Henry SinnreichAlan B. Johnston

Internet CommunicationsUsing SIP

Delivering VoIP and Multimedia Serviceswith Session Initiation Protocol

Second Edition

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Internet Communications Using SIP

Second Edition

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Henry SinnreichAlan B. Johnston

Internet CommunicationsUsing SIP

Delivering VoIP and Multimedia Serviceswith Session Initiation Protocol

Second Edition

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Internet Communications Using SIP: Delivering VoIP and Multimedia Services with SessionInitiation Protocol, Second EditionPublished byWiley Publishing, Inc.10475 Crosspoint BoulevardIndianapolis, IN 46256www.wiley.com

Copyright © 2006 by Wiley Publishing, Inc., Indianapolis, IndianaPublished simultaneously in CanadaISBN-13: 978-0-471-77657-4ISBN-10: 0-471-77657-2Manufactured in the United States of America10 9 8 7 6 5 4 3 2 12MA/QW/QX/QW/IN

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Library of Congress Cataloging-in-Publication DataSinnreich, Henry.Internet communications using SIP : delivering VoIP and multimedia services with Section Ini-

tiation Protocol / Henry Sinnreich, Alan B. Johnston. — 2nd ed.p. cm.

Includes index.ISBN-13: 978-0-471-77657-4 (cloth)ISBN-10: 0-471-77657-2 (cloth)1. Computer network protocols. 2. Internet telephony. 3. Multimedia systems. I. Title. TK5105.55.S56 2006621.3850285’4678—dc22

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Trademarks: Wiley, the Wiley logo, and related trade dress are trademarks or registered trade-marks of John Wiley & Sons, Inc. and/or its affiliates, in the United States and other countries,and may not be used without written permission. All other trademarks are the property of theirrespective owners. Wiley Publishing, Inc., is not associated with any product or vendor men-tioned in this book.

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www.wiley.com

We could not have written this book without the support of our forgivingspouses, Fabienne and Lisa, who held the fort while we were working on

SIP. And to both our family members shouting, “Your SIP phone is ringing.”

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Dr. Henry Sinnreich (Richardson, TX) is Chief Technology Officer at Pulver.com,a leading media company for VoIP and Internet communication services. Dr.Sinnreich has held engineering and executive positions at MCI where he wasan MCI fellow and has been involved in Internet and multimedia services formore than 12 years, including the development of the flagship MCI Advantageservice based on SIP. Henry Sinnreich is also a contributor to IETF standardsfor Internet communications in such areas as SIP telephony devices and usingRTP extensions for voice quality monitoring. He was awarded the title Pioneerfor VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been acofounder and board member of the International SIP Forum based in Stock-holm. He is a frequent speaker and is known as the leading evangelist, world-wide, for SIP based VoIP, presence, IM, multimedia, and integration ofapplications with communications. Dr. Sinnreich is also a guest lecturer at theEngineering School of the Southern Methodist University in Dallas, TX.

Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff atAvaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and fourother SIP related RFCs. He is the co-chair of the IETF Centralized ConferencingWorking Group and is on the board of directors of the International SIP Forum.His current areas of interest include peer-to-peer SIP and security. Dr. Johnstonis a frequent speaker and lecturer on SIP and contributor to various publications,and is an adjunct professor at Washington University in St. Louis, MO.

About the Authors

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Credits

ix

Executive EditorCarol Long

Development EditorKevin Shafer

Production EditorPamela Hanley

Copy EditorFoxxe Editorial Services

Editorial ManagerMary Beth Wakefield

Production ManagerTim Tate

Vice President and ExecutiveGroup Publisher

Richard Swadley

Vice President and Executive Publisher

Joseph B. Wikert

Project CoordinatorRyan Steffen

Graphics and ProductionSpecialists

Stephanie D. JumperHeather RyanAlicia B. South

Quality Control TechnicianLaura Albert

Proofreading and IndexingJoe NiesenPalmer Publishing Services

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Contents

xi

Foreword xxi

Acknowledgments xxiii

Introduction xxv

Chapter 1 Introduction 1Problem: Too Many Public Networks 1Incompatible Enterprise Communications 4Network Consolidation: The Internet 4Voice over IP 5Presence—The Dial Tone for the Twenty-First Century? 6The Value Proposition of SIP 6SIP Is Not a Miracle Protocol 6The Short History of SIP 7References in This Book 8SIP Open Source Code and SIP Products 9References for Telephony 10Summary 10References 10

Chapter 2 Internet Communications Enabled by SIP 11Internet Multimedia Protocols 12The Value of Signaling 13Protocols for Media Description, Media Transport, and other

Multimedia Delivery 14Addressing 15SIP in a Nutshell 15SIP Capabilities 17Overview of Services Provided by SIP Servers 18Peer-to-Peer SIP (P2PSIP) 19

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Caller Preferences 19Mobility in the Wider Concept 20

Global Telephone Number Portability 20SIP Application-Level Mobility 20

Context-Aware Communications: Presence and IM 21SIP Presence 21Instant Messaging 23The Integration of Communications with Applications 23

E-Commerce: Customer Relations Management 23Conferencing and Collaboration 24Telephony Call Control Services 25Intelligent Network Services Using SIP: ITU Services CS-1

and CS-2 25SIP Service Creation—Telephony-Style 26ENUM 27SIP Interworking with ITU-T Protocols 27Mixed Internet-PSTN Services 29

PSTN and INTerworking (PINT) 29SPIRITS 29TRIP 29

SIP Security 31SIP Accessibility to Communications for the Hearing and

Speech Disabled 31SIP Orphans 32Commercial SIP Products 32What SIP Does Not Do 33

Divergent Views on the Network 34Summary 35

References 35

Chapter 3 Architectural Principles of the Internet 39Telecom Architecture 39Internet Architecture 42The Internet Backbone Architecture 44The Internet Standards Process 48Protocols and Application Programming Interfaces 49

Is XML the Presentation Layer of the Internet Protocol Architecture? 50

Middle-Age Symptoms of the Internet 50Fighting Complexity 51Summary 52References 52

Chapter 4 DNS and ENUM 53Introduction 53Addressing on the Internet 54

The Universal Resource Identifier (URI) 54mailto: 55

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The Universal Resource Locator (URL) 55Tel URI 56

The phone-context 56SIP URI 57IANA ENUM Service Registrations 58The Domain Name System 58

Delegation 59Caching 59A Partial DNS Glossary 60

DNS and ENUM Usage Example 62Finding an Outgoing SIP Server 63Finding an Incoming SIP Server in the ENUM Case 64Call Setup Delay 67

DNS-Based Routing Service Using SIP 67SIP URI or Telephone Number? 67

The ENUM Functional Architecture 69ENUM and Number Portability 71Implementation Issues 71

DNS and SIP User Preferences 72Application Scenarios for SIP Service Using ENUM 73

PBX Enterprise Voice Network 74Enterprise System with IP Communications 74Residential User with ENUM Service 76

Miscellaneous: ENUM Lookup of the Display Name 76DNS and Security 77

Impersonation 77Eavesdropping 77Data Tampering 78Malicious Redirection 78Denial of Service 78

Summary 79References 79

Chapter 5 Real-Time Internet Multimedia 81Introduction 81Freshening Up on IP 83Multicast Protocols 85

Multicast Address Allocation 85Application-Level Multicast 86

Transport Protocols 86IP Network Layer Services 87

Differentiated Services 88Resource Reservation 88Integrated Services and DiffServ Networks 89Multiprotocol Label Switching 89

Media and Data Formats 90Media Transport Using RTP 91RTP Payloads and Payload Format Specifications 92

Contents xiii

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Multimedia Server Recording and Playback Control 93Session Description 93Session Announcements 93Session Invitation 93Authentication and Key Distribution 94Summary 94References 94

Chapter 6 SIP Overview 97What Makes SIP Special 97

SIP Enabled Network 98Watching How Sausages Are Being Made 101What SIP Is Not 102

Introduction to SIP 102Elements of a SIP Network 106

User Agents 106Servers 106Location Services 107

SIP Functions 107Address Resolution 108Session-Related Functions 110

Session Setup 110Media Negotiation 111Session Modification 114Session Termination and Cancellation 116Mid-Call Signaling 117Call Control 118Preconditions Call Setup 121

Nonsession-Related Functions 123Mobility 124Message Transport 126Event Subscription and Notification 127Presence Publication 128Authentication Challenges 128Extensibility 130

Summary 132References 132

Chapter 7 SIP Service Creation 135Services in SIP 135

Service Example 136Server Implementation 136Called User Agent Implementation 137Calling User Agent Implementation 138Comparison 140

New Methods and Headers 141Service Creation Options 142

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Call Processing Language 142Introduction to CPL 142Example of CPL Scripts 146

SIP Common Gateway Interface 147SIP Application Programming Interfaces 148

SIP Servlets 149JAIN 149

SIP and VoiceXML 149Summary 150References 150

Chapter 8 User Preferences 153Introduction 153Preferences of Caller 154

Example for Contact 156Example for Accept-Contact 156Example for Reject-Contact 156

Preferences of the Called Party 157Server Support for User Preferences and for Policies 157Summary 157References 158

Chapter 9 SIP Security 159Threats 159

Session Setup 160Presence and IM 161

Security Mechanisms 162Authentication 162Confidentiality 163Secure SIP URI Scheme 164Integrity 165Identity 165

Media Security 166SRTP 166MIKEY 167SDP Security Descriptions 167

New Directions 168DTLS 169ZRTP 169

Summary 169References 170

Chapter 10 NAT and Firewall Traversal 173Network Address Translators 174Firewalls 177

STUN, TURN, and ICE 179Application Layer Gateways 180Privacy Considerations 183

Contents xv

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Summary 184References 184

Chapter 11 SIP Telephony 185Basic Telephony Services 185

SIP and PSTN Interworking 185Gateway Location and Routing 186

SIP/PSTN Protocol Interworking 187Types of Gateways 188SIP and Early Media 188SIP Telephony and ISUP Tunneling 190

Enhanced Telephony Services 196Call Control Services and Third-Party Call Control 199

Problem Statement 199The REFER Method 201

SIP Third-Party Call Control 202Basic Third-Party Call Control 203Security for Third-Party Call Control 203Peer-to-Peer Third-Party Call Control 205

Summary 206References 207

Chapter 12 Voicemail and Universal Messaging 209Problem Statement for Unified Messaging 209Architecture and Operation 211

RTSP-Enabled Voice Message Retrieval 212Depositing of Voice Messages 214Notification for Waiting Messages 217

Simple Message Notification Format 217Rich Message Notification Format 220

Retrieval of Messages 221Summary 221References 221

Chapter 13 Presence and Instant Messaging 223The Potential of SIP Presence, Events, and IM 224The Evolution of IM and Presence 225The IETF Model for Presence and IM 226Client Server and Peer-to-Peer Presence and IM 228SIP Event-Based Communications and Applications 229

Presence Event Package 231Presence Information Data Format 233

The Data Model for Presence 235Indication of Message Composition for IM 236Rich Presence Information 236SIP Extensions for Instant Messaging 239Summary 241References 242

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Chapter 14 SIP Conferencing 245Introduction 245SIP Conferencing Models 246

Ad Hoc and Scheduled Conferences 249Changing the Nature of a Conference 249

Centralized Conferencing 251Summary 251References 251

Chapter 15 SIP Application Level Mobility 253Mobility in Different Protocol Layers 254Dimensions of Mobility 255Examples of SIP Application-Layer Mobility 256SIP Network-Based Fixed-Mobile Convergence 261SIP Device-Based Fixed-Mobile Convergence 263SIP Application-Layer Mobility and Mobile IP 263Multimodal Mobile Device Technology and Issues 265

Network Control versus User Control of Mobility 266IEEE 802.21 Media-Independent Handover (MIH) 267Network Selection Issues 269

Summary 270References 270

Chapter 16 Emergency and Preemption Communication Services 273Requirements 274Location Information 275

Types of Location Information 275Sources of Location Information 275DNS-Based Location Information 275

Internet-Based Emergency Calling 277Identifying an Internet Emergency Call: The SOS URI 278Internet Emergency Call Routing 278Security for Emergency Call Services 279

Using the PSTN for VoIP Emergency Calls 280Emergency Communication Services 281

Emergency Call Preemption Using SIP 282Linking SIP Preemption to IP Network and Link Layer

Preemption 284Summary 285References 285

Chapter 17 Accessibility for the Disabled 287About Accessibility 287Accessibility on Legacy Networks and on the Internet 288Requirements for Accessibility 289Text over IP (ToIP) 290

Performance Metrics for ToIP 293

Contents xvii

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Transcoding Services 294Transcoding Scenarios 294Call Control Models for Transcoding Services 296

Summary 298References 299

Chapter 18 Quality of Service for Real-Time Internet Communications 301Voice Quality Metrics 303

Delay Limits for Voice 303Burst vs. Average Packet Loss 304

Acoustics and the Network 304Internet Codecs 305Codecs in Wireless Networks and Transcoding 307

Codec Bandwidth 307The Endpoint Quality for Voice 308The Internet Performance 308

Concerns Regarding Congestion Control 309Internet Traffic Statistics: Voice Is Negligible 309

A Summary of Internet QoS Technologies 311Best Effort Is for the Best Reasons 313Monitoring QoS for Real-Time Communications 314Summary 315References 315

Chapter 19 SIP Component Services 317Master/Slave VoIP Systems 318IP Telephony Gateways 320The Converged Applications Environment 323The Control of Service Context 326

Voicemail 328Collecting DTMF Digits 330Interactive Voice Response System 333Scheduled Conference Service 335

Summary 337References 337

Chapter 20 Peer-to-Peer SIP 339Definitions for P2P Networks 340

Overlay Networks 340Peer-to-Peer Networks 341Distributed Hash Tables (DHTs) 342

Characteristics of P2P Computing 344Security of P2P Networks 344The Chord Protocol 345P2P SIP 346

CS SIP Model 347P2P SIP Model 348

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Use Cases for P2P SIP 348Disruption of the VoIP Infrastructure Model 349Summary 350References 351

Chapter 21 Conclusions and Future Directions 353Short Term Challenges 355Future Services: The Internet Is the Service 355Still to Develop: Peer-to-Peer SIP Standards 355Prediction: The Long Road Ahead 356Summary 356References 356

Index 357

Contents xix

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About 10 years ago, the first drafts describing the Session Initiation Protocol(1996) were published, with the rather modest ambition of setting up multicastgroups for multimedia conferences. In the intervening decade, a draft of about20 pages has turned into an ecosystem of dozens of RFCs, hundreds of Inter-net drafts—and several books, conferences, and a magazine. It has become dif-ficult to get a feel for the overall landscape, to distinguish the important coreconcepts from the niche applications. This book offers a detailed, technicallyinformed, yet accessible, introduction to the overall SIP ecosystem, suitableboth for someone who needs to understand the technology to make strategicdecisions and implementers who need to build new components.

SIP is part of the second wave of Internet application protocol. While thefirst wave largely focused on asynchronous communications (such as e-mail,and data transfer), this second wave introduces the notion of interactive,human-to-human communication that allows integration with any media, notjust voice. As SIP and interactive communications have matured, the goal forhuman-to-human communication has shifted. Initially, cell phones promisedvoice communication at any time, at any place. Multimedia communications,on PCs and maybe emerging cellular networks, allow us to add “any media.”However, the “any time, any place, any media” can also turn us into slaves ofour communications devices, interrupting our ability to think, to eat in peace,and to meet in person. Thus, our goal has to be to design communicationstechnology that offers the right media, at the right place, and at the right time.With some of the advanced functionality of SIP, such as presence, location-based services, user-created services, and caller preferences, we can get closerto creating communication systems that support our work and enhance ourpersonal life.

Foreword

xxi

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With new communications technologies, there is always the temptation tomimic the old. E-mail inherited aspects of the interoffice memo and fax; webpages attempted to look like newsprint and brochures. However, in VoIP, thereis the particular temptation to recreate old technology features, as interoper-ability with the old PSTN will remain important for at least another decade.Fax-to-email gateways were never quite as important as VoIP-to-PSTN gate-ways. This emphasis on interoperability with 100-year-old technology hasprovided a financial motivation—provide the same service more cheaply.However, this may also hold back the promise offered by Internet-based mul-timedia communications, such as the integration of presence, the ability notjust to communicate by voice and maybe video but also to share any applica-tion, or the ability to customize the user experience and integrate interactivecommunications with existing Internet tools and applications. Just as mostmicroprocessors are embedded in household appliances and cars, not desktopPCs and laptops, we might find that Internet-based voice and multimediacommunications will be integrated into games, appliances, and cameras, or behidden behind a link on a web page, rather than dialed by name or number. Asfor many of the most innovative applications, users will likely not even con-sider them phone services at all, but extensions that make some other applica-tion more productive or more fun.

This book is like a good tour guide to a foreign country. It doesn’t justdescribe the major sites and tourist attractions; it lets the reader share in thehistory, spirit, language, and culture of the place. Natives write the best tourguides, and the authors have been living and working in SIP land since it wasa small outpost in one large country called the IETF. The authors have servedas ambassadors in lands near and far, but have also made major contributionsto the development of this part of the Internet landscape, always remindingothers of the original goals of the first inhabitants. After taking the tour, thereader will be ready not just to show off a stamp on a passport or certificate butalso to contribute to new modes of communications. SIP land is still young andneeds lots of pioneers who can push the frontiers of Internet-enabled commu-nications. There might not always be gold in those hills, but enriching humancommunications will always be its own reward.

Henning SchulzrinneProfessor, Columbia University

xxii Foreword

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Acknowledgments

xxiii

We have enjoyed the benefit of early and significant support from colleaguesand management in MCI. Vint Cerf was, as mentioned, one of the early sup-porters, and so were Teresa Hastings, John Gallant, Bob Spry, and RobertOliver who first took the responsibility for developing and deploying SIP intheir respective engineering departments. John Truetken, Lance Lockhart, andmany other engineers in MCI also had critical contributions to the implemen-tation of SIP. Fred Briggs, Patrice Carroll, Barry Zip, and Leo Cyr from MCIhelped with the challenge to develop marketable services based on SIP. Wewere fortunate to work jointly in the development and deployment of SIP ser-vices with Steve Donovan, Diana Rawlins, Dean Willis, Robert Sparks, BenCampbell, Chris Cunningham, Kevin Summers, and many other engineersfrom MCI and elsewhere in the industry engaged in the development of SIP inthe Internet Engineering Task Force (IETF).

Most ideas and inspirations driving SIP are due to Prof. HenningSchulzrinne from Columbia University and to Jonathan Rosenberg fromDynamicSoft and are reflected in this book. Among the many industry con-tributors, we gratefully acknowledge discussions and guidance from RohanMahy from Cisco Corporation, Gonzalo Camarillo and Adam Roach fromL.M. Ericsson. Jiri Kuthan from GMD Focus, Berlin, was helpful with SIP tuto-rial charts and with discussions in transatlantic calls using SIP phones—again,calls of crystal clear clarity to our surprise. The authors are grateful to RichardShockey from NeuStar, Inc. and Douglas Ranalli from NetNumber, Inc. fornumerous discussions regarding ENUM. Theodore Havinis has contributed tothe SIP-QoS-AAA aspect for mobile users.

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We acknowledge countless helpful discussions and insight from many par-ticipants in the IETF and especially to Scott Bradner for holding the authorsand others in the IETF SIP community in line to the true conceptual, technical,and procedural spirit of the Internet.

Jeff Pulver has played a special role in providing a platform and leadingexhibition of products for what was initially an obscure and unknown proto-col in the Voice ON the Net (VON) and other conferences held in America,Europe, and Asia.

Carrol Long, Kevin Shafer, and Adoabi Obi Tulton from John Wiley & Sonshave been instrumental in editing this book.

xxiv Acknowledgments

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Introduction

xxv

The second edition of Internet Communications Using SIP had to be rewrittenalmost from the ground up, because of the dramatic changes in the industry inthe five years that have passed since the first edition. Some of the developmentshad been envisaged in the first edition, but naturally, some have not.

The Internet Has Replaced the Telephone System andthe Telecommunication NetworksSince the publication in 2001 of the first edition of this book, Internet Commu-nications Using SIP, Voice over IP (VoIP) has developed from an emerging tech-nology to the recognized replacement of existing global telephone systemsbased on Time Division Multiplex (TDM) circuit switching. The Internet hasalso replaced the proposed connection-oriented offsprings of TDM, such as theIntegrated Services Digital Network (ISDN) and the Asynchronous TransferMultiplex (ATM) based broadband version BISDN, envisaged for the telecom-munications industry by the International Telecommunications Union ITU-Tstandards body. TDM, ATM, ISDN, and BISDN are now history.

All wired and wireless communications are instead migrating to the Internetstandards developed by the Internet Engineering Task Force (IETF). The legacytelecommunication networks, while still dominant, are recognized as a present-day cash cow only and are scheduled for replacement by IP networks.

The end-to-end nature of the Internet that places intelligence in the applica-tions running in the endpoints and gives control to the user at the endpointshas indeed replaced TDM-based telephony with central control. The Internet

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has also proven to be the home network for other types of communications,information, entertainment, and data applications. To quote Jon Peterson, areadirector of the IETF:

“The Internet is the service.”

The Session Initiation Protocol Is the Standard for VoIP and Multimedia CommunicationsAnother change from the first edition of this book is the Session Initiation Pro-tocol (SIP), which has been adopted by practically all public VoIP serviceproviders for wired and wireless communications. The discussions about SIPversus H.323 standardized by the ITU-T are over as well. The installed base ofH.323 is considered a liability and planned for replacement by SIP sooner orlater.

A global industry has emerged to take advantage of SIP and its associatedIETF standards for real-time communications. More than 560 VoIP serviceproviders have been reported [1] in early 2006, most of them using SIP-basednetworks. The list of SIP-based equipment (such as SIP phones, software forPCs, and mobile devices, servers, gateways, and so on) is now large and stillgrowing. Actually, all equipment and system vendors are now supporting SIP.

Presence and Instant Messaging Are Mainstream CommunicationsPresence and instant messaging (IM) are now mainstream with consumersand, in the enterprise, complementing or sometimes replacing voice commu-nications in specific situations (such as in circumstances where silence isrequired). Even for VoIP, presence has emerged not only as a valuableenhancement, but presence may be the dial tone of the twenty-first century.

Presence and event-based communications have enabled the integration ofcommunications with applications. Presence and IM are discussed in Chapter13, “Presence and Instant Messaging.”

The so-called IM services provided by large Internet companies, such asAOL, Apple, Google, IBM, Microsoft, Skype (not SIP-based), and Yahoo!, actu-ally carry at present most of the public VoIP traffic between end users aroundthe globe.

It is not far-fetched to see the IM Internet companies replacing the formertelephone companies in the voice communication business. Many legacytelecommunication companies are also using VoIP to replace the internal TDMvoice networks, but their VoIP services may not survive the advanced tech-nologies deployed by the IM Internet companies and the challenge posed bypeer-to-peer (P2P) communications.

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Redefining Communications: Mobility, Emergency andEqual Access for the DisabledInternet communications have been known not to be dependent on the loca-tion on the Internet. Application-level mobility based on SIP is a key compo-nent to seamless mobile communications, as discussed in Chapter 15, “SIPApplication Level Mobility.”

Emergency calling services by users in distress using the Internet (such as911 in the United States or 112 in Europe) are far more powerful and cost lessthan the Public Switched Telephone Network (PSTN) based emergency ser-vices. Internet-based emergency calling is indeed in the design stage in a num-ber of countries. Chapter 16, “Emergency and Preemption CommunicationServices,” discusses Internet-based emergency services.

The multimedia nature of Internet communications gives hearing- andspeech-impaired people the opportunity to fully participate in rich communi-cations for work and in personal life. Chapter 17, “Accessibility for the Dis-abled,” discusses access to communications for disabled people.

The Rise of Peer-to-Peer CommunicationsP2P traffic has risen in the Internet since around 2000 and became the domi-nant part of Internet traffic by 2004. Since 2004, Skype (which is based on P2PVoIP, IM, and presence) has also become by far the dominant VoIP providerworldwide. Since P2P SIP standards work is just emerging as of this writing,Skype can be considered a prestandard P2P Internet communication service.

The reasons for the emergence of overlay networks and P2P applicationsand their nature are discussed in Chapter 20, “Peer-to-Peer SIP,” and also inChapter 6, “SIP Overview.” Though the present VoIP industry is built onclient-server (CS) SIP, this may significantly change. To quote David Bryanfrom p2p.org:

“P2P SIP may change VoIP to the same extent that VoIP has changed telecom-munications.”

VoIP and Multimedia Communications Services Are StillFragmentedIn spite of all the technological progress, VoIP, IM, presence, and multimediaservices are still a highly fragmented industry:

■■ Telephone services based on VoIP operate as islands and can intercon-nect (as of this writing) using mostly the legacy Public Switched Tele-phone Network (PSTN). The service model is giving broadband users

Introduction xxvii

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access to the legacy telephone system, actually a voice gateway servicebetween the Internet and TDM. The business model of most VoIP ser-vice providers is just lower cost for legacy-style telephone service, alsocalled PSTN over IP. The PSTN gateway services are using IP insidetheir networks, but users are not exposed to the rich IP services, exceptwhen all parties are on the same network.

■■ The most successful public voice, IM, and presence service is Skype,which is not standards-based.

■■ Walled gardens: The fragmentation of communications is still activelypursued by most mobile service providers by deploying systems wheretheir users can get rich IP multimedia services only on their own net-works. The fees to communicate between mobile service providers are asignificant part of the business model, and open connectivity to theInternet (“Internet neutrality”) is still a hotly debated issue. Internetneutrality is also still debated by many broadband Internet accessproviders (such as DSL and cable companies), although we believe thatenlightened government regulators in the developed countries willweigh in favor of users and open network access in general.

The proliferation of islands for communications makes them less useful themore there are, since this proliferation is in denial of Metcalf’s law that thevalue of a network increases with the square of the number of points attachedto the network. The Internet with more than 1 billion attached endpoints hasthus the highest value for communications. By contrast, the mobile phoneindustry boasts 3 billion users, but in many fragmented networks.

Past Obsessions and Present Dangers: QoS and Security Network-based quality of service (QoS) for voice and the reliability of thelegacy telephone network have long been used by telephone industry mar-keters to scare users away from VoIP. In the meantime, all public VoIP serviceshave proven that Internet best-effort QoS works just fine, as long network con-gestion is avoided. Internet-based voice can actually be much better than the3.1 kHz voice over the PSTN. As for reliability, all recent major man-made andnatural disasters have proven the Internet and VoIP to be more resilient thanthe existing wireline and wireless telephone networks.

Chapter 18, “Quality of Service for Real-Time Internet Communications,” isaimed at a balanced approach for QoS, and Chapter 16, “Emergency and Pre-emption Communication Services,” discusses the Emergency Services basedon SIP.

xxviii Introduction

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