Internet Communications Using SIP

Delivering VoIP and Multimedia Services ... cofounder and board member of the International SIP Forum based in ... Session Termination and Cancellatio...

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Internet Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation Protocol Second Edition

Henry Sinnreich Alan B. Johnston

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Internet Communications Using SIP Second Edition

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Internet Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation Protocol Second Edition

Henry Sinnreich Alan B. Johnston

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Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol, Second Edition Published by Wiley Publishing, Inc. 10475 Crosspoint Boulevard Indianapolis, IN 46256 www.wiley.com Copyright © 2006 by Wiley Publishing, Inc., Indianapolis, Indiana Published simultaneously in Canada ISBN-13: 978-0-471-77657-4 ISBN-10: 0-471-77657-2 Manufactured in the United States of America 10 9 8 7 6 5 4 3 2 1 2MA/QW/QX/QW/IN No part of this publication may be reproduced, stored in a retrieval system or transmitted in any form or by any means, electronic, mechanical, photocopying, recording, scanning or otherwise, except as permitted under Sections 107 or 108 of the 1976 United States Copyright Act, without either the prior written permission of the Publisher, or authorization through payment of the appropriate per-copy fee to the Copyright Clearance Center, 222 Rosewood Drive, Danvers, MA 01923, (978) 750-8400, fax (978) 646-8600. Requests to the Publisher for permission should be addressed to the Legal Department, Wiley Publishing, Inc., 10475 Crosspoint Blvd., Indianapolis, IN 46256, (317) 572-3447, fax (317) 572-4355, or online at http://www.wiley.com/go/permissions. Limit of Liability/Disclaimer of Warranty: The publisher and the author make no representations or warranties with respect to the accuracy or completeness of the contents of this work and specifically disclaim all warranties, including without limitation warranties of fitness for a particular purpose. No warranty may be created or extended by sales or promotional materials. The advice and strategies contained herein may not be suitable for every situation. This work is sold with the understanding that the publisher is not engaged in rendering legal, accounting, or other professional services. If professional assistance is required, the services of a competent professional person should be sought. Neither the publisher nor the author shall be liable for damages arising herefrom. The fact that an organization or Website is referred to in this work as a citation and/or a potential source of further information does not mean that the author or the publisher endorses the information the organization or Website may provide or recommendations it may make. Further, readers should be aware that Internet Websites listed in this work may have changed or disappeared between when this work was written and when it is read. For general information on our other products and services or to obtain technical support, please contact our Customer Care Department within the U.S. at (800) 762-2974, outside the U.S. at (317) 572-3993 or fax (317) 572-4002. Library of Congress Cataloging-in-Publication Data Sinnreich, Henry. Internet communications using SIP : delivering VoIP and multimedia services with Section Initiation 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 2006 621.3850285’4678—dc22 2006009325 Trademarks: Wiley, the Wiley logo, and related trade dress are trademarks or registered trademarks 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 their respective owners. Wiley Publishing, Inc., is not associated with any product or vendor mentioned in this book. Wiley also publishes its books in a variety of electronic formats. Some content that appears in print may not be available in electronic books.

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We could not have written this book without the support of our forgiving spouses, 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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About the Authors

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 was an MCI fellow and has been involved in Internet and multimedia services for more than 12 years, including the development of the flagship MCI Advantage service based on SIP. Henry Sinnreich is also a contributor to IETF standards for Internet communications in such areas as SIP telephony devices and using RTP extensions for voice quality monitoring. He was awarded the title Pioneer for VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been a cofounder and board member of the International SIP Forum based in Stockholm. He is a frequent speaker and is known as the leading evangelist, worldwide, for SIP based VoIP, presence, IM, multimedia, and integration of applications with communications. Dr. Sinnreich is also a guest lecturer at the Engineering School of the Southern Methodist University in Dallas, TX. Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff at Avaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and four other SIP related RFCs. He is the co-chair of the IETF Centralized Conferencing Working 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. Johnston is a frequent speaker and lecturer on SIP and contributor to various publications, and is an adjunct professor at Washington University in St. Louis, MO.

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Credits

Executive Editor Carol Long

Project Coordinator Ryan Steffen

Development Editor Kevin Shafer

Graphics and Production Specialists Stephanie D. Jumper Heather Ryan Alicia B. South

Production Editor Pamela Hanley Copy Editor Foxxe Editorial Services Editorial Manager Mary Beth Wakefield Production Manager Tim Tate

Quality Control Technician Laura Albert Proofreading and Indexing Joe Niesen Palmer Publishing Services

Vice President and Executive Group Publisher Richard Swadley Vice President and Executive Publisher Joseph B. Wikert

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Contents

Foreword

xxi

Acknowledgments Introduction

xxiii xxv

Chapter 1

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

1 1 4 4 5 6 6 6 7 8 9 10 10 10

Chapter 2

Internet Communications Enabled by SIP Internet Multimedia Protocols The Value of Signaling Protocols for Media Description, Media Transport, and other Multimedia Delivery Addressing SIP in a Nutshell SIP Capabilities Overview of Services Provided by SIP Servers Peer-to-Peer SIP (P2PSIP)

11 12 13 14 15 15 17 18 19 xi

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Contents Caller Preferences Mobility in the Wider Concept Global Telephone Number Portability SIP Application-Level Mobility

Context-Aware Communications: Presence and IM

21 21 23 23

E-Commerce: Customer Relations Management Conferencing and Collaboration Telephony Call Control Services Intelligent Network Services Using SIP: ITU Services CS-1 and CS-2 SIP Service Creation—Telephony-Style ENUM SIP Interworking with ITU-T Protocols Mixed Internet-PSTN Services

23 24 25

SIP Security SIP Accessibility to Communications for the Hearing and Speech Disabled SIP Orphans Commercial SIP Products What SIP Does Not Do Divergent Views on the Network

Summary References

Architectural Principles of the Internet Telecom Architecture Internet Architecture The Internet Backbone Architecture The Internet Standards Process Protocols and Application Programming Interfaces Is XML the Presentation Layer of the Internet Protocol Architecture?

Chapter 4

20 20

SIP Presence Instant Messaging The Integration of Communications with Applications

PSTN and INTerworking (PINT) SPIRITS TRIP

Chapter 3

19 20

25 26 27 27 29 29 29 29

31 31 32 32 33 34

35 35

39 39 42 44 48 49 50

Middle-Age Symptoms of the Internet Fighting Complexity Summary References

50 51 52 52

DNS and ENUM Introduction Addressing on the Internet

53 53 54

The Universal Resource Identifier (URI) mailto:

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Contents The Universal Resource Locator (URL) Tel URI The phone-context SIP URI IANA ENUM Service Registrations The Domain Name System Delegation Caching A Partial DNS Glossary DNS and ENUM Usage Example Finding an Outgoing SIP Server Finding an Incoming SIP Server in the ENUM Case Call Setup Delay DNS-Based Routing Service Using SIP SIP URI or Telephone Number?

The ENUM Functional Architecture

69

ENUM and Number Portability Implementation Issues

71 71

DNS and SIP User Preferences Application Scenarios for SIP Service Using ENUM PBX Enterprise Voice Network Enterprise System with IP Communications Residential User with ENUM Service Miscellaneous: ENUM Lookup of the Display Name

DNS and Security Impersonation Eavesdropping Data Tampering Malicious Redirection Denial of Service

Chapter 5

55 56 56 57 58 58 59 59 60 62 63 64 67 67 67

72 73 74 74 76 76

77 77 77 78 78 78

Summary References

79 79

Real-Time Internet Multimedia Introduction Freshening Up on IP Multicast Protocols

81 81 83 85

Multicast Address Allocation Application-Level Multicast

85 86

Transport Protocols IP Network Layer Services Differentiated Services Resource Reservation Integrated Services and DiffServ Networks Multiprotocol Label Switching

Media and Data Formats Media Transport Using RTP RTP Payloads and Payload Format Specifications

86 87 88 88 89 89

90 91 92

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Contents

Chapter 6

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

93 93 93 93 94 94 94

SIP Overview What Makes SIP Special

97 97

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

Introduction to SIP Elements of a SIP Network User Agents Servers Location Services

SIP Functions Address Resolution Session-Related Functions Session Setup Media Negotiation Session Modification Session Termination and Cancellation Mid-Call Signaling Call Control Preconditions Call Setup Nonsession-Related Functions Mobility Message Transport Event Subscription and Notification Presence Publication Authentication Challenges Extensibility

Chapter 7

98 101 102

102 106 106 106 107

107 108 110 110 111 114 116 117 118 121 123 124 126 127 128 128 130

Summary References

132 132

SIP Service Creation Services in SIP

135 135

Service Example Server Implementation Called User Agent Implementation Calling User Agent Implementation Comparison

New Methods and Headers Service Creation Options

136 136 137 138 140

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Contents Call Processing Language Introduction to CPL Example of CPL Scripts SIP Common Gateway Interface SIP Application Programming Interfaces SIP Servlets JAIN

Chapter 8

SIP and VoiceXML Summary References

149 150 150

User Preferences Introduction Preferences of Caller

153 153 154

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

Chapter 9

142 142 146 147 148 149 149

156 156 156

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

157 157 157 158

SIP Security Threats

159 159

Session Setup Presence and IM

160 161

Security Mechanisms

162

Authentication Confidentiality Secure SIP URI Scheme Integrity Identity

Media Security SRTP MIKEY SDP Security Descriptions

New Directions

162 163 164 165 165

166 166 167 167

168

DTLS ZRTP

169 169

Summary References

169 170

Chapter 10 NAT and Firewall Traversal Network Address Translators Firewalls

173 174 177

STUN, TURN, and ICE Application Layer Gateways Privacy Considerations

179 180 183

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Contents Summary References

184 184

Chapter 11 SIP Telephony Basic Telephony Services

185 185

SIP and PSTN Interworking Gateway Location and Routing SIP/PSTN Protocol Interworking Types of Gateways SIP and Early Media SIP Telephony and ISUP Tunneling

Enhanced Telephony Services Call Control Services and Third-Party Call Control Problem Statement The REFER Method SIP Third-Party Call Control Basic Third-Party Call Control Security for Third-Party Call Control Peer-to-Peer Third-Party Call Control

Summary References Chapter 12 Voicemail and Universal Messaging Problem Statement for Unified Messaging Architecture and Operation RTSP-Enabled Voice Message Retrieval

Depositing of Voice Messages Notification for Waiting Messages Simple Message Notification Format Rich Message Notification Format

Retrieval of Messages Summary References

185 186 187 188 188 190

196 199 199 201 202 203 203 205

206 207 209 209 211 212

214 217 217 220

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

223 224 225 226 228 229

Presence Event Package Presence Information Data Format

231 233

The Data Model for Presence Indication of Message Composition for IM Rich Presence Information SIP Extensions for Instant Messaging Summary References

235 236 236 239 241 242

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Contents Chapter 14 SIP Conferencing Introduction SIP Conferencing Models Ad Hoc and Scheduled Conferences Changing the Nature of a Conference

Centralized Conferencing Summary References

245 245 246 249 249

251 251 251

Chapter 15 SIP Application Level Mobility Mobility in Different Protocol Layers Dimensions of Mobility Examples of SIP Application-Layer Mobility SIP Network-Based Fixed-Mobile Convergence SIP Device-Based Fixed-Mobile Convergence SIP Application-Layer Mobility and Mobile IP Multimodal Mobile Device Technology and Issues

253 254 255 256 261 263 263 265

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

266 267 269

Summary References Chapter 16 Emergency and Preemption Communication Services Requirements Location Information

270 270 273 274 275

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

275 275 275

Internet-Based Emergency Calling

277

Identifying an Internet Emergency Call: The SOS URI Internet Emergency Call Routing Security for Emergency Call Services

Using the PSTN for VoIP Emergency Calls Emergency Communication Services Emergency Call Preemption Using SIP Linking SIP Preemption to IP Network and Link Layer Preemption

Summary References Chapter 17 Accessibility for the Disabled About Accessibility Accessibility on Legacy Networks and on the Internet Requirements for Accessibility Text over IP (ToIP) Performance Metrics for ToIP

278 278 279

280 281 282 284

285 285 287 287 288 289 290 293

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xviii Contents Transcoding Services Transcoding Scenarios Call Control Models for Transcoding Services

Summary References Chapter 18 Quality of Service for Real-Time Internet Communications Voice Quality Metrics Delay Limits for Voice Burst vs. Average Packet Loss

Acoustics and the Network Internet Codecs Codecs in Wireless Networks and Transcoding Codec Bandwidth

The Endpoint Quality for Voice The Internet Performance

294 294 296

298 299 301 303 303 304

304 305 307 307

308 308

Concerns Regarding Congestion Control Internet Traffic Statistics: Voice Is Negligible

309 309

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

311 313 314 315 315

Chapter 19 SIP Component Services Master/Slave VoIP Systems IP Telephony Gateways The Converged Applications Environment The Control of Service Context Voicemail Collecting DTMF Digits Interactive Voice Response System Scheduled Conference Service

Summary References Chapter 20 Peer-to-Peer SIP Definitions for P2P Networks Overlay Networks Peer-to-Peer Networks Distributed Hash Tables (DHTs)

Characteristics of P2P Computing Security of P2P Networks The Chord Protocol P2P SIP CS SIP Model P2P SIP Model

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337 337 339 340 340 341 342

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Contents Use Cases for P2P SIP Disruption of the VoIP Infrastructure Model Summary References

348 349 350 351

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

353 355 355 355 356 356 356

Index

357

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Foreword

About 10 years ago, the first drafts describing the Session Initiation Protocol (1996) were published, with the rather modest ambition of setting up multicast groups for multimedia conferences. In the intervening decade, a draft of about 20 pages has turned into an ecosystem of dozens of RFCs, hundreds of Internet drafts—and several books, conferences, and a magazine. It has become difficult to get a feel for the overall landscape, to distinguish the important core concepts from the niche applications. This book offers a detailed, technically informed, yet accessible, introduction to the overall SIP ecosystem, suitable both for someone who needs to understand the technology to make strategic decisions and implementers who need to build new components. SIP is part of the second wave of Internet application protocol. While the first 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, not just voice. As SIP and interactive communications have matured, the goal for human-to-human communication has shifted. Initially, cell phones promised voice 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 of our communications devices, interrupting our ability to think, to eat in peace, and to meet in person. Thus, our goal has to be to design communications technology 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, locationbased services, user-created services, and caller preferences, we can get closer to creating communication systems that support our work and enhance our personal life.

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Foreword

With new communications technologies, there is always the temptation to mimic the old. E-mail inherited aspects of the interoffice memo and fax; web pages attempted to look like newsprint and brochures. However, in VoIP, there is the particular temptation to recreate old technology features, as interoperability 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 gateways. This emphasis on interoperability with 100-year-old technology has provided a financial motivation—provide the same service more cheaply. However, this may also hold back the promise offered by Internet-based multimedia communications, such as the integration of presence, the ability not just to communicate by voice and maybe video but also to share any application, or the ability to customize the user experience and integrate interactive communications with existing Internet tools and applications. Just as most microprocessors are embedded in household appliances and cars, not desktop PCs and laptops, we might find that Internet-based voice and multimedia communications will be integrated into games, appliances, and cameras, or be hidden behind a link on a web page, rather than dialed by name or number. As for many of the most innovative applications, users will likely not even consider them phone services at all, but extensions that make some other application more productive or more fun. This book is like a good tour guide to a foreign country. It doesn’t just describe the major sites and tourist attractions; it lets the reader share in the history, spirit, language, and culture of the place. Natives write the best tour guides, and the authors have been living and working in SIP land since it was a small outpost in one large country called the IETF. The authors have served as ambassadors in lands near and far, but have also made major contributions to the development of this part of the Internet landscape, always reminding others of the original goals of the first inhabitants. After taking the tour, the reader will be ready not just to show off a stamp on a passport or certificate but also to contribute to new modes of communications. SIP land is still young and needs lots of pioneers who can push the frontiers of Internet-enabled communications. There might not always be gold in those hills, but enriching human communications will always be its own reward. Henning Schulzrinne Professor, Columbia University

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Acknowledgments

We have enjoyed the benefit of early and significant support from colleagues and management in MCI. Vint Cerf was, as mentioned, one of the early supporters, and so were Teresa Hastings, John Gallant, Bob Spry, and Robert Oliver who first took the responsibility for developing and deploying SIP in their respective engineering departments. John Truetken, Lance Lockhart, and many other engineers in MCI also had critical contributions to the implementation of SIP. Fred Briggs, Patrice Carroll, Barry Zip, and Leo Cyr from MCI helped with the challenge to develop marketable services based on SIP. We were fortunate to work jointly in the development and deployment of SIP services with Steve Donovan, Diana Rawlins, Dean Willis, Robert Sparks, Ben Campbell, Chris Cunningham, Kevin Summers, and many other engineers from MCI and elsewhere in the industry engaged in the development of SIP in the Internet Engineering Task Force (IETF). Most ideas and inspirations driving SIP are due to Prof. Henning Schulzrinne from Columbia University and to Jonathan Rosenberg from DynamicSoft and are reflected in this book. Among the many industry contributors, we gratefully acknowledge discussions and guidance from Rohan Mahy from Cisco Corporation, Gonzalo Camarillo and Adam Roach from L.M. Ericsson. Jiri Kuthan from GMD Focus, Berlin, was helpful with SIP tutorial charts and with discussions in transatlantic calls using SIP phones—again, calls of crystal clear clarity to our surprise. The authors are grateful to Richard Shockey from NeuStar, Inc. and Douglas Ranalli from NetNumber, Inc. for numerous discussions regarding ENUM. Theodore Havinis has contributed to the SIP-QoS-AAA aspect for mobile users.

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xxiv Acknowledgments

We acknowledge countless helpful discussions and insight from many participants in the IETF and especially to Scott Bradner for holding the authors and 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 leading exhibition of products for what was initially an obscure and unknown protocol 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 & Sons have been instrumental in editing this book.

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Introduction

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

The Internet Has Replaced the Telephone System and the Telecommunication Networks Since the publication in 2001 of the first edition of this book, Internet Communications Using SIP, Voice over IP (VoIP) has developed from an emerging technology to the recognized replacement of existing global telephone systems based on Time Division Multiplex (TDM) circuit switching. The Internet has also replaced the proposed connection-oriented offsprings of TDM, such as the Integrated Services Digital Network (ISDN) and the Asynchronous Transfer Multiplex (ATM) based broadband version BISDN, envisaged for the telecommunications industry by the International Telecommunications Union ITU-T standards body. TDM, ATM, ISDN, and BISDN are now history. All wired and wireless communications are instead migrating to the Internet standards developed by the Internet Engineering Task Force (IETF). The legacy telecommunication networks, while still dominant, are recognized as a presentday cash cow only and are scheduled for replacement by IP networks. The end-to-end nature of the Internet that places intelligence in the applications running in the endpoints and gives control to the user at the endpoints has 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, area director of the IETF: “The Internet is the service.”

The Session Initiation Protocol Is the Standard for VoIP and Multimedia Communications Another change from the first edition of this book is the Session Initiation Protocol (SIP), which has been adopted by practically all public VoIP service providers for wired and wireless communications. The discussions about SIP versus H.323 standardized by the ITU-T are over as well. The installed base of H.323 is considered a liability and planned for replacement by SIP sooner or later. A global industry has emerged to take advantage of SIP and its associated IETF standards for real-time communications. More than 560 VoIP service providers have been reported [1] in early 2006, most of them using SIP-based networks. The list of SIP-based equipment (such as SIP phones, software for PCs, and mobile devices, servers, gateways, and so on) is now large and still growing. Actually, all equipment and system vendors are now supporting SIP.

Presence and Instant Messaging Are Mainstream Communications Presence and instant messaging (IM) are now mainstream with consumers and, in the enterprise, complementing or sometimes replacing voice communications in specific situations (such as in circumstances where silence is required). Even for VoIP, presence has emerged not only as a valuable enhancement, but presence may be the dial tone of the twenty-first century. Presence and event-based communications have enabled the integration of communications with applications. Presence and IM are discussed in Chapter 13, “Presence and Instant Messaging.” The so-called IM services provided by large Internet companies, such as AOL, Apple, Google, IBM, Microsoft, Skype (not SIP-based), and Yahoo!, actually carry at present most of the public VoIP traffic between end users around the globe. It is not far-fetched to see the IM Internet companies replacing the former telephone companies in the voice communication business. Many legacy telecommunication companies are also using VoIP to replace the internal TDM voice networks, but their VoIP services may not survive the advanced technologies deployed by the IM Internet companies and the challenge posed by peer-to-peer (P2P) communications.

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Redefining Communications: Mobility, Emergency and Equal Access for the Disabled Internet communications have been known not to be dependent on the location on the Internet. Application-level mobility based on SIP is a key component to seamless mobile communications, as discussed in Chapter 15, “SIP Application Level Mobility.” Emergency calling services by users in distress using the Internet (such as 911 in the United States or 112 in Europe) are far more powerful and cost less than the Public Switched Telephone Network (PSTN) based emergency services. Internet-based emergency calling is indeed in the design stage in a number of countries. Chapter 16, “Emergency and Preemption Communication Services,” discusses Internet-based emergency services. The multimedia nature of Internet communications gives hearing- and speech-impaired people the opportunity to fully participate in rich communications for work and in personal life. Chapter 17, “Accessibility for the Disabled,” discusses access to communications for disabled people.

The Rise of Peer-to-Peer Communications P2P traffic has risen in the Internet since around 2000 and became the dominant part of Internet traffic by 2004. Since 2004, Skype (which is based on P2P VoIP, IM, and presence) has also become by far the dominant VoIP provider worldwide. 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 applications and their nature are discussed in Chapter 20, “Peer-to-Peer SIP,” and also in Chapter 6, “SIP Overview.” Though the present VoIP industry is built on client-server (CS) SIP, this may significantly change. To quote David Bryan from p2p.org: “P2P SIP may change VoIP to the same extent that VoIP has changed telecommunications.”

VoIP and Multimedia Communications Services Are Still Fragmented In spite of all the technological progress, VoIP, IM, presence, and multimedia services are still a highly fragmented industry: ■■

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

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access to the legacy telephone system, actually a voice gateway service between the Internet and TDM. The business model of most VoIP service providers is just lower cost for legacy-style telephone service, also called PSTN over IP. The PSTN gateway services are using IP inside their networks, but users are not exposed to the rich IP services, except when 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 actively pursued by most mobile service providers by deploying systems where their users can get rich IP multimedia services only on their own networks. The fees to communicate between mobile service providers are a significant part of the business model, and open connectivity to the Internet (“Internet neutrality”) is still a hotly debated issue. Internet neutrality is also still debated by many broadband Internet access providers (such as DSL and cable companies), although we believe that enlightened government regulators in the developed countries will weigh in favor of users and open network access in general.

The proliferation of islands for communications makes them less useful the more there are, since this proliferation is in denial of Metcalf’s law that the value of a network increases with the square of the number of points attached to the network. The Internet with more than 1 billion attached endpoints has thus the highest value for communications. By contrast, the mobile phone industry 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 the legacy telephone network have long been used by telephone industry marketers to scare users away from VoIP. In the meantime, all public VoIP services have proven that Internet best-effort QoS works just fine, as long network congestion is avoided. Internet-based voice can actually be much better than the 3.1 kHz voice over the PSTN. As for reliability, all recent major man-made and natural disasters have proven the Internet and VoIP to be more resilient than the existing wireline and wireless telephone networks. Chapter 18, “Quality of Service for Real-Time Internet Communications,” is aimed at a balanced approach for QoS, and Chapter 16, “Emergency and Preemption Communication Services,” discusses the Emergency Services based on SIP.