All you need to know about deploying VoIP protocols in one comprehensive and highly practical reference - Now updated with coverage on SIP and the IMS infrastructure This book provides a comprehensive and practical overview of the technology behind Internet Telephony (IP), providing essential information to Network Engineers, Designers, and Managers who need to understand the protocols. Furthermore, the author explores the issues involved in the migration of existing telephony infrastructure to an IP - based real time communication service. Assuming a working knowledge of IP and networking, it addresses the technical aspects of real-time applications over IP. Drawing on his extensive research and practical development experience in VoIP from its earliest stages, the author provides an accessible reference to all the relevant standards and cutting-edge techniques in a single resource.
Key Features:
- Updated with a chapter on SIP and the IMS infrastructure
- Covers ALL the major VoIP protocols – SIP, H323 and MGCP
- Includes a large section on practical deployment issues gleaned from the authors’ own experience
- Chapter on the rationale for IP telephony and description of the technical and business drivers for transitioning to all IP networks
This book will be a valuable guide for professional network engineers, designers and managers, decision makers and project managers overseeing VoIP implementations, market analysts, and consultants. Advanced undergraduate and graduate students undertaking data/voice/multimedia communications courses will also find this book of interest.
Olivier Hersent founded NetCentrex, a leading provider of VoIP infrastructure for service providers, then became CTO of Comverse after the acquisition of NetCentrex. He now manages Actility, provider of IMS based M2M and smartgrid infrastructure and applications.
Author(s): Olivier Hersent
Edition: 2
Publisher: Wiley
Year: 2010
Language: English
Pages: 475
IP Telephony......Page 3
Contents......Page 7
Abbreviations......Page 11
Glossary......Page 23
Preface......Page 31
1.1.1 A Darwinian view of voice transport......Page 37
1.1.2 Voice and video over IP with RTP and RTCP......Page 41
1.2.1 Codecs......Page 52
1.2.2 DTMF......Page 75
1.2.3 Fax......Page 76
2.1 Introduction......Page 85
2.1.1 Understanding H.323......Page 86
2.1.2 Development of the standard......Page 88
2.1.3 Relation between H.323 and H.245 versions, H.323 annexes, and related specifications......Page 91
2.1.4 Where to find the documentation......Page 93
2.2.1 The ‘hello world case’: simple voice call from terminal A to terminal B......Page 94
2.2.2 A more complex case: calling a public phone from the Internet using a gatekeeper......Page 108
2.2.3 The gatekeeper-routed model......Page 115
2.2.4 H.323 calls across multiple zones or administrative domains......Page 122
2.3.1 Issues in H.323v1......Page 131
2.3.3 The ‘fast-connect’ procedure......Page 135
2.3.4 H.245 tunneling......Page 139
2.3.6 Using RAS properly and only when required......Page 142
2.4.1 The MCU conference bridge, MC and MP subsystems......Page 144
2.4.2 Creating or joining a conference......Page 145
2.4.3 H.332......Page 149
2.5.1 Introduction......Page 150
2.5.2 Contacting an email alias with H.323 and the DNS......Page 151
2.5.3 E164 numbers and IP telephony......Page 152
2.6.1 Typical deployment cases......Page 160
2.6.2 H.235......Page 167
2.7.1 Supplementary services using H.450......Page 184
2.7.2 Proper use of H.450 supplementary services, future directions for implementation of supplementary services......Page 190
2.8 Future work on H.323......Page 191
3.1 The origin and purpose of SIP......Page 195
3.1.1 From RFC 2543 to RFC 3261......Page 199
3.1.2 From RFC 3261 to 3GPP, 3GPP2 and TISPAN......Page 202
3.2.1 Basic call scenario......Page 203
3.2.2 Syntax of SIP messages......Page 205
3.3 Call handling services with SIP......Page 255
3.3.1 Location and registration......Page 256
3.3.2 The proxy function, back to back user agents......Page 266
3.3.3 Some common services......Page 278
3.3.4 Multiparty conferencing......Page 280
3.4.1 Media security......Page 286
3.4.2 Message exchange security......Page 287
3.5 Instant messaging (IM) and presence......Page 290
3.5.1 Common profile for instant messaging (CPIM)......Page 291
3.5.2 RFC 3265, Specific Event Notification......Page 296
3.5.3 RFC 3428: SIP extensions for instant messaging......Page 302
4.1.1 Centralized value added services platforms on switched telephone networks: the ‘tromboning’ issue......Page 305
4.1.3 How VoIP solves the ‘tromboning’ issue. The value added services architecture of 3GPP IMS......Page 306
4.1.4 The IMS architecture is ideal for mobile networks . . . but not only......Page 309
4.2.1 Registration......Page 310
4.2.2 SIP session establishment in an IMS environment......Page 311
4.2.3 A few remarks on the IMS architecture......Page 314
4.3.1 The Proxy-CSCF......Page 315
4.3.2 The Serving-CSCF (S-CSCF) and Application Servers (AS)......Page 318
4.3.3 The Media Resource Function (MRF)......Page 322
4.4 The full picture: 3GPP release 8, TISPAN......Page 324
4.4.1 The packet core domain: the evolved packet system......Page 325
4.4.2 The IMS domain......Page 335
4.4.3 Summary of SIP extensions required in an IMS network......Page 347
5.1.1 Stimulus protocols......Page 349
5.1.2 Decomposed gateways......Page 351
5.1.3 Some history......Page 353
5.2 MGCP 1.0......Page 354
5.2.1 The MGCP connection model......Page 357
5.2.2 The protocol......Page 359
5.2.3 Handling of fax......Page 386
5.2.4 Extensions for phone user interface control......Page 390
5.3.1 Call set-up......Page 394
5.3.3 Call release......Page 400
5.4 The future of MGCP......Page 401
6.1.1 Call transfer, call forward, call deflection......Page 403
6.1.2 Summary of major issues......Page 404
6.1.3 Reference network configurations in the PSTN......Page 407
6.1.4 Reference network configurations with VoIP......Page 410
6.1.5 How to signal call transfer?......Page 423
6.1.6 VoIP call redirection and call routing......Page 424
6.1.7 Conclusion......Page 426
7.1.1 One-to-one NAT......Page 429
7.1.2 NAPT......Page 430
7.1.3 Issues with NAT and NAPT......Page 432
7.2.1 Ringing the proper phone......Page 434
7.2.3 STUN......Page 435
7.2.4 Other proposals: COMEDIA and TURN......Page 438
7.3 Recommended network design for service providers......Page 440
7.3.1 Avoid NAT in the customer premises for VoIP......Page 441
7.3.2 Media proxies......Page 448
7.3.3 Security considerations......Page 451
7.4 Conclusion......Page 452
Annex......Page 453
Index......Page 463
