530
authentication key, 692693
824 INDEX
authentication protocol examples, 700705
authoritative DNS servers, 134137, 499
hostnames, 139
IP addresses, 142
names, 142
Automatic Repeat reQuest protocols. See
ARQ
autonomous system number. See ASN
autonomous systems. See ASs
available bit-rate. See ABR
average throughput, 44
B
backbone area, 389
bandwidth, 29, 281
guaranteed minimal, 311
link-level allocation, 639640
use-it-or-lose-it resource, 640
video, 588589, 594
bandwidth flooding, 57
bandwidth provisioning, 635
bandwidth-sensitive applications, 92
Baran, Paul, 60
base HTML file, 99
base station controller. See BSC
base stations, 516518, 528
handoff between, 572574
base station system. See BSS
base transceiver station. See BTS
Basic Encoding Rules. See BER
basic service set. See BSS
beacon frames, 529530
Bellman-Ford equation, 371372
Bellovin, Steven M., 753754
BER (Basic Encoding Rules), 780
BER (bit error rate), 520521
Berners-Lee, Tim, 64
best-effort delivery service, 190
best-effort networks, 634636
best-effort service, 190, 311312,
612614, 633634
BGP (Border Gateway Protocol),
390399, 498499
ASN (autonomous system number), 394
attributes, 394, 395
BGP peers, 391
BGP sessions, 393
complexity, 391, 393
DV (distance-vector) algorithm, 374
eBGP (external BGP) session, 393395
elimination rules for routes, 396
iBGP (internal BGP) session, 393395
inter-AS routing protocols, 390391,
393399
peers, 391
prefix bits, 342, 344, 393
route advertisement, 391, 396399
routes, 394395
route selection, 395396
routing policy, 397399
routing tables, 399
session, 393
TCP connections, 391, 393
bidirectional data transfer, 205206
binary exponential backoff algorithm,
457458
BIND (Berkeley Internet Name Domain),
131
bit error rate. See BER
bit-level error detection and correction,
438445
BITNET, 63
bits, 19
propagation delay, 3738
BitTorrent, 86, 149151
accepting connections from other
hosts, 352
chunks, 149
developing, 182
Kademlia DHT, 156
P2P (peer-to-peer) protocol, 145, 182
swarming data principles, 183
trading algorithm, 150
blades, 490
block ciphers, 678681
Bluetooth, 518, 544545
Boggs, David, 470, 473
Border Gateway Protocol. See BGP
border routers, 491
botnet, 56
INDEX 825
bottleneck link, 45, 279280
broadcasting (link layer), 433, 464
channels, 433, 446
channel propagation delay, 456
frame, 467
links, 445, 475
broadcasting (network layer)
broadcast routing algorithms, 405
broadcast storm, 401
flooding, 401403
N-way-unicast, 400401
sequence number, 401
spanning-tree broadcast, 403405
browsers, 101102
client processes, 88
HTTP requests directed to Web cache,
110111
BSC (base station controller), 550
BSS (base station system), 550
BSS (basic service set), 527528
MAC address, 539
mobility among, 541542
BTS (base transceiver station), 548549
buffered distributors, 475
buffering
packets, 218
streaming video, 592
buffers
packet loss, 25
switch output interfaces, 476
TCP connection, 233
burst size, 646
bursty traffic, 60
bus, switching via, 326
C
cable Internet access, 1415
DOCSIS protocol, 52, 460
link-layer protocols, 460461
cable modems, 15, 20, 460461
cable modem termination system. See
CMTS
CA (Certification Authority), 697699
certifying public keys, 708
call admission, 653654
call setup protocol, 654
canonical hostnames, 131132, 140
care-of-address. See COA
carrier sense multiple access protocol. See
CSMA protocol
carrier sense multiple access with colli-
sion detection. See CSMA/CD
carrier sensing, 454456
CBC (cipher-block chaining), 681682
CBR (constant bit rate) ATM network
service, 312
CDMA (code division multiple access),
522526
CDNs (Content Distribution Networks),
588, 603608
access networks of ISPs (Internet
Service Providers), 603604
cluster selection strategy, 606608
data centers, 604
delay and loss performance
measurements, 606
DNS intercept/redirect, 604605
IP anycast, 607608
Netflix, 608
operation, 604605
replicating content across clusters,
604
cell phones
growth of, 513
Internet access, 65, 546554
cellular networks
architecture, 547550
cells, 548549
handoffs in GSM, 572574
managing mobility, 570575
routing calls to cellular user, 571572
2G, 3G, 4G networks, 547554
Cerf, Vinton G., 62, 231, 431432, 511
CERT Coordination Center, 674
certificates, 698699, 713714
Certification Authority. See CA
channel numbers, 529
channel partitioning protocols, 447
CDMA (code division multiple access)
protocol, 449, 522526
826 INDEX
FDM (frequency-division multiplex-
ing), 448
TDM (time-division multiplexing),
448
channels
with bit errors and reliable data trans-
fer, 207212
802.11 wireless LANs, 529531
losing packets, 212215
propagation delay, 456
checksums, 203, 442
ACK/NAK packets, 210
calculation, 203
Internet, 203
poor cryptographic hash function, 690
TCP (Transmission Control Protocol),
334
UDP (User Datagram Protocol),
202204, 334
chipping rate, 522
choke packet, 266
chosen-plaintext attack, 678
chunks, 149
delaying playout, 615618
CI (congestion indication) bit, 268269
CIDR (Classless Interdomain Routing),
342, 344, 496
cipher-block chaining. See CBC
ciphertext, 675, 676
ciphertext-only attack, 677
circuit-switched networks
end-to-end connection, 28
multiplexing, 2830
versus packet switching, 3031
reserving time slots, 3031
sending packets, 28
telephone networks, 27
circuit-switched routing algorithm, 379
Cisco Systems, 65, 323, 732, 740
routers and switches, 325, 326
dominating network core, 323
Clark, Dave, 303, 585
classful addressing, 344
Classless Interdomain Routing. See CIDR
cleartext, 675
Clear to Send control frame. See CTS con-
trol frame
client application buffer, 597598
client buffering, 594595
clients, 10, 86, 88, 89, 156
initiating contact with server, 163
HTTP, 198
IP addresses, 162
matching received replies with sent
queries, 140
port number, 162
prefetching video, 597
receiving and processing packets from,
162
TCP socket creation, 163
Web caches as, 111
client-server application
developing, 157168
TCP, 157
UDP, 157
well-known port number, 157
client side socket interface, 99
client socket, 160, 163, 165
cloud applications and data center
networking, 490495
cluster selection strategy, 606608
CMTS (cable modem termination
system), 15, 460
CNAME record, 140
COA (care-of-address), 559, 565
coaxial cable, 20, 474
code division multiple access. See CDMA
collision detection, 454455
collisions, switch elimination, 479
Comcast, 601, 763764
commercial ISPs (Internet Service
Providers), 64
communication satellite, 21
computer networking
history of, 6066
computer networks, 2
conditional GET, 114116
confidentiality, 672673, 706707, 718,
720, 724
Configuration Management, 759, 764
INDEX 827
congestion
causes and costs, 259265
dropping packets, 265
large queuing delays, 261
transmission capacity wasted, 265
unneeded retransmissions by sender,
263
congestion avoidance, 274276
congestion control, 190, 240, 250, 596
ABR (available bit-rate) service, 259
AIMD (additive-increase,
multiplicative-decrease), 277278
approaches, 265266
ATM (asynchronous transfer mode)
networks, 259
end-to-end, 266, 269
network-assisted, 266, 269
principles, 259269
source quench message, 353
TCP (Transmission Control Protocol),
269272, 274283
UDP (User Datagram Protocol), 282
congestion indication bit. See CI bit
congestion window, 269270, 272277,
282
connection-establishment request, 195
connectionless service, 313
connectionless transport and UDP (User
Datagram Protocol), 198204
connection-oriented service, 94, 313314
connection-oriented transport and TCP
(Transmission Control Protocol),
230258
connection replay attack, 717
connections, persistent and non-
persistent, 100103
connection setup, 310
connection sockets and processes, 198
connection state, 231, 315
constant bit rate ATM network service.
See CBR ATM network service
container-based MDCs (modular data cen-
ters), 494
Content Distribution Networks. See CDNs
content provider networks, 3435
continuous playout, 591592
control connection, 117
controlled flooding, 401403
control plane software, 331
conversational applications, 592593
cookies, 108110
correspondent, 557, 561, 563
corrupted ACKs or NAKs, 209210
countdown timer, 214
coverage area, 515
CRC (cyclic redundancy check) codes,
443445
crossbar switch, 326
cryptographic hash functions, 689691
cryptographic techniques, 675
cryptography, 675688
ciphertext, 675676
cleartext, 675
confidentiality, 675
cryptographic hash functions,
689691
decryption algorithm, 676
encryption algorithm, 675676
keys, 676
plaintext, 675676
public key encryption, 683688
symmetric key cryptography,
676682
CSMA (carrier sense multiple access),
453456
carrier sensing, 455456
collisions, 455456
CSMA/CA (carrier sense multiple access
with collision avoidance), 526,
531537
CSMA/CD (carrier sense multiple access
with collision detection), 456459
collision detection, 456459
efficiency, 458459
Ethernet, 475
CSNET (computer science network), 63
CTS (Clear to Send) control frame,
535537
cumulative acknowledgments, 222, 236,
243
828 INDEX
customer ISPs, 34
customer-provider relationship, 33
cyclic redundancy check codes. See CRC
codes
D
DARPA (US Department of Defense
Advanced Research Projects
Agency), 62, 431
DASH (Dynamic Adaptive Streaming over
HTTP), 600601
databases, implementing in P2P network,
151156
data center networks, 490495
data centers, 86
border routers, 491
CDNs (Content Distribution
Networks), 604
container-based MDCs (modular data
centers), 494
costs, 490
data center networks, 490495
hierarchy of routers and switches,
492493
hosts, 490
interconnection architectures and net-
work protocols, 493
Internet services, 86
load balancing, 491492
servers, 11
data definition language, 765
data encryption algorithm and WEP (Wired
Equivalent Privacy), 726728
Data Encryption Standard. See DES
DATA frame, 535
datagram networks, 313, 317320
datagrams, 55, 189
detecting bit errors, 334
extending IP header, 335
fragmentation, 335338
Internet Protocol (IP) v4 format, 332338
labeling, 487
length, 334
moving between hosts, 5152
passed to transport layer, 337
reassembling in end systems, 336
sending off subnet, 468469
source and destination IP addresses,
334336
TOS (type of service) bits, 333
transport-layer segments, 242
TTL (time-to-live) field, 334
datagram service and network layer, 364
data integrity, 363, 712
data link layer, 50, 53
Data-Over-Cable Service Interface
Specifications. See DOCSIS
data plane and hardware, 331
DATA SMTP command, 123, 124
data transfer, 713
bidirectional, 205206
reliable, 91, 204230
SSL (Secure Sockets Layer), 714715
unidirectional, 205
unreliable, 206
VC (virtual-circuit) networks, 316
DDoS (distributed denial-of-service)
attacks, 5658, 143144
DECnet architecture, 266
decentralized routing algorithm, 366
decryption algorithm, 676, 283
deep packet inspection, 739741
Deering, Steve, 584
default name server, 136
default router, 364
delaying playout, 615618
delays
comparing transmission and propaga-
tion delays, 3839
end systems, 4344
end-to-end, 4244
nodal processing, 36
packetization, 44
packets, 36
packet-switched networks, 3539
processing, 3637
propagation, 3637
queuing, 3637, 3942
total nodal, 36
transmission, 3637
INDEX 829
delay-sensitive, 592
demilitarized zone. See DMZ
demultiplexing, 191198
connectionless, 193194
connection-oriented, 194197
denial-of-service attacks. See DoS
attacks
DES (Data Encryption Standard), 680
destination port numbers, 192, 234
destination router, 364
DHCP (Dynamic Host Configuration
Protocol), 345349
IP addresses dynamically assigned,
630
DHCP request message, 348, 495496
DHCP servers, 346347, 350, 495, 497
DHTs (distributed hash tables), 145,
151156
dial-up access, 16
DIAMETER protocol, 530, 730
differentiated service, 634
Diffie-Hellman algorithm, 683, 687, 725
Diffserv, 648652
DIFS (Distributed Inter-frame Space), 533
digital audio, 590
digital signatures, 707
compared with MAC (message authen-
tication code), 696697
hash functions, 695696
message integrity, 695
overheads of encryption and decryp-
tion, 695
PGP (Pretty Good Privacy), 710
public key certification, 697699
public-key cryptography, 693694
digital subscriber line. See DSL
digital subscriber line access multiplexer.
See DSLAM
Dijkstras least-cost path algorithm,
367368, 388, 390
OSPF (Open-Shortest Path First),
388
dimensioning best-effort networks,
634636
directory service, 97
direct routing, 563564
Direct Sequence Wideband CDMA. See
DS-WCDMA
distance vector, 372
distance-vector algorithm. See DV
algorithm
Distance-Vector Multicast Routing
Protocol. See DVMRP
distributed applications, 56
distributed attacks. See DDoS attacks
Distributed Inter-frame Space. See DIFS
DMZ (demilitarized zone), 741
DNS
DNS database, 142, 144
DNS (domain name system), 51, 63, 98,
130144, 497
additional delay to Internet applica-
tions, 131
application-layer protocols, 131132
attacks and vulnerabilities, 143144
caching, 138139
CDN requests, 604605
client-server paradigm, 132
DDoS attack against targeted host,
143144
distributed and hierarchical database,
131, 134138
host aliasing, 131132
hostname-to-IP-address translation
service, 133139
improving delay performance, 138
iterative queries, 137138
load distribution, 132133
obtaining presence in, 392
operational overview, 133139
querying and replaying messages, 133
query messages, 136, 140142, 497
recursive queries, 137138
reply messages, 136, 140142, 499
rotation, 132133
RRs (resource records), 139140,
498499
secure communication, 674
servers, 134
services provided by, 131133
830 INDEX
translating hostnames to IP addresses,
133139
underlying end-to-end transport
protocol, 132
UPDATE option, 144
DNS servers, 131135, 495
authoritative DNS servers, 134,
135136, 140
BIND (Berkeley Internet Name
Domain), 131
DDoS bandwidth-flooding attack, 143
discarding cached information, 139
distant centralized database, 134
hierarchy, 136
local DNS server, 136137
recursion, 140
root DNS servers, 134, 135
single point of failure, 133
TLD (top-level domain) DNS servers,
134
DOCSIS (Data-Over-Cable Service
Interface Specifications), 15, 52,
460
domain name system. See DNS
DoS (denial-of-service) attacks, 57, 735,
740
fragmentation and, 338
SYN flood attack, 253, 257
dotted-decimal notation, 339
dropping packets, 265
drop-tail, 329
DSLAM (digital subscriber line access
multiplexer), 1314
DSL (digital subscriber line), 1314, 20
DS-WCDMA (Direct Sequence Wideband
CDMA), 552
dual-stack approach, 359
duplicate ACKs, 211, 247248
duplicate data packets, 210, 213214
DV (distance-vector) algorithm, 366,
371380
DVMRP (Distance-Vector Multicast
Routing Protocol), 411412
Dynamic Adaptive Streaming over HTTP.
See DASH
Dynamic Host Configuration Protocol. See
DHCP
dynamic routing algorithm, 366
E
EAP (Extensible Authentication Protocol),
729730
eavesdropping, 673
eBGP (external BGP), 393395, 397
EC2, 66
edge router, 12
EFCI (explicit forward congestion
indication) bit, 268
802.15.1, 544545
802.11i, 728731
802.11n wireless network, 527
802.1Q frames, 484486
802.11 wireless LANs, 17, 515, 518,
526546, 548
address fields, 538540
APs (access points), 517, 528
architecture, 527531
association, 529531
authentication, 530, 728731
base station, 528
BSS (basic service set), 527528
channels, 529531
collision detection, 532
CSMA/CA (CSMA with collision
avoidance), 526, 531537
data rates, 526527
frequency band, 526
link-layer frames, 526
MAC addresses, 528
MAC protocols, 531537
reducing transmission rate, 526
transmitting frames, 532
elastic applications, 92
electromagnetic noise, 519
e-mail, 64, 86, 97, 118130
access protocols, 125126
application-layer protocols, 98
security, 705711
Web-based e-mail system, 120
encapsulation, 5355, 565567
INDEX 831
encapsulation/decapsulation and mobile
IP, 565
Encapsulation Security Payload
protocol. See ESP
encryption, 712
cryptography, 675688
SNMPv3, 777
end-point authentication, 59, 673
end systems, 2, 46, 10
API (Application Programming
Interface), 6
applications running on, 6
communication links, 4
connecting, 5
datagram reassembly, 336
delays, 4344
hosts, 10
IP addresses, 26
processes, 84, 88
TCP protocol, 231
transport-layer protocols, 188
end-to-end congestion control, 266
end-to-end delay, 4243, 613
Diffserv, 652
Traceroute program, 4243
end-to-end principle, 203
end-to-end throughput, 4447
enterprises and access networks, 1617
EPC (Evolved Packet Core), 553
ER (explicit rate) field, 269
error checking
link-layer protocols, 203
parity checks, 440442
transport layer, 203
error concealment, 620621
error detection
ARQ (Automatic Repeat reQuest) pro-
tocols, 208
checksumming methods, 203,
442443
CRC (cyclic redundancy check) codes,
443445
parity bits, 440442
two-dimensional parity scheme,
441442
ESP (Encapsulation Security Payload),
720, 723
ESTABLISHED state, 254
Estrin, Deborah, 303, 584585
Ethernet, 1617, 52, 63, 437, 445,
469476
binary exponential backoff algorithm,
457458
broadcast link, 475
carrier-sensing random access,
531532
collision-detection algorithm, 532
CSMA/CD protocol, 474475
frame format, 474
home networks, 17
link-layer and physical-layer specifica-
tion, 474
local area networking, 470
MSS (maximum segment size), 233
MTU (maximum transmission unit),
471
multiplexing network-layer protocols,
471472
physical-layer protocols, 52, 473474
standardized, 474
store-and-forward packet switching, 475
switch-based star topology, 475
switches, 17, 470, 475, 496
10BASE-2, 473474
10BASE-T, 473474
10GBASE-T, 474475
EWMA (exponential weighted moving
average), 240
expedited forwarding PHB, 651
explicit forward congestion indication bit.
See EFCI
explicit rate field. See ER (explicit rate)
field
exponential weighted moving average. See
EWMA
extended FSM (finite-state machine), 220,
223
Extensible Authentication Protocol. See
EAP
external BGP session. See eBGP session
832 INDEX
F
fairness
AIMD algorithm, 280
congestion-control mechanism, 280
parallel TCP connections, 282
TCP (Transmission Control Protocol),
279282
TDM (time-division multiplexing), 448
UDP (User Datagram Protocol), 282
Fault Management, 758759, 764
FCFS (first-come-first-served) scheduling,
329, 641
FDDI (fiber distributed data interface),
460, 470
FDM (frequency-division multiplexing),
2830, 448, 549
FDM/TDM systems, 549550
FEC (forward error correction), 442, 613,
618619
FHSS (frequency-hopping spread
spectrum), 544
fiber distributed data interface protocol. See
FDDI
fiber optics, 2021
100 Mbps Ethernet, 475
fiber to the home. See FTTH
FIFO (first-in-first-out), 637638, 641642
file sharing, 86
file transfer, 97, 116118
filtering, 476477
FIN bit, 235, 254
finite-state machine. See FSM
FIOS service, 1516
firewalls, 673, 730738
application gateways, 732, 736738
authorized traffic, 731
blocking packets, 355, 596
connection table, 732736
filtering, 733736
malicious packet attacks, 355
first-come-first-served scheduling.
See FCFS
first-in-first-out order. See FIFO
fixed-length labels, 487
fixed playout delay, 615
flag field, 235, 334336
flooding, 401405
flow control, 240
different from congestion control, 220
TCP, 250
flow (of packets), 311, 357
foreign address, 559
foreign agent, 557, 566
COA (care-of-address), 567
mobile IP, 565
receiving and decapsulating datagram,
561, 562
registration with home agent, 562,
568569
foreign networks, 557559
forwarding, 305, 308310, 321322, 322,
476477, 649
forwarding function, 320321
forwarding plane, 321
forwarding tables, 2627, 308309,
317318, 322323
adding entries, 396397
configuring, 308309, 364
datagram networks, 319
directing datagrams to foreign
network, 558
routing algorithms, 364
VC networks, 319
4G systems, 18, 553554
fourth-generation of wide-area wireless
networks. See 4G
fragmentation of datagrams, 334338
frames, 52, 433434, 436
frequency-division multiplexing. See FDM
frequency-hopping spread spectrum. See
FHSS
FSM (finite-state machine), 206207
extended, 220, 223
initial state, 207
FTP (File Transfer Protocol), 51, 86,
9798, 116118
FTTH (fiber to the home), 1516
full-duplex service, 232
full-table block ciphers, 679680
fully connected topology, 493494
INDEX 833
G
Gateway GPRS Support Nodes. See
GGSNs
gateway routers, 380381
import policy, 395
packet filtering, 732
prefixes, 393
GBN (Go-Back-N) protocol, 218224
Generalized Packet Radio Service. See
GPRS
generator, 443
geographically closest, 606
geostationary satellites, 21
GET method, 104105, 115
GGSNs (Gateway GPRS Support Nodes),
552
Gigabit Ethernet, 475
global routing algorithm, 365366
Global System for Mobile
Communications standards. See
GSM
global transit ISPs (Internet Service
Providers), 3233
GMSC (Gateway Mobile services
Switching Center), 570
Go-Back-N protocol. See GBN protocol
Google, 3425, 6466, 86, 114, 431,
603, 610
GPRS (Generalized Packet Radio
Service), 552
graphs, 364365
ground stations, 21
GSM (Global System for Mobile
Communications), 547, 549550,
570
anchor MSC, 574
BTS (base transceiver station),
548549
cells, 548549
encoding audio, 628
handoffs, 572574
home network, 570
home PLMN (home public land mobile
network), 570
indirect routing, 570
mobility, 576
routing cells to mobile user, 571572
visited network, 570
guaranteed delivery, 311
guided media, 19
H
handoff, 517518
GSM, 572574
handshake, 94, 231232, 252253, 713,
716
hard guarantee, 634
hardware data plane, 331
hardware-implemented protocols, 9
hash functions, 152, 707, 714
digital signatures, 695696
HDLC (high-level data link control), 445
header checksum, 334
header fields, 55
head-of-line blocking. See HOL blocking
HFC (hybrid fiber coax), 14
hidden terminals, 521, 534537
hierarchical routing, 379383
higher-layer protocols and wireless net-
works and mobility, 575577
high-level data link control. See HDLC
HLR (home location register), 570, 572
HMAC standard, 692693
HOL (head-of-the-line) blocking, 330331
home agents, 557562, 565566, 568569
home location register. See HLR
home MSC, 574
home networks, 13, 557
Ethernet, 17
GMSC (Gateway Mobile services
Switching Center), 570
GSM (Global System for Mobile
Communications), 570
HLR (home location register), 570
home agent, 558559
IP addresses, 349352
mobile nodes, 559
PLMN (home public land mobile net-
work), 570
WiFi, 17
834 INDEX
hostnames, 130
alias, 132
canonical, 131
translating to IP addresses, 131, 132,
133139
hosts, 2, 4, 1011
aliasing, 131132, 140
ARP table, 466
assigning IP addresses to, 345349
canonical hostname, 140
changing base station, 517518
connected into network, 338
data centers, 490
default router, 364
DHCP discovery message, 530
dial-up modem connection, 486
hostnames, 130131
interfaces, 338
IP addresses, 90, 130131, 190,
465
link-layer addresses, 462463
load balancer, 491492
local DNS server, 136
MAC addresses, 465
monitoring, 756
moving datagrams between, 5152
network-layer addresses, 462, 465
network-layer protocols, 471472
process-to-process delivery service,
191198
storing routing information, 380
wireless, 514
hot-potato routing, 382
HSP (High Speed Packet Access),
552
HTML, 64, 100101
HTTP (HyperText Transfer Protocol), 51,
97100, 127
client program, 98
conditional GET, 114116
cookies, 108110
GET request, 103105, 499500, 596,
601, 610
If-Modified-Since: header line, 115
message format, 98, 103108
non-persistent connections,
100103
persistent connections, 103, 124
pull protocol, 124
POST method, 104105
PUT method, 105
response message, 105108, 124, 500,
596
server program, 98, 100
SMTP comparison with, 124
stateless, 108
SSL security, 712
stateless protocols, 100, 117
TCP and, 99100, 116, 200
Web caching, 110114
HTTP streaming, 593, 597600, 611
prefetching video, 596597
hubs, 470
Hulu, streaming stored video, 591
human protocols, 78
human speech, 590
hybrid fiber coax. See HFC
hypertext, 64
HyperText Transfer Protocol. See HTTP
I
iBGP (internal BGP), 393395, 397
IBM SNA architecture, 62, 266
ICANN (Internet Corporation for
Assigned Names and Numbers),
142, 345
ICMP (Internet Control Message
Protocol), 306, 353355, 359
datagram, 258
messages, 353354
IPv6, 359
IDEA, 710
identification field (IP), 336
IDSs (intrusion detection systems), 355,
673, 739742
IEEE 802.3 CSMA/CD (Ethernet) work-
ing group, 474
IEEE 802 LAN/MAN Standards
Committee, 5
IEEE managing MAC address space, 464
INDEX 835
IETF (Internet Engineering Task Force), 5,
668
Address Lifetime Expectations work-
ing group, 356
standards for CAs, 699
If-Modified-Since: header line, 115
IGMP (Internet Group Management
Protocol), 359, 407409
IKE (Internet Key Exchange), 725
IMAP (Internet Mail Access Protocol),
127, 129
IMAP server, 129
import policy, 395
in-band, 117
indirect routing,
GSM (Global System for Mobile
Communications), 570
mobile IP standard, 562
mobile node, 559562
triangle routing problem, 563
infrastructure wireless networks, 517, 528
in-order packet delivery, 311
input ports, 320
packet queues, 327331
processing, 322324
switching to output ports, 324326
input processing, 322324
instantaneous throughput, 44
instant messaging, 65, 83, 632
Intel 8254x controller, 437
interactivity, 591
inter-AS routing, 390391, 393399
inter-AS routing protocols, 382, 398
BGP (Border Gateway Protocol),
390391, 393399
interconnection network, 326
interfaces
assigning IP addresses to, 345349
IP addresses, 338339
routers, 468
interior gateway protocols, 384
interleaving, 618620
Intermediate-System-to-Intermediate-
System routing algorithm. See IS-IS
routing algorithm
internal BGP session. See iBGP session
International Organization for
Standardization. See ISO
Internet
access networks, 1218
address assignment strategy, 342
addressing, 331363
best-effort service model, 190,
311313, 612614, 636
cellular access, 546554
commercialization, 64
connecting end systems, 5
connectivity, 392
delivering data, 6
difficulty making changes to, 303
distributed applications, 56
e-mail, 64, 118130, 705711
end systems, 2, 4, 10
flag day, 359
forwarding, 331363
global traffic, 4
history of, 6066
hosts, 2, 4
inter-AS routing, 390391, 393399
inter-domain protocol, 498
intra-AS routing, 384390
ISPs (Internet Service Providers), 4,
3235
IXPs (Internet exchange points), 34
layered architecture, 4753
link-layer overview, 434438
link-layer switches, 4, 476482
mobile devices, 550552
multicast routing, 411412
network layer, 332
network of networks, 35, 3235
obtaining presence on, 392393
packet forwarding, 2627
protocols, 5, 9
protocol stack, 50
root DNS servers, 135
routers, 4
routing protocols, 27, 383399
service classes, 636640
service provider private networks, 66
services description, 57
standards, 5
836 INDEX
3G and 4G cellular networks, 65,
547554
TLD (top-level domain) servers, 135
transport layer overview, 189191
transport services, 9396
Web, 64, 98100, 111
wireless access, 1718
Internet API, 6, 156158
Internet applications, 83, 100
application-layer protocols, 9697
Internet checksum, 203, 334, 442
Internet commerce, 64, 86
cloud, 66
Internet Control Message Protocol. See
ICMP
Internet Corporation for Assigned Names
and Numbers. See ICANN
Internet Engineering Task Force. See IETF
Internet exchange points. See IXPs
Internet Group Management Protocol. See
IGMP
Internet hosts. See hosts
Internet Key Exchange. See IKE
Internet Mail Access Protocol. See IMAP
Internet phone application, 612614,
618621
Internet Protocol. See IP
Internet registrar, 392
Internet routing protocols, 353
Internet Service Providers. See ISPs
Internet-Standard Management
Framework, 764778
Internet telephony, 87, 612623
Skype, 621623
internetting, 62
Internet transport protocols, 95
intra-AS (autonomous system) routing
protocols, 380, 381, 397398
intra-AS routing, 388390
intra-domain routing and DNS servers,
498499
intrusion detection, 758
intrusion prevention system. See IPS
intrusion detection systems. See IDSs
IP addresses, 26, 90, 130131
adapters, 465
address aggregation, 342
allocating, 345
authoritative DNS server, 142
block, 345
CIDR (Classless Interdomain Routing),
342, 344
Classful addressing, 344
destination, 334335
distinguishing among devices, 344
DNS, 130144
dotted-decimal notation, 339
globally unique, 339
hierarchical structure, 130, 464
home networks, 349352
increasing size of, 356
interfaces, 338339
IP broadcast, 344
lease time, 347
local DNS server, 136
mobility, 556559
obtaining, 495
prefix, 342, 344
private addresses, 349350
range, 392
routers, 465, 468
setting up call to known, 627629
source, 334335
subnets, 340, 342, 344
translating hostnames to, 131139
IP anycast, 607608
IP broadcast address 344, 347
IP datagrams, 233, 496
attacks and, 355
encryption of payloads, 363
Ethernet frames, 471
fragmentation, 335338
MTU (maximum transmission unit), 335
security, 718
transport-layer protocols, 334
IP header, 202, 335
IP-in-IP encapsulation, 567
IP (Internet Protocol), 5, 5152, 190, 306,
331363, 353, 387
best-effort delivery service, 190,
311312, 612614, 633634
confining Internets development, 512
INDEX 837
IP (Internet Protocol) (continued)
datagrams, 242, 332338
de facto standard for internetworking,
512
Internet addressing and forwarding,
331363
multicasting, 593
network layer, 305307, 332
pervasiveness, 431
routers, 53
security, 362363
separation from TCP, 62
IP multicast, 412
IPng (Next Generation IP), 356
IP protocol version 4. See IPv4
IP protocol version 6. See IPv6
IPsec, 362, 72724
IPS (intrusion prevention system), 355, 740
IP spoofing, 5960, 701
IP subnet, 541542
IP tunnels, 303
IPTV, 87
IPv4 addresses, 334, 356
IPv6 addresses, 356, 512
IPv4 addressing, 338352
IPv4 datagrams, 332335
format, 306
IPv6 datagrams, 356359
IPv4 header, 636637
IPv6 header, 359
IPv4 (IP protocol version 4), 160, 165,
331, 336
IPsec, 362
transitioning to IPv6, 359362
IPv6 (IP protocol version 6), 306, 332,
356362
dual-stack approach, 360
flow labeling and priority, 357
IP addresses, 351, 356
transitioning from IPv4, 359362
tunneling, 360361
IPv6/IPv4 nodes, 359360
IS-IS (Intermediate-System-to-
Intermediate-System), 405, 498
IS-IS protocol, 384
ISO (International Organization for
Standardization), 52, 758, 770
ISPs (Internet Service Providers), 3235
access networks, 1218
ASs (autonomous systems), 383
customer-provider relationship, 33
end-to-end service, 651652
global transit, 32
low-tier and upper-tier, 5
multi-homing, 3334
naming and addressing conventions, 5
obtaining set of addresses from, 345
peer, 34
peering agreements, 399
iterative queries, 137138
ITU-T (International Telecommunication
Union), 699, 770
IXPs (Internet exchange points), 33, 34
J
Jacobson, Van, 302, 303, 584
Java and client-server programming, 157
jitter, 614618
K
Kademlia DHT, 156
Kahn, Robert, 62, 231, 431432, 511
KanKan, 87, 588, 611612
key derivation, 713714
key management, 726
Kleinrock, Leonard, 6062, 8082, 431, 511
known-plaintext attack, 677
L
label-switched routers, 488
Lam, Simon S., 511512
LANs (local area networks), 16
broadcast address, 464
configured hierarchically, 482483
Ethernet, 470
hub-based star topology, 470
MAC address, 463
switches, 17, 470, 475, 483, 496
VLANs (virtual local area networks),
482486
838 INDEX
Last-Modified: header line, 106, 115
layered architecture, 4748, 5153
layer-2 packet switch, 480
layer-4 switch, 492
LDNS (Local DNS Server), 136137,
605606
leaky bucket mechanism, 646648
least-cost path, 365, 367368
Bellman-Ford equation, 371372
LEO (low-earth orbiting) satellites, 2122
Limelight, 35, 114, 604, 609
limited-scope flooding, 405
link-cost changes and DV (distance-
vector) algorithm, 376377
link-layer acknowledgments, 532
link-layer addressing, 462469
link-layer frames, 55, 434, 436, 438445,
461462
link-layer protocols, 52
cable Internet access, 460461
error detection and correction, 437
services 436437
link layer, 52, 433436
bit-level error detection and correction,
438445
broadcast channels, 433
CRC (cyclic redundancy check) codes,
443445
error-detection and-correction tech-
niques, 438445
IEEE protocols, 444
implementation, 437438
link-layer frame, 434
multiple access protocols, 445461
networks as, 486490
point-to-point communication link, 434
services, 436437
switches, 461
wireless links, 1718, 519522
link-layer switches, 4, 22, 5354, 310,
476482
link rates, 515
links, 434
broadcast, 445
heterogeneous, 479
MPLS header format, 488
point-to-point, 445
transmission rates, 4
link-scheduling disciplines
FIFO (first-in-first-out), 641642
priority queuing, 642643
round robin queuing discipline, 643644
WFQ (weighted fair queuing), 644645
work-conserving round robin disci-
pline, 644
link-state advertisements. See LSAs
link-state broadcast, 366367
link-state messages, 689
link-state protocols, 388, 400
link-state routing algorithms, 366371
link virtualization, 486490
link weights, 390
load balancer, 491492
load distribution, 132133
load-insensitive routing algorithm, 366
local area networks. See LANs
Local DNS Server. See LDNS
local ISP, 392
logical communication between processes,
186
longest prefix matching rule, 318319
Long-Term Evolution. See LTE
loss recovery schemes, 618
loss-tolerant, 91, 592593
lost packets, 262263
low-earth orbiting satellites. See LEO
satellites
LSAs (link-state advertisements), 405
LTE (Long-Term Evolution), 18, 553554
M
MAC addresses, 463465, 497
adapters, 464465, 465, 471
AP (access point), 528
BSS, 539
802.11 wireless LAN, 528
flat structure, 464
no two adapters have same, 464
permanent, 463464
switches, 477, 480
INDEX 839
MAC-based VLANs (virtual local area
networks), 486
MAC (message authentication code),
691693, 777
compared with digital signatures,
696697
MAC (multiple access protocols), 436,
464, 531
802.11 wireless LANs, 531537
mail access protocols, 125130
mailbox, 120121
mail clients, 97, 125
mail servers, 97, 119121, 125126
main-in-the-middle attack and DNS
(domain name system), 143
malicious packet attacks, 355
malware, 5657
managed device, 761
managed objects, 761
Management Information Base. See MIB
managing entity, 761
MANETs (mobile ad hoc networks), 518
manifest file, 601, 610
MAP message, 460
Master Key. See MK
Master Secret. See MS
maximum segment size. See MSS
maximum transmission unit. See MTU
MBone multicast network, 411
MCR (minimum cell transmission rate), 313
MD5, 710
authentication, 389
MDCs (modular data centers), 494
MD5 hash algorithm, 690
message authentication code. See MAC
message digests, 707
message integrity, 688693, 706
cryptographic techniques, 675
digital signatures, 695
secure e-mail system, 707708
message queue, 121
messages, 51
application-layer, 51
authenticating, 689
breaking into shorter segments, 51
confidentiality, 672673
eavesdropping, 673
encrypted, 672
HTTP format, 103108
integrity, 673
segmentation, 77
semantics of fields, 97
syntax, 96
transmitting and receiving, 78
Metcalfe, Bob, 454, 470, 473
metering function, 650
method field, 104
MIB (Management Information Base),
761762, 765, 770
MIB modules, 765, 770772
MIB objects, 765
Microsoft, 6466, 66, 114
middleboxes, 303
MIMO (multiple-input, multiple output)
antennas, 553
minimum cell transmission rate. See MCR
minimum spanning tree, 403
Minitel project, 6364
MK (Master Key), 730
mobile ad hoc networks. See MANETs
mobile devices, 81
Internet, 550552
power management, 543544
mobile IP, 349, 563569, 576, 568569
mobile IP standard, 562
mobile nodes, 554564
COA (care-of-address), 559
direct routing, 563564
foreign address, 559
foreign network, 559
home network, 559
indirect routing, 559562
location protocol, 563
permanent address, 559
permanent home, 557
registering with foreign agent, 566567
routing to, 559564
sending datagrams to correspondent, 561
mobile-node-to-foreign-agent protocol,
562
840 INDEX
mobile phones and wireless LAN base sta-
tions, 548
mobile station roaming number. See
MSRN
mobile switching center. See MSC
mobility, 513514
addressing, 556559
cellular network management, 570577
GSM (Global System for Mobile
Communications), 576
in IP subnet, 541542
management, 555564
mobile IP, 576
network-layer functionality required,
561562
routing to mobile node, 559564
scalability, 558
wireless networks, 575577
mobility-related services, 513
modems, 15
modular arithmetic, 685
modular data centers. See MDCs
modulo arithmetic, 687688
MPEG, 624
MPLS (Multiprotocol Label Switching),
487490
MPLS paths, 303
MP3 (MPEG 1 layer 3), 590
MSC (mobile switching center), 550
MSDP (Multicast Source Discovery
Protocol), 412
MS (Master Secret), 714, 716
MSRN (mobile station roaming
number), 571572
MSS (maximum segment size), 232234
MTU (maximum transmission unit),
232233
Ethernet, 471
IP datagrams, 335
multicast routing, 405412
multicast addresses, 356
multicast group, 406
multicast packets, 405412
multicast protocols, 362
multicast routing, 399, 407412
Multicast Source Discovery Protocol. See
MSDP
multicast trees and RTP packets, 624625
multi-homed stub network, 397
multi-homing, 33
multi-hop wireless networks, 518
multihop paths, 263265
multimedia
network supported for, 632655
streaming stored video, 593612
system-level approach for delivering,
633
VoIP (Voice-over-IP), 612623
multimedia applications, 588
audio properties, 590591
bandwidth sensitive, 92
conversational voice, 587, 592593
improving quality, 635
Internet telephony, 592593
streaming live audio and video, 587,
593
streaming stored audio and video, 587,
591592
TCP (Transmission Control Protocol),
200
types, 591593
UDP (User Datagram Protocol),
200201, 282
video over IP, 592593
video properties, 588589
multipath propagation, 519
multi-player online games, 83
multiple access links, 445461
multiple access problem, 445
multiple access protocols, 445461
ALOHA protocol, 63
CDMA (code division multiple access),
449
channel partitioning protocols, 447448
characteristics, 447448
FDM (frequency-division multiplex-
ing), 448
random access protocols, 447
taking-turns protocols, 447, 459460
TDM (time-division multiplexing), 448
INDEX 841
multiple classes of service, 636640
multiple-input, multiple output antennas.
See MIMO antennas
multiplexing, 191198
circuit-switched networks, 2830
connectionless, 193194
connection-oriented, 194197
multiplexing/demultiplexing service and
transport layer, 198199
Multiprotocol Label Switching. See MPLS
multi-tier hierarchy, 33
MX record, 140, 142
N
NAKs (negative acknowledgments), 207,
208
named data, 585
name translation and SIP (Session
Initiation Protocol), 630632
Napster, 65
NAT (network address translation), 306,
322324, 349352
Skype, 622
NCP (network-control protocol), 62
negative acknowledgements. See NAKs
neighbors, 364
neighbor-to-neighbor communication,
372375
Netflix, 65, 83, 588, 608610
Netscape Communications Corporation,
64
network adapter, 437438
network address translation. See NAT
network applications, 83
application-layer protocols, 97
architectures, 8688
client program, 156
communication between client and
server, 156
creation, 156168
principles of, 8498
processes communicating, 88
proprietary, 156157
server program, 156
Web, 98116
network-assisted congestion control,
266269
network attacks, 5560
network cache servers, 106
network-control protocol. See NCP
network core
circuit switching, 2732
network of networks, 3235
packet switching, 2227
network dimensioning, 634635
network interface card. See NIC
network-layer addresses, 462, 465
network-layer components, 266
network-layer datagram, 55
network-layer multicast routing
algorithms, 408
network-layer packets, 5152
network-layer protocols
constrained by service model, 189
difficulty changing, 362
hosts supporting multiple, 471472
IP (Internet Protocol), 190
logical communication between hosts,
186, 188189
network layer, 5053, 189190
best-effort service, 311312
broadcast protocols, 405
complexity, 305
confidentiality, 718
connectionless service, 313
connection service, 313
connection setup, 310
datagram networks, 313
datagram service, 364
encapsulating transport-layer segment
into IP datagram, 199
extracting transport-layer segment from
datagram, 186
forwarding, 305, 308310, 315
guaranteed delivery, 311
host-to-host communication, 305
host-to-host services, 313
ICMP (Internet Control Message
Protocol), 353
in-order packet delivery, 311
842 INDEX
Internet, 332
Internet routing protocols, 353
IP protocol, 5152, 332, 353
mobility, 561562
passing segments to, 191198
path between sender and receiver, 315
process-to-process delivery service,
191198
reporting errors in datagrams, 332
reserving resources, 316
routing, 5152, 305306, 308310
security, 705, 718725
services offered by, 310313
transport layer relationship, 186189
VC (virtual-circuit) networks, 313315
network-layer service models, 319
network management
accounting management, 759
Comcast case study, 763764
configuration management, 759
definition, 759
fault management, 758759
host monitoring, 756
infrastructure, 760762
intrusion detection, 758
managed device, 761
managed objects, 761
managing entity, 761
MIB (Management Information Base),
761762
monitoring traffic, 756758
network management protocol, 762
performance management, 758
security management, 759
SLAs (Service Level Agreements), 758
standards, 762
network management agent, 762
network management protocol, 762
network mapping, 740
network of networks, 3235, 62
network prefix, 342
network protocols, 89
networks, 340
access networks, 1218
attacks, 5758
circuit-switched, 2730
cellular, 546554
components, 921
crossbar, 326
datagram, 317319
differentiated service, 634, 648652
dimensioning best-effort, 634636
DMZ (demilitarized zone), 741
foreign, 557
growth of, 62
interconnecting, 62
linking universities, 63
as link layer, 486490
links, 635
mobility, 513514
MPLS (Multiprotocol Label Switching)
networks, 487490
multiple classes of service,
636640
packet delay and loss, 3544, 635
packet-switched, 4, 2227
per-connection QoS (Quality-of-
Service) guarantees, 652655
physical media, 1822
private, 718
programs communicating on, 84
routes, 4
scheduling mechanisms, 640645
security, 5556, 671674
sockets, 8990
switched local area networks,
461486
VC (virtual-circuit), 314317
visited, 557
wireless LANS, 515, 518, 526546
network service models, 310313
NEXT-HOP attribute, 394395, 397
NIC (network Interface card), link layer
implementation, 437438
NI (no increase) bit, 268269
nmap port scanning tool, 196, 258
NOC (network operations center), 755
nodal processing, 36
no increase bit. See NI bit
nomadic computing, 81
INDEX 843
nonces, 704705, 716717
non-persistent connections, 100103, 198
nonpreemptive priority queuing
discipline, 643
non-real-time applications, 9293
nonrepudiation and cryptographic
techniques, 675
nslookup program, 141142
N-way-unicast, 400401
O
OBJECT IDENTIFIER data type, 766
objects, 99, 103
OC (Optical Carrier) standard link, 21
odd parity schemes, 440
offered load, 261
OLT (optical line terminator), 16
ONT (optical network terminator), 16
Open-Shortest Path First. See OSPF
Open Systems Interconnection model. See
OSI model
operational security, 673, 731742
Optical Carrier standard link. See OC
standard link
optical distribution network, 1516
optical line terminator. See OLT
optical network terminator. See ONT
options field, 235
origin authentication, 363
orthogonal frequency division multiplex-
ing. See OFDM
OSI (Open Systems Interconnection)
model, 5253
OSPF (Open-Shortest Path First),
366367, 384, 388390, 498
LSAs (link-state advertisements), 405
router authentication, 388389
OS vulnerability attacks, 740
out-of-band, 117
out-of-order segments, 236
output buffers, 25
output ports, 320326
packet queues, 327331
packet scheduler, 329
processing, 326
output queue, 25
overlapping fragments, 338
overlay network, 154, 486
P
packet classification, 648649
packet delay, 3544, 102, 635
packet-discarding policy, 641
packet filtering, 732, 737
packet forwarding, 2627
packet loss, 25, 4142, 91, 259, 613, 635
error concealment, 620621
FEC (forward error correction),
618619
interleaving, 618620
predicting imminent, 278
recovering from, 213, 618621
packet marking, 638
packet-radio networks, 62, 511
packet repetition, 621
packets, 4, 22
average rate, 645
bit errors, 207
buffering, 218
burst size, 646
controlled flooding, 401403
cumulative acknowledgment, 222
delays, 3637
delivering, 204205
destination, 35
destination IP address, 158
detecting loss, 212215
dropping, 41, 329
duplicate, 210, 213214
duplicate ACKs, 211
end-to-end delay, 612
FIFO (first-in-first-out), 641642
format of, 5
forwarding, 4, 308310, 321322, 649
header fields, 55
IP spoofing, 5960
jitter, 614
moving between nodes, 52
path, 35
payload fields, 55
844 INDEX
peak rate, 646
prefix destination address, 318
priority queuing, 642643
processing delays, 3637
queuing delays, 25, 37, 3942
round-robin queuing, 643644
round-trip delays, 4243
routing, 308310
RTT (round-trip time), 102103
same priority class, 642
senders source address, 158
sending, 24
sending multiple, 218
sequence numbers, 210, 218, 220
sockets, 158
source, 35
source address, 162
switching, 324326
tracing, 4243
transmitting, 213214
uncontrolled flooding, 401
VC number, 315
WFQ (weighted fair queuing),
644645
what to do when loss occurs, 212215
where queuing occurs, 327331
Packet Satellite, 511
packet scheduler, 329
packet sniffers, 5859, 78
packet-switched networks, 4, 25
ARPAnet, 61
comparing transmission and propaga-
tion delay, 3839
delays, 3539
end-to-end delays, 4244
packet loss, 4142
processing delays, 3637
propagation delays, 3738
queuing delays, 37, 3942
sending packets, 28
transmission delays, 37
packet switches, 4, 310
facilitating exchange of data, 6
link-layer switches, 22, 53, 310
output buffers, 25
routers, 22, 53, 310
store-and-forward transmission,
22, 24
packet switching, 3031
alternative to circuit switching, 60
forwarding tables, 2627
packet loss, 25
queuing delays, 25
queuing theory, 60
routing protocols, 2627
secure voice over military networks, 60
store-and-forward transmission,
22, 24
VC (virtual-circuit) approach, 267
packet-switching networks, 62
paging, 550
parity checks, 440442
passive optical networks. See PONs
passive spanning, 530
passwords, 703, 710
paths, 4, 365
least-cost, 365
multihop, 263265
payload fields, 55
PBXs (private branch exchanges), 627
PCM -law, 628629
PCM (pulse code modulation), 590
PDU and SNMP applications, 776777
peak rate, 646
peer, 34
peer churn and DHTs (distributed hash
tables), 155156
peering, 33
peers, 86, 144145
DHT, 155156
file sharing, 145151
torrent, 149
peer-to-peer applications. See P2P applica-
tions
per-connection QoS (Quality-of-Service)
guarantees, 634, 652655
per-connection throughput, 260
perfectly reliable channel, 206207
Performance Management, 758, 763764
per-hop behavior. See PHB
INDEX 845
permanent address, 559
persistent connections, 100103
persistent HTTP, 198
personal area networks
Bluetooth, 544545
Zigbee, 545546
PGP (Pretty Good Privacy), 678, 706,
709711
PHB (Per-hop behavior), 649651
physical address, 463
physical layer, 50, 5253
physical media, 4
coaxial cable, 20
costs, 19
fiber optics, 2021
guided media, 19
radio channels, 21
satellite radio channels, 2122
twisted-pair copper wire, 1920
unguided media, 19
piconet, 544
piggybacked acknowledgment, 237
PIM (Protocol-Independent Multicast),
411412, 584
ping program, 353
pipelined reliable data transfer protocols,
215218
pipelining, 218
persistent connections, 103
TCP (Transmission Control Protocol),
240
plaintext, 675
playback attacks, 703, 777
playout delay, 614
plug-and-play protocol, 346
plug-and-play switches, 479480
PMS (Pre-Master Secret), 716
points of presence. See PoPs
point-to-point, 232
point-to-point communication link, 434
point-to-point links, 436, 445
Point-to-Point Protocol. See PPP
poisoned reverse, 377378
poisoning attack, 143
policing disciplines, 645648
policing mechanisms, 640
polling protocols, 459
polls, 459
polyalphabetic encryption, 678
polynomial codes, 443
PONs (passive optical networks), 1516
POP3 (Post Office Protocol-Version 3),
127129
POP3 server, 127, 129
PoPs (points of presence), 33
POP3 user agent, 128
port-based VLAN, 483484
port numbers, 90, 158
addressing processes, 351
destination, 234
protocols, 90
source, 234
Web servers, 197198
well-known, 192, 196
port-based VLAN, 483484
port scanners, 196
port scans, 196, 740
postive acknowledgment. See ACK
POST method, 104105
Post Office Protocol-Version 3. See POP3
power management, 543544
P2P (peer to peer)
applications, 9798
architecture, 8688, 144148
BitTorrent, 149151
connection reversal, 352
DHTs (distributed hash tables), 145,
151156
file distribution, 83, 88, 45151
NAT, 351352
Skype, 621622
video streaming applications, 592
PPP (point-to-point protocol), 434, 445
PPstream, 87
PPTV and P2P delivery, 611
prefetching, 592
video, 596597
prefixes, 342, 344, 393
awareness of, 396397
BGP attributes, 394
846 INDEX
forwarding table, 396397
gateway routers, 393
Pre-Master Secret. See PMS
prerecorded video, 591
presentation layer, 53
presentation service, 780
Pretty Good Privacy. See PGP
priority queuing, 642643
privacy
cookies, 108
proxy servers, 738
QQ, 623
Skype, 623
SSL (Secure Sockets Layer), 738
Web sites, 738
private branch exchanges. See PBXs
private CDNs (Content Distribution
Networks), 603
private key, 684685, 693, 708
passwords, 710
private networks, 66, 718
processes, 8890
communicating by sending messages to
sockets, 157
communicating using UDP sockets,
158
connection sockets, 198
handshaking, 231
logical communication between, 186
sockets, 191
processing delays, 3637
programming, event-based, 223
propagation delays, 24, 3639, 456
proprietary network applications,
156157
Protocol-Independent Multicast. See PIM
protocols, 5, 68
alternating-bit, 214
application-layer, 4950
congestion-control, 9
defining, 79
hardware-implemented, 9
human analogy, 78
interior gateway, 384
Internet, 9
IP (Internet Protocol), 5
layering, 4950
nonce, 704705
packet sizes, 335
plug-and-play, 346
port numbers, 90
real-time interactive applications,
623632
routing, 5152
RTP, 623626
SIP, 626632
soft state, 408409
SR (selective repeat), 223230
stateless, 100
stop-and-wait, 209, 215, 217
TCP, 5
UDP, 5
transmission and receipt of messages,
78
transport-layer, 50
protocol stack, 50
provider, 32
proxy servers, 106, 110, 738
public key algorithm, 716
public key certification, 697699
public-key cryptography, 708
digital signatures, 693694
private key, 693
public key, 693, 697
secure e-mail system, 706707
public key encryption, 683688
public-key encryption algorithm, 687
public keys, 684685, 693, 706708,
713714
binding to particular entity, 697698
certifying, 708
encryption/decryption algorithms,
684
public key systems, 676
pull protocol, 124
pulse code modulation. See PCM (pulse
code modulation)
push protocol, 124
PUT method, 105
Python, 157, 160, 193
INDEX 847
Q
QAM16 modulation, 521
Q2931b protocol, 654
QoS (Quality-of-Service), 329, 653654
QQ, 592, 623
Quality-of-Service. See QoS
quantization, 590
query
ARP message, 467
information about, 141
query messages, 140142
queues
FIFO (first-in-first-out), 641642
packet-discarding policy, 641
priority queuing, 642643
provable maximum delay, 647648
round robin queuing discipline,
643644
WFQ (weighted fair queuing),
644645
work-conserving round robin
discipline, 644
queuing, 327331
queuing delays, 25, 3637, 3942, 60
R
radio channels, 21
Radio Network Controller. See RNC
RADIUS protocol, 530, 730
random access protocols, 447, 473
Aloha protocol, 452453
CSMA (carrier sense multiple access)
protocol, 453456
CSMA/CD (carrier sense multiple
access with collision detection),
455459
slotted ALOHA protocol, 450452
Random Early Detection algorithm. See
RED algorithm
rarest first, 149
rate adaptation, 542543
RC4 algorithm, 727728
RCP (Routing Control Platform), 786
rdt (reliable data transfer protocol), 204
building, 206215
packet reordering, 229230
pipelined, 215218
TCP (Transmission Control Protocol),
204
unreliable layer below, 204
real-time applications
timing, 92
UDP (User Datagram Protocol), 200
real-time interactive applications
protocols, 623632
RTP (Real-Time Transport Protocol),
623626
SIP, 626632
real-time measurements of delay and loss
performance, 606
Real-Time Streaming Protocol. See RTSP
Real-Time Transport Protocol. See RTP
receive buffer, 233
receiver authentication, 706
receiver-based recovery, 621
receiver feedback, 208
receivers
ACK generation policy, 247
defining operation, 206
sequence number of packet acknowl-
edged by ACK message, 212
receiver-side transport layer, 54
receive window, 250252
receive window field, 234
receiving adapter, 472
receiving processes addresses, 90
records, inserting in DNS database, 142,
144
recursive queries and DNS servers,
137138, 140
RED (Random Early Detection)
algorithm, 329
regional ISPs, 33
registrars, 142
registration with home agent, 568569
relays, 622623
reliable channel, 204
reliable data transfer, 91, 190
application layer, 204
channel with bit errors, 207212
848 INDEX
link layer, 204
lossy channel with bit errors, 212215
perfectly reliable channel, 206207
principles, 204230
reliable channel, 204
TCP (Transmission Control Protocol),
230231, 240, 242250
transport layer, 204
transport-layer protocols, 91
reliable data transfer protocol. See rdt
reliable data transfer service, 235
reliable delivery, 436
reliable transport service, 269
remote host, transferring files, 116118
rendezvous point, 404
repeater, 474
replicated servers, 132
reply messages and DNS (domain name
system), 140142
repositioning video, 600
request messages and HTTP, 103105
request-response mode, 772
requests for comments. See RFCs
Request to Send control frame. See RTS
control frame
residential ISPs, 87
resource-management cells. See RM cells
resource records. See RRs
resource reservation protocols, 362
resources
admitting or blocking flows, 653
efficient use of, 640
reservations, 653654
response ARP, 467
response messages and HTTP, 105108
retransmission, 208, 212
retransmitting data, 241, 262
retransmitting packets, 259, 261263
reverse path broadcast. See RPB
reverse path forwarding. See RPF
Rexford, Jennifer, 786787
RFCs (requests for comments), 5
RIP advertisements, 384385
RIP request message, 387
RIP response message, 384
RIP routers, 386387
RIP (Routing Information Protocol), 384,
498
hops, 384
implementation aspects, 386388
IP network-layer protocol, 387
lower-tier ISPs, 388
modifying local routing table and prop-
agating information, 387
RIP messages, 384385
RIP table, 385386
routing updates, 384
UDP transport-layer protocol, 387
UNIX implementation, 387388
Rivest, Ron, 684, 690
RM (resource-management cells),
267269
RNC (Radio Network Controller), 552
roaming number, 572
Roberts, Larry, 61, 511
root DNS servers, 134136
round robin queuing discipline, 643644
round-trip delays, 43
round-trip time. See RTT
route aggregation, 342
route attributes, 395
router control plane functions, 322
router discovery message, 566567
router forwarding plane, 321
routers, 4, 12, 22, 53, 303, 310
access control lists, 734
adapters, 468
address of, 43
administrative autonomy, 380
area border, 389
ARP modules, 468
AS-PATH attribute, 394
ASs (autonomous systems), 380
authenticated and encrypted channel
between, 725
buffering packet bits, 24
buffer sizing, 328329
connected into network, 338
connection state information, 315
control functions, 321322
INDEX 849
routers (continued)
control plane implemented in, 331
data center hierarchy, 492493
default, 364
destination, 364
finite buffers, 261265
firewalls, 355, 481
first-hop, 364
fixed-length labels, 487
forwarding function, 320322
forwarding table, 26, 308309,
317318, 322323, 394, 396397,
469
gateway, 380381
implementing layers 1 through 3, 53
incident links, 22
input ports, 320
input processing, 322324
interfaces, 338, 468
intra-AS routing protocols, 397
IP addresses, 394, 465, 468
IP protocol, 53
label-switched, 488
layer-2 packet switch, 480
link-layer and MAC addresses,
462463, 465
longest prefix matching rule, 318319
lookup, 323324
looping advertisements, 394
memory access times, 324
network core, 4
network-layer addresses, 462, 465
output ports, 320321
output processing, 326
packet-forwarding decisions, 364
packet loss, 327
packets not cycling through, 481
physical links between, 364
plug-and-play, 481
primary role, 306
processing datagrams, 480
processing packets, 351
protocols, 9
queuing, 327331
routing control plane, 331
routing packets, 380382
routing processor, 321
routing tables, 385386
scale, 379380
self-synchronizing, 371
source, 364
spanning tree, 481
store-and-forward, 22, 24
store-and-forward packet switches, 480
versus switches, 480482
switching, 320, 324326
terminating incoming physical link, 320
VC setup, 316
routes, 4, 394396
route summarization, 342
routing, 305306, 308310
advertising information, 382383
broadcast, 399405
calls to mobile user, 571572
distance vector, 384
hierarchical, 379383
hot-potato, 382
to mobile node, 559564
multicast, 399, 405412
storing information, 379380
routing algorithms, 309, 363383
ARPAnet, 366
circuit-switched, 379
decentralized, 366
DV (distance-vector) algorithm, 366,
371379
dynamic, 366
forwarding tables, 364
global, 365366
hierarchical routing, 379383
least costly paths, 365
load-sensitive, 366
LS (link-state) algorithms, 366371
path from source to destination router,
364
scale of routers, 379380
static, 366
switches, 494495
viewing packet traffic flows, 379
routing control plane, 331
850 INDEX
Routing Control Platform. See RCP
routing daemons, 674
Routing Information Protocol. See RIP
routing loop, 377
routing protocols, 2627, 5152
BGP (Border Gateway Protocol),
390399, 498499
DV (distance vector) algorithms,
374375
executing, 321
inter-AS, 382
Internet, 383399
intra-AS, 380381
IS-IS, 384
messages, 309
OSPF (Open-Shortest Path First), 384
RIP (Routing Information Protocol),
384
RPB (reverse path broadcast), 402
RPF (reverse path forwarding), 402403,
411
RRs (resource records), 139141
RSA algorithm, 684688, 710
RST flag bit and segment 235, 258
RSVP, RSVP-TE protocol, 489, 654
RTP packets, 624625
RTP (Real-Time Transport Protocol), 588,
623626, 668
UDP streaming, 595
RTS/CTS exchange, 537
RTS frame, 536537
RTSP (Real-Time Streaming Protocol),
117, 595, 668
RTS (Request to Send) control frame,
535537
RTT (round-trip time), 102103
EWMA (exponential weighted
moving average), 240
TCP (Transmission Control Protocol),
238241
S
SAD (Security Association Database), 721
SA (security association), 720721
satellite links, 16, 2122
scalability and P2P architecture, 145148
scheduling mechanisms, 640645
Schulzrinne, Henning, 623, 632, 668670
SDN (Software Defined Networking), 786
secure communication, 672674
secure e-mail system, 706708
Secure Hash Algorithm. See SHA-1
secure networking protocols and
message integrity, 689
Secure Network Programming, 511
Secure Sockets Layer. See SSL
security, 5556
application-layer protocol, 705
attacks, 674
cryptography, 675688
data link layer, 705
digital signatures, 688699
e-mail, 705711
end-point authentication, 700705
IEEE 802.11i, 728731
IP datagrams, 718
IP (Internet Protocol), 362363
IPsec, 362
message integrity, 688693
mobile IP, 566
network layer, 705, 718725
networks, 671674
operational, 673, 731742
OSPF (Open-Shortest Path First),
388389
P2P architecture, 88
public key encryption, 683688
RSA, 687
SNMPv3, 775778
switches, 479
TCP connection, 711717
transport-layer protocols, 93, 705
transport services, 93
user-based, 777
WEP (Wired Equivalent Privacy),
726728
wireless LANs, 726731
security and administration capabilities,
765
security association. See SA
INDEX 851
Security Association Database. See SAD
Security Management, 759, 764
Security Policy Database. See SPD
segments, 51, 186, 189
acknowledgment number, 236
destination port number field, 192
fast retransmit, 248
fields, 191192
out-of-order, 236
piggybacked acknowledgment, 237
sequence numbers, 235238
source port number field, 192
TCP (Transmission Control Protocol),
233
unique identifiers, 192
selective acknowledgment, 250
selective repeat protocols. See SR
protocols
self-learning, 478479, 497, 542
self-replicating, 56
self-scalability, 87
send buffer, 232
sender
countdown timer, 214
defining operation, 206
detecting and recovering from lost
packets, 212215
leftmost state, 208
receive window, 250
rightmost state, 208
sending multiple packets without
acknowledgments, 218
sequence number of packet, 212
utilization, 217
sender authentication, 706708
sender-to-receiver channel, 213214
sending rates, 260
send side states rdt2.0 protocols, 208
sequence-number-controlled flooding,
401403, 405
sequence numbers, 210, 212, 218220,
234, 614615, 618, 717
IPsec, 724
RTP packets, 625
SSL (Secure Sockets Layer), 715
SYN segment, 252253
TCP segments, 235236
TCP (Transmission Control Protocol),
244, 249
Telnet, 237238
server authentication, 712
server processes, 88, 164, 232
server program, 156, 163
servers, 2, 1011, 8889
always on, 86
dedicated socket, 167
hostname of, 160
IP addresses, 86, 160, 161, 163
network attacks, 5758
non-persistent connections, 198
persistent HTTP, 198
port number, 161, 167
TCP socket creation, 167
Web caches as, 111
server SMTP, 122
server socket TCP connection, 163
server-to-client throughput, 4445
Service Level Agreements. See SLAs
service model, 49
service providers and private
networks, 66
services, 49
description of Internet, 57
DNS (domain name system),
131133
flow of packets, 311
transport layer, 186
transport protocols, 189
Service Set Identifier. See SSID
Serving GPRS Support Nodes. See SGSNs
session encryption key, 714
Session Initiation Protocol. See SIP
session keys, 687, 707, 714
session layer, 53
SGMP (Simple Gateway Monitoring
Protocol), 764
SGSNs (Serving GPRS Support Nodes),
552
SHA, 710
Shamir, Adi, 684
852 INDEX
Shannon, Claude, 80, 82
shared medium, 20
SHA-1 (Secure Hash Algorithm), 691
shortest paths, 365
SIFS (Shorter Inter-frame Spacing), 532
signaling messages, 316
signaling protocols, 317
signal-to-noise ratio. See SNR
signature-based IDSs (intrusion detection
systems), 741742
silent periods, 2930
simple authentication, 389
Simple Gateway Monitoring Protocol. See
SGMP
Simple Mail Transfer Protocol. See
SMTP
Simple Network Management Protocol.
See SNMP
single-hop, wireless networks, 518
SIP (Session Initiation Protocol), 588,
626632, 668669
Skype, 65, 83, 87, 588, 621623
conversational voice and voice, 592
proprietary application-layer protocols,
97
UDP (User Datagram Protocol), 613
SLAs (Service Level Agreements), 758
sliding-window protocol, 220
slotted ALOHA protocol, 450452
nodes decision to transmit, 453455
small office, home office subnets. See
SOHO subnets
SMI (Structure of Management
Information), 765, 766769
SMTP clients, 122123
SMTP servers, 123
SMTP (Simple Mail Transfer Protocol),
51, 97, 117, 120127
SNMP applications, 776777
SNMP messages, 777
SNMP (Simple Network Management
Protocol), 758759, 762, 764778
SNMPv3, 765, 775778
SNMPv2 (Simple Network Management
Protocol version 2), 772, 773775
Snort IDS system, 740742
SNR (signal-to-noise ratio), 520521
social networking, 83, 86
social networks, 6465, 100
socket interface, 100
socket module, 160
socket programming
TCP (Transmission Control Protocol),
158, 163
UDP, 157158
sockets, 8991, 91, 191
assigning port number, 162
port number, 158
soft guarantee, 634
soft state protocols, 408409
software control plane, 331
Software Defined Networking. See SDN
SOHO (small office, home office) subnets
and IP addresses, 349352
source
host and source router, 364
total delay to destination, 4244
source port numbers, 192, 194, 196, 234
source quench message, 353
source router, 364
source-specific congestion-control actions,
267
source-specific multicast. See SSM
spam, 56
spanning-tree broadcast, 403405
spanning trees, 403405, 481
spatial redundancy, 589
SPD (Security Policy Database), 724
special socket server program, 163
speed-matching service, 250
SPI (Security Parameter Index), 721
split-connection approaches, 577
Sprint, 5, 33, 758
spyware, 56
SRAM, 324
SR (selective repeat) protocols,
223230
SSH protocol, 237
SSID (Service Set Identifier), 529
SSL record, 715716
INDEX 853
SSL (Secure Sockets Layer), 711
anonymity, 738
API (Application Programmer
Interface) with sockets, 712
block ciphers, 678
breaking data stream into records, 714
connection closure, 717
cryptographic algorithms, 716
data transfer, 713715
designed by Netscape, 711
handshake, 713714, 716717
HTTP transactions security, 712
key derivation, 713714
nonces, 717
popularity, 711
privacy, 738
public key certification, 697
sequence numbers, 715
SSL classes/libraries, 712
SSL record, 715716
transport protocols, 712
SSM (source-specific multicast), 412
state, 117
stateful packet filters, 732, 735736
stateless protocols, 100
static routing algorithm, 366
stations, 531533
status line in HTTP response messages,
106
steaming prerecorded videos, 591
stop-and-wait protocols, 209210, 215,
217
store-and-forward packet switches, 22, 24,
480
stream ciphers, 678
streaming, 591
live audio and video, 587, 593
stored audio and video, 587, 591592
video, 589
streaming stored video, 593612
adaptive HTTP streaming, 593
adaptive streaming, 600601
bandwidth, 594
CDNs (content distribution
networks), 602608
client buffering, 594595
continuous playout, 591592
DASH (Dynamic Adaptive Streaming
over HTTP), 600601
end-to-end delays, 594
HTTP streaming, 593, 596600
interactivity, 591
KanKan, 611612
Netflix, 608610
streaming, 591
UDP streaming, 593, 595596
YouTube, 610611
streaming video, 592
TCP (Transmission Control Protocol),
596
Structure of Management Information. See
SMI
stub network, 397398
multi-homed, 397
subnet mask, 340
subnets, 340
advertising existence to Internet,
391
class A, B and C networks, 344
defining, 341
DHCP offer message, 347
DHCP servers, 346
IP addresses, 340, 342, 345
IP definition of, 340341
prefixes, 393
sending datagrams off, 468469
shortest-path tree, 388
successful slots, 451
switched Ethernet, 470
switched-LANs
ARP (Address Resolution Protocol),
465468
Ethernet, 469476
link-layer addressing, 462469
link-layer switches, 476482
MAC addresses, 463465
switch poisoning, 480
VLANs (virtual local area networks),
482486
switched networks, 481
854 INDEX
switches, 80
aging time, 478
broadcasting frames, 464
broadcast storms, 481
collisions elimination, 479
congestion-related information, 268
data center hierarchy, 492493
enhanced security, 479
Ethernet, 470, 475
filtering, 476477
filtering frame, 477
forwarding, 476477
gathering statistics, 479
heterogeneous links, 479
high filtering and forwarding rates, 480
link-layer, 461, 476482
link-layer addresses, 462
link-layer frames, 476
MAC addresses, 480
management, 479
plug-and-play devices, 479480
processing frames, 480
versus routers, 480482
routing algorithms, 494495
self-learning, 478479, 497, 542
small networks, 482
store-and-forward packet switches,
480
switch table, 476
tracking behavior of senders, 267
transparent, 476
trunk port to interconnect, 484
VLANs (virtual local area
networks), 483484
switch fabric, 320, 322, 327, 329330
switching and routers, 324326
switch output interfaces buffers, 476
switch poisoning, 480
switch table, 476477
symmetric algorithm, 716
symmetric key, 706707, 707
symmetric key algorithm
block ciphers, 678681
Caesar cipher, 676
monoalphabetic cipher, 676677
polyalphabetic encryption, 678
stream ciphers, 678
symmetric key encryption and CBC
(cipher-block chaining), 681682
SYNACK segment, 257258
SYN bit, 235, 253
SYN cookies, 257
SYN flood attack, 252, 253, 257
SYN packet, 258
SYN segments, 252254, 257258
SYN_SENT state, 254
T
taking-turns protocols, 447, 459460
TCAMs (Ternary Content Address
Memories), 324
TCP buffers, 597598
TCPClient.py client program, 164166
TCP clients, 195, 253255
TCP congestion-control algorithm,
272277, 279
TCP connections, 57, 94
allocating buffers and variables, 253
bandwidth, 281
bottleneck link, 279281
buffers, 233
between client and server, 166
client process, 232
client-side TCP sending TCP
segment to server-side TCP,
252253
client socket, 163
connection-granted segment, 253
ending, 253254
establishing, 232, 252253, 713
full-duplex service, 232
HTTP server, 596
management, 252256, 258
out-of-order segments, 236
packet loss, 281
parallel and fairness, 282
point-to-point, 232
processes sending data, 232233
receive buffer, 233, 250
regulating rate of traffic, 190
INDEX 855
TCP connections (continued)
security, 711717
send buffer, 232
server process, 232
server socket, 163
socket connection to process, 233
split-connection approaches, 577
three-way handshake, 102103, 166,
232
throughput, 280
transporting request message and
response message, 101
variables, 233
TCP header, 234235
TCP/IP (Transmission Control
Protocol/Internet Protocol), 5, 63,
93, 231, 431
TCP ports, 258
TCP Reno, 276, 278
TCP segments, 233236, 253
with different source IP addresses,
194195
header overhead, 200
loss, 266
reordering, 715
structure, 233238
TCP sender, 242243, 269, 270
awareness of wireless links, 577
congestion control, 250
TCP server, 163, 195
TCPServer.py server program, 166168
TCP sockets, 165166, 497, 499
server-side connection socket, 163
welcoming socket, 163
TCP splitting, 273
TCP streaming and prefetching video, 597
TCP SYNACK segment, 499
TCP SYN segment, 499
TCP Tahoe, 276
TCP (Transmission Control Protocol), 5,
51, 93, 189, 313, 338
acknowledgment numbers, 244
block ciphers, 678
buffer and out-of-order segments, 249
buffer overflow, 251
byte stream, 242
checksum, 334
client-server application, 157
congestion avoidance, 272276
congestion control, 95, 190, 199200,
240, 247, 269272, 274283,
576577, 596, 613
congestion window, 269270,
276277, 576
connection-establishment delays, 200
connection-oriented, 94, 163, 230238
connection state, 200, 231
continued evolution of, 279
cumulative acknowledgments, 236,
243, 248249
duplicate ACK, 247248
early versions, 62
end-to-end congestion control, 266,
269
extending IPs delivery service, 190
fairness, 279282
fast retransmit, 247248
flow control, 240, 250252
full-duplex, 235
GBN (Go-Back-N) protocol, 248250
high-bandwidth paths, 279
host-based congestion control, 63
HTTP and, 116, 200
implicit NAK mechanism, 240
integrity checking, 190
Internet checksum, 442
lost acknowledgment, 244
lost segments, 238
MSS (maximum segment size),
232234
MTU (maximum transmission unit),
232233
multimedia applications, 200
negative acknowledgments, 248
packet loss, 247248, 613
pipelining, 240
positive acknowledgments, 240
receive buffer, 270
receiver-so-sender ACK, 576
receive window, 251
856 INDEX
reliable data transfer, 96, 190,
230231, 240
reliable data transfer service, 95, 100,
123, 163, 199200, 235, 242250
resending segment until acknowledged,
199
retransmission timeout interval, 241
retransmission timer, 242
retransmitting data, 473
retransmitting segments, 239240, 246,
249, 575576
RST segment, 258
RTT (round-trip time) estimation,
238241
security services, 95
segments, 189
selective acknowledgment, 250
separation of IP, 62
sequence numbers, 244, 249
server-to-client transmission rate, 596
services, 9495
socket programming, 158, 163
states, 254
state variable, 243
steady-state behavior, 278279
streaming media, 200201
streaming video, 596
SYNACK segment, 258
SYN segments, 257258
TCP Reno, 276, 278
TCP segments, 233
TCP Tahoe, 276
TCP Vegas, 278
32-bit sequence number, 220
three-way handshake, 163, 200, 253
throughput macroscopic description,
278279
timeout, 238241, 243
timeout, 244247
timeout/retransmit mechanism, 238
transmission rate, 278
Web servers, 197198
window size, 266
wireless networks, 575577
TCP Vegas, 278
TDM (time-division multiplexing), 2830,
31, 448, 549
telco (telephone company), 1314
Telenet, 62
telephone company. See telco
telephone networks, 27
circuit switching, 60
complexity, 319
frequency band, 29
packet switching, 31
Telnet, 86
blocked, 737
sending message to mail server, 125
SMTP server, 124
TCP example, 234, 237238
temporary IP address, 346
10BASE-2, 473474
10BASE-T, 473474
10GBASE-T, 474475
Ternary Content Address Memories. See
TCAMs
3GPP (3rd Generation Partnership
Project), 550, 552, 362
third-party CDNs (Content Distribution
Networks), 603
3DES, 680
3G cellular data networks, 550552
3G cellular mobile systems versus wire-
less LANs, 548
3G core network, 550552
3G networks, 669
3G radio access networks, 552
3G systems, 547
3G UMTS and DS-WCDMA (Direct
Sequence Wideband CDMA), 552
three-way handshake, 102103, 232, 253,
499, 735
throughput, 260
average, 44
end-to-end, 4447
fluctuations in, 92
instantaneous, 44
macroscopic description for TCP,
278279
server-to-client, 4445
INDEX 857
throughput (continued)
streaming video, 592
TCP connection, 280
transmission rates of links, 47
transport-layer protocols, 92
zero in heavy traffic, 265
tier-1 ISPs, 3334
time-division multiplexing. See TDM
timeout
doubling interval, 246247
event, 222, 244
length of intervals, 238239
setting and managing interval, 241
TCP (Transmission Control Protocol),
238241, 243
timer management and overhead, 242
time-sensitive applications, 95
time-sharing networks, 62
time slots, 448
timestamps, 614615, 617, 625
time-to-live field. See TTL (time-to-live)
field
timing guarantees, 9293
TLD (top-level domain) DNS servers,
134136, 143
DNS servers, 134
TLS (Transport Layer Security), 711
TLV (Type, Length, Value) approach, 780
token-passing protocol, 459460
top-down approach, 50
top-level domain DNS servers. See TLD
DNS servers
top-level domains, 135
Top of Rack switch. See TOR switch
top-tier switch, 492
TOR anonymizing and privacy service,
738
torrents, 149
TOR (Top of Rack) switch, 490, 492
TOS (type of service) bits, 333
total nodal delay, 36
Traceroute program, 27, 353355
end-to-end delays, 4243
tracker, 149
traditional packet filters, 732734
traffic
bursty, 60
conditioning, 648649
intensity, 40
traffic engineering, 489
traffic isolation, 638640
traffic policing, 638639
traffic profile, 650
transferring files, 116118
transfer time, 45
Transmission Control Protocol. See TCP
Transmission Control Protocol/Internet
Protocol.
transmission delays, 3639
transmission rates, 4, 4546
transmitting
frames, 532
packets in datagram networks, 317
transport layer, 51, 53, 185
application-layer message, 54
automatically assigning port number,
193194
checksumming, 442443
congestion control, 266
connectionless service, 313
connection-oriented service, 313314
datagram passed, 337
delivering data to socket, 191
demultiplexing, 191198
destination host, 191
error checking, 203
multiplexing, 191198
multiplexing/demultiplexing
service, 198199
network layer relationship, 186189
overview, 189191
process-to-process communication,
305, 313
reliable data transfer, 204
responsibility of delivering data to
appropriate application, 191
segments, 189
services, 186
transport-layer multiplexing, 192
transport-layer packets, 186
858 INDEX
transport-layer protocols, 50, 91
end systems implementation, 186
IP datagrams, 334
living in end systems, 188
logical communication between
processes, 186, 188189
reliable data transfer, 91
reliable delivery, 436
security, 93, 705
TCP (Transmission Control Protocol),
189
throughput, 92
timing, 9293
UDP (User Datagram Protocol), 189
Transport Layer Security. See TLS
transport-layer segments, 5455, 186
datagrams, 242
delivering data to correct socket,
191198
fields, 191
unreliability, 242
transport mode, 721
transport protocols
Internet applications, 96
services, 189
SSL (Secure Sockets Layer), 712
TCP, 51
UDP, 51
transport services
available to applications, 9193
connection-oriented service, 94
provided by Internet, 9396
reliable data transfer, 91
security, 93
TCP services, 9495
throughput, 92
timing, 9293
UDP, 95
trap messages, 773
tree-join messages, 404405
triangle routing problem, 563
triple-DES, 710
truncation attack, 717
TTL (time-to-live) field, 139140, 334
tunneling, 360361, 561
tunnel mode, 721722
twisted-pair copper wire, 1920, 475
Twitter, 65, 83, 86
two-dimensional parity scheme, 441442
2G cellular networks architecture,
548550
Type, Length, Value approach. See TLV
approach
type of service bits. See TOS bits
U
UDP checksum, 202204
UDPClient.py client program, 158161
UDP header, 202
UDP packet, 258, 346, 595
UDP ports, 258
UDP segments, 202204, 495497, 613
UDPServer.py server program, 158, 161,
194
UDP sockets
communicating to processes, 158
creation, 161
identifying, 194
port numbers, 193194
UDP streaming, 593, 595596
UDP (User Datagram Protocol), 51, 93,
189, 387
checksum, 208, 334
client-server application, 157
congestion control, 201, 282
connection establishment, 200
connectionless transport, 95, 198204
connection state, 200
datagrams, 189
delays, 200
destination port number, 199
development, 62
directly talking with IP, 199
discarding damaged segment, 204
DNS and, 199200
end-to-end principle, 203
end-to-end throughput, 95
error checking, 199
error detection, 202204
extending IPs delivery service, 190
INDEX 859
UDP (User Datagram Protocol)
(continued)
fairness, 282
finer application-level control over
data, 199
flow control, 252
gaps in data, 473
handshaking, 199
header overhead, 200
integrity checking, 190
Internet checksum, 442
Internet telephony applications, 96
multimedia applications, 200201,
282
multiplexing/demultiplexing
function, 199
network management data, 200
no-frills segment-delivery service, 199
packet loss, 613
passing damaged segment to
application, 204
real-time applications, 200
reliable data transfer, 201
RIP routing table updates, 200
RTP and, 624
segments, 189
small packet header overhead, 200
socket programming, 157158
transport services, 95
unreliability, 95, 190
wireless networks, 575577, 301
UMTS (Universal Mobile
Telecommunications Service)
3G standards, 550
unchoked, 150
uncontrolled flooding, 401
undetected bit errors, 440
unguided media, 19
unicast addresses, 356
unicast applications and RTP packets,
624
unicast communication and IP addresses,
406
unidirectional data transfer, 205
Universal Plug and Play. See UPnP
UNIX
BSD (Berkeley Software Distribution)
version, 384
nslookup program, 141142
RIP implemented in, 387388
Snort, 742
unreliable data transfer, 206
unreliable service, 190
unshielded twisted pair. See UTP
UPnP (Universal Plug and Play), 352
urgent data pointer field, 235
URL field, 104
URLs, 99
US Department of Defense Advanced
Research Projects Agency. See
DARPA
user agents, 119121, 126127
user-based security, 777
user-server interaction and HTTP
(HyperText Transfer Protocol),
108110
utilization, 217
UTP (unshielded twisted pair), 1920
V
VANET (vehicular ad hoc network), 518
variables and TCP connection, 233
VC networks, 314317, 319320
VC (virtual-circuit), 267, 314
roots in telephony world, 319
terminating, 316
vehicular ad hoc network. See VANET
Verizon, 758
FIOS service and PONs (passive opti-
cal networks), 1516
version number, 333
video, 588589
P2P delivery, 611
prefetching, 596597
prerecorded, 591
repositioning, 600
streaming stored, 593612
timing considerations and tolerance of
data loss, 592
traversing firewalls and NATs, 596
860 INDEX
video conferencing, 83
video over IP, 592593
video stream, 625
virtual-circuit. See VC (virtual-circuit)
virtual local area networks. See VLANs
virtual private networks. See VPNs
viruses, 56, 740
visited MSC, 574
visited networks, 557, 570
visitor location register. See VLR
VLANs (virtual local area networks),
482486
VLAN tag, 484486
VLAN trunking, 484485
VLR (visitor location register), 570
voice and video applications, 83
VoIP (Voice-over-IP), 83
adaptive playout delay, 615618
end-to-end delay, 613614
enhancing over best-effort network, 612
fixed playout delay, 615
jitter and audio, 614618
media packetization delays, 44
packet loss, 613
recovering from packet loss, 618621
sequence numbers, 615
timestamps, 615
wireless systems, 668
VPNs (virtual private networks), 362
confidentiality, 720
end points, 725
IPsec, 718720
IPv4, 719
MPLS (Multiprotocol Label
Switching), 489490
SA (security association), 720
tunnel mode, 721
vulnerability attacks, 57
W
Web, 64, 86, 97
client-server application architecture,
100
HTTP (HyperText Transfer Protocol),
98100
network applications, 98116
operating on demand, 98
platform for applications emerging
after 2003, 98
terminology, 9899
Web applications, 97
client and server processes, 88
client-server architecture, 86
Web-based e-mail, 86, 129130
Web browsers, 97
client side of HTTP, 99
GUI interfaces, 64
Web caches, 59, 110115
Web client-server interaction, 499
web of trust, 710
Web pages, 99
displaying, 101
requests, 495500
Web proxy caches, 104
Web servers, 89, 97
deleting objects, 105
initial versions, 64
IP addresses, 392
port numbers, 197198
server processes, 88
server side of HTTP, 99
spawning new process for connections,
198
TCP (Transmission Control Protocol),
197198
uploading objects to, 105
Web sites, 108
anonymity, 738
privacy, 738
weighted fair queuing. See WFQ
well-known port number, 192
WEP (Wired Equivalent Privacy), 726728
WFQ (weighted fair queuing), 329, 644645
leaky bucket, 647648
wide-area wireless access, 18
WiFi, 17, 52, 526546
high-speed, 65
home networks, 17
hotspots, 515, 546
public access, 515
INDEX 861
WiMAX (World Interoperability for
Microwave Access), 554, 668
Windows
nslookup program, 141142
Snort, 742
Wireshark packet sniffer, 78
window size, 220
wired-access ISPs tiered levels of
service, 636
wired broadcast links, 521
wired environments and packet sniffer,
5859
Wired Equivalent Privacy. See WEP
wired link differences from wireless links,
519
wired networks, 519
wireless, 513514
wireless communication links, 515516
wireless devices, 5859
wireless hosts, 514, 516517, 530
wireless LANs, 445
access point, 17
LAN base stations, 548
DHCP (Dynamic Host Configuration
Protocol), 346
versus 3G cellular mobile systems, 548
IEEE 802.11 technology, 17
security, 726731
WiFi, 17
wireless LANs and 802.11 standards, 526
wireless links
bit errors, 519
decreasing signal strength, 519
differences from wired links, 519
fading signals strength, 521522
hidden terminal problem, 521
interference from other sources, 519
multipath propagation, 519
TCP sender awareness, 577
undetectable collisions, 521522
wireless mesh networks, 518
wireless networks, 513
application layer, 575
base station, 516518
CDMA (code division multiple access)
protocol, 522526
characteristics, 519526
802.11 wireless LANs, 526546
link layer, 575
link rates, 515
mobility, 575577
multi-hop, infrastructure-based, 518
multi-hop, infrastructure-less, 518
network infrastructure, 518
network layer, 575
single-hop, infrastructure-based, 518
single-hop, infrastructure-less, 518
TCP (Transmission Control Protocol),
575577
UDP (User Datagram Protocol),
575577
wireless communication links, 515516
wireless hosts, 514
wireless personal area network. See WPAN
Wireless Philadelphia, 515
wireless station, 529530
Wireshark labs, 59, 78
work-conserving round robin discipline, 644
workload model, 635
World Wide Web. See Web
worms, 5657, 740
WPAN (wireless personal area network),
544
X
X.25, 512
XNS (Xerox Network Systems)
architecture, 384
Y
Yahoo!, 65, 86, 130
YouTube, 65, 588, 610611
HTTP streaming (over TCP), 596
streaming stored video, 591
video, 602
Z
Zigbee, 545546
862 INDEX
Cover
Title Page
Copyright Page
Table of Contents
Chapter 1 Computer Networks and the Internet
1.1 What Is the Internet?
1.1.1 A Nuts-and-Bolts Description
1.1.2 A Services Description
1.1.3 What Is a Protocol?
1.2 The Network Edge
1.2.1 Access Networks
1.2.2 Physical Media
1.3 The Network Core
1.3.1 Packet Switching
1.3.2 Circuit Switching
1.3.3 A Network of Networks
1.4 Delay, Loss, and Throughput in Packet-Switched Networks
1.4.1 Overview of Delay in Packet-Switched Networks
1.4.2 Queuing Delay and Packet Loss
1.4.3 End-to-End Delay
1.4.4 Throughput in Computer Networks
1.5 Protocol Layers and Their Service Models
1.5.1 Layered Architecture
1.5.2 Encapsulation
1.6 Networks Under Attack
1.7 History of Computer Networking and the Internet
1.7.1 The Development of Packet Switching: 19611972
1.7.2 Proprietary Networks and Internetworking: 19721980
1.7.3 A Proliferation of Networks: 19801990
1.7.4 The Internet Explosion: The 1990s
1.7.5 The New Millennium
1.8 Summary
Homework Problems and Questions
Wireshark Lab
Interview: Leonard Kleinrock
Chapter 2 Application Layer
2.1 Principles of Network Applications
2.1.1 Network Application Architectures
2.1.2 Processes Communicating
2.1.3 Transport Services Available to Applications
2.1.4 Transport Services Provided by the Internet
2.1.5 Application-Layer Protocols
2.1.6 Network Applications Covered in This Book
2.2 The Web and HTTP
2.2.1 Overview of HTTP
2.2.2 Non-Persistent and Persistent Connections
2.2.3 HTTP Message Format
2.2.4 User-Server Interaction: Cookies
2.2.5 Web Caching
2.2.6 The Conditional GET
2.3 File Transfer: FTP
2.3.1 FTP Commands and Replies
2.4 Electronic Mail in the Internet
2.4.1 SMTP
2.4.2 Comparison with HTTP
2.4.3 Mail Message Format
2.4.4 Mail Access Protocols
2.5 DNSThe Internets Directory Service
2.5.1 Services Provided by DNS
2.5.2 Overview of How DNS Works
2.5.3 DNS Records and Messages
2.6 Peer-to-Peer Applications
2.6.1 P2P File Distribution
2.6.2 Distributed Hash Tables (DHTs)
2.7 Socket Programming: Creating Network Applications
2.7.1 Socket Programming with UDP
2.7.2 Socket Programming with TCP
2.8 Summary
Homework Problems and Questions
Socket Programming Assignments
Wireshark Labs: HTTP, DNS
Interview: Marc Andreessen
Chapter 3 Transport Layer
3.1 Introduction and Transport-Layer Services
3.1.1 Relationship Between Transport and Network Layers
3.1.2 Overview of the Transport Layer in the Internet
3.2 Multiplexing and Demultiplexing
3.3 Connectionless Transport: UDP
3.3.1 UDP Segment Structure
3.3.2 UDP Checksum
3.4 Principles of Reliable Data Transfer
3.4.1 Building a Reliable Data Transfer Protocol
3.4.2 Pipelined Reliable Data Transfer Protocols
3.4.3 Go-Back-N (GBN)
3.4.4 Selective Repeat (SR)
3.5 Connection-Oriented Transport: TCP
3.5.1 The TCP Connection
3.5.2 TCP Segment Structure
3.5.3 Round-Trip Time Estimation and Timeout
3.5.4 Reliable Data Transfer
3.5.5 Flow Control
3.5.6 TCP Connection Management
3.6 Principles of Congestion Control
3.6.1 The Causes and the Costs of Congestion
3.6.2 Approaches to Congestion Control
3.6.3 Network-Assisted Congestion-Control Example: ATM ABR Congestion Control
3.7 TCP Congestion Control
3.7.1 Fairness
3.8 Summary
Homework Problems and Questions
Programming Assignments
Wireshark Labs: TCP, UDP
Interview: Van Jacobson
Chapter 4 The Network Layer
4.1 Introduction
4.1.1 Forwarding and Routing
4.1.2 Network Service Models
4.2 Virtual Circuit and Datagram Networks
4.2.1 Virtual-Circuit Networks
4.2.2 Datagram Networks
4.2.3 Origins of VC and Datagram Networks
4.3 Whats Inside a Router?
4.3.1 Input Processing
4.3.2 Switching
4.3.3 Output Processing
4.3.4 Where Does Queuing Occur?
4.3.5 The Routing Control Plane
4.4 The Internet Protocol (IP): Forwarding and Addressing in the Internet
4.4.1 Datagram Format
4.4.2 IPv4 Addressing
4.4.3 Internet Control Message Protocol (ICMP)
4.4.4 IPv6
4.4.5 A Brief Foray into IP Security
4.5 Routing Algorithms
4.5.1 The Link-State (LS) Routing Algorithm
4.5.2 The Distance-Vector (DV) Routing Algorithm
4.5.3 Hierarchical Routing
4.6 Routing in the Internet
4.6.1 Intra-AS Routing in the Internet: RIP
4.6.2 Intra-AS Routing in the Internet: OSPF
4.6.3 Inter-AS Routing: BGP
4.7 Broadcast and Multicast Routing
4.7.1 Broadcast Routing Algorithms
4.7.2 Multicast
4.8 Summary
Homework Problems and Questions
Programming Assignments
Wireshark Labs: IP, ICMP
Interview: Vinton G. Cerf
Chapter 5 The Link Layer: Links, Access Networks, and LANs
5.1 Introduction to the Link Layer
5.1.1 The Services Provided by the Link Layer
5.1.2 Where Is the Link Layer Implemented?
5.2 Error-Detection and -Correction Techniques
5.2.1 Parity Checks
5.2.2 Checksumming Methods
5.2.3 Cyclic Redundancy Check (CRC)
5.3 Multiple Access Links and Protocols
5.3.1 Channel Partitioning Protocols
5.3.2 Random Access Protocols
5.3.3 Taking-Turns Protocols
5.3.4 DOCSIS: The Link-Layer Protocol for Cable Internet Access
5.4 Switched Local Area Networks
5.4.1 Link-Layer Addressing and ARP
5.4.2 Ethernet
5.4.3 Link-Layer Switches
5.4.4 Virtual Local Area Networks (VLANs)
5.5 Link Virtualization: A Network as a Link Layer
5.5.1 Multiprotocol Label Switching (MPLS)
5.6 Data Center Networking
5.7 Retrospective: A Day in the Life of a Web Page Request
5.7.1 Getting Started: DHCP, UDP, IP, and Ethernet
5.7.2 Still Getting Started: DNS and ARP
5.7.3 Still Getting Started: Intra-Domain Routing to the DNS Server
5.7.4 Web Client-Server Interaction: TCP and HTTP
5.8 Summary
Homework Problems and Questions
Wireshark Labs: Ethernet and ARP, DHCP
Interview: Simon S. Lam
Chapter 6 Wireless and Mobile Networks
6.1 Introduction
6.2 Wireless Links and Network Characteristics
6.2.1 CDMA
6.3 WiFi: 802.11 Wireless LANs
6.3.1 The 802.11 Architecture
6.3.2 The 802.11 MAC Protocol
6.3.3 The IEEE 802.11 Frame
6.3.4 Mobility in the Same IP Subnet
6.3.5 Advanced Features in 802.11
6.3.6 Personal Area Networks: Bluetooth and Zigbee
6.4 Cellular Internet Access
6.4.1 An Overview of Cellular Network Architecture
6.4.2 3G Cellular Data Networks: Extending the Internet to Cellular Subscribers
6.4.3 On to 4G: LTE
6.5 Mobility Management: Principles
6.5.1 Addressing
6.5.2 Routing to a Mobile Node
6.6 Mobile IP
6.7 Managing Mobility in Cellular Networks
6.7.1 Routing Calls to a Mobile User
6.7.2 Handoffs in GSM
6.8 Wireless and Mobility: Impact on Higher-Layer Protocols
6.9 Summary
Homework Problems and Questions
Wireshark Lab: IEEE 802.11 (WiFi)
Interview: Deborah Estrin
Chapter 7 Multimedia Networking
7.1 Multimedia Networking Applications
7.1.1 Properties of Video
7.1.2 Properties of Audio
7.1.3 Types of Multimedia Network Applications
7.2 Streaming Stored Video
7.2.1 UDP Streaming
7.2.2 HTTP Streaming
7.2.3 Adaptive Streaming and DASH
7.2.4 Content Distribution Networks
7.2.5 Case Studies: Netflix, YouTube, and Kankan
7.3 Voice-over-IP
7.3.1 Limitations of the Best-Effort IP Service
7.3.2 Removing Jitter at the Receiver for Audio
7.3.3 Recovering from Packet Loss
7.3.4 Case Study: VoIP with Skype
7.4 Protocols for Real-Time Conversational Applications
7.4.1 RTP
7.4.2 SIP
7.5 Network Support for Multimedia
7.5.1 Dimensioning Best-Effort Networks
7.5.2 Providing Multiple Classes of Service
7.5.3 Diffserv
7.5.4 Per-Connection Quality-of-Service (QoS) Guarantees: Resource Reservation and Call Admission
7.6 Summary
Homework Problems and Questions
Programming Assignment
Interview: Henning Schulzrinne
Chapter 8 Security in Computer Networks
8.1 What Is Network Security?
8.2 Principles of Cryptography
8.2.1 Symmetric Key Cryptography
8.2.2 Public Key Encryption
8.3 Message Integrity and Digital Signatures
8.3.1 Cryptographic Hash Functions
8.3.2 Message Authentication Code
8.3.3 Digital Signatures
8.4 End-Point Authentication
8.4.1 Authentication Protocol ap1.0
8.4.2 Authentication Protocol ap2.0
8.4.3 Authentication Protocol ap3.0
8.4.4 Authentication Protocol ap3.1
8.4.5 Authentication Protocol ap4.0
8.5 Securing E-Mail
8.5.1 Secure E-Mail
8.5.2 PGP
8.6 Securing TCP Connections: SSL
8.6.1 The Big Picture
8.6.2 A More Complete Picture
8.7 Network-Layer Security: IPsec and Virtual Private Networks
8.7.1 IPsec and Virtual Private Networks (VPNs)
8.7.2 The AH and ESP Protocols
8.7.3 Security Associations
8.7.4 The IPsec Datagram
8.7.5 IKE: Key Management in IPsec
8.8 Securing Wireless LANs
8.8.1 Wired Equivalent Privacy (WEP)
8.8.2 IEEE 802.11i
8.9 Operational Security: Firewalls and Intrusion Detection Systems
8.9.1 Firewalls
8.9.2 Intrusion Detection Systems
8.10 Summary
Homework Problems and Questions
Wireshark Lab: SSL
IPsec Lab
Interview: Steven M. Bellovin
Chapter 9 Network Management
9.1 What Is Network Management?
9.2 The Infrastructure for Network Management
9.3 The Internet-Standard Management Framework
9.3.1 Structure of Management Information: SMI
9.3.2 Management Information Base: MIB
9.3.3 SNMP Protocol Operations and Transport Mappings
9.3.4 Security and Administration
9.4 ASN.1
9.5 Conclusion
Homework Problems and Questions
Interview: Jennifer Rexford
References
Index
A
B
C
D
E
F
G
H
I
J
K
L
M
N
O
P
Q
R
S
T
U
V
W
X
Y
Z
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