IP Telephony: Deploying VoIP Protocols and IMS Infrastructure / Edition 2

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Overview

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 two new chapters; SIP and IMS Infrastructure

Covers ALL the major VoIP protocols - SIP, H323 and MGCP

Includes a large section on practical deployment issues gleaned from the author's 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.

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Product Details

  • ISBN-13: 9780470665848
  • Publisher: Wiley
  • Publication date: 12/28/2010
  • Edition description: New Edition
  • Edition number: 2
  • Pages: 474
  • Product dimensions: 6.80 (w) x 9.90 (h) x 1.10 (d)

Table of Contents

Abbreviations ix

Glossary xxi

Preface xxix

1 Multimedia Over Packet 1

1.1 Transporting voice, fax, and video over a packet network 1

1.1.1 A Darwinian view of voice transport 1

1.1.2 Voice and video over IP with RTP and RTCP 5

1.2 Encoding media streams 16

1.2.1 Codecs 16

1.2.2 DTMF 39

1.2.3 Fax 40

2 H.323: Packet-based Multimedia Communications Systems 49

2.1 Introduction 49

2.1.1 Understanding H.323 50

2.1.2 Development of the standard 52

2.1.3 Relation between H.323 and H.245 versions, H.323 annexes, and related specifications 55

2.1.4 Where to find the documentation 57

2.2 H.323 step by step 58

2.2.1 The 'hello world case': simple voice call from terminal A to terminal B 58

2.2.2 A more complex case: calling a public phone from the Internet using a gatekeeper 72

2.2.3 The gatekeeper-routed model 79

2.2.4 H.323 calls across multiple zones or administrative domains 86

2.3 Optimizing and enhancing H.323 95

2.3.1 Issues in H.323v1 95

2.3.2 The 'early H.245' procedure 99

2.3.3 The 'fast-connect' procedure 99

2.3.4 H.245 tunneling 103

2.3.5 Reverting to normal operation 106

2.3.6 Using RAS properly and only when required 106

2.4 Conferencing with H.323 108

2.4.1 The MCU conference bridge, MC and MP subsystems 108

2.4.2 Creating or joining a conference 109

2.4.3 H.332 113

2.5 Directories and numbering 114

2.5.1 Introduction 114

2.5.2 Contacting an email alias with H.323 and the DNS 115

2.5.3 E164 numbers and IP telephony 116

2.6 H.323 security 124

2.6.1 Typical deployment cases 124

2.6.2 H.235 131

2.7 Supplementary services 148

2.7.1 Supplementary services using H.450 148

2.7.2 Proper use of H.450 supplementary services, future directions for implementation of supplementary services 154

2.8 Future work on H.323 155

3 The Session Initiation Protocol (SIP) 159

3.1 The origin and purpose of SIP 159

3.1.1 From RFC 2543 to RFC 3261 163

3.1.2 From RFC 3261 to 3GPP, 3GPP2 and TISPAN 166

3.2 Overview of a simple SIP call 167

3.2.1 Basic call scenario 167

3.2.2 Syntax of SIP messages 169

3.3 Call handling services with SIP 219

3.3.1 Location and registration 220

3.3.2 The proxy function, back to back user agents 230

3.3.3 Some common services 242

3.3.4 Multiparty conferencing 244

3.4 SIP security 250

3.4.1 Media security 250

3.4.2 Message exchange security 251

3.5 Instant messaging (IM) and presence 254

3.5.1 Common profile for instant messaging (CPIM) 255

3.5.2 RFC 3265, Specific Event Notification 260

3.5.3 RFC 3428: SIP extensions for instant messaging 266

4 The 3GPP IP Multimedia Subsystem (IMS) Architecture 269

4.1 Introduction 269

4.1.1 Centralized value added services platforms on Switched telephone networks: the 'tromboning' issue 269

4.1.2 The 'Intelligent Network' (IN) 270

4.1.3 How VoIP solves the 'tromboning' issue. The value added services architecture of 3GPP IMS 270

4.1.4 The IMS architecture is ideal for mobile Networks … but not only 273

4.2 Overview of the IMS architecture 274

4.2.1 Registration 274

4.2.2 SIP session establishment in an IMS environment 275

4.2.3 A few remarks on the IMS architecture 278

4.3 The IMS CSCFs 279

4.3.1 The Proxy-CSCF 279

4.3.2 The Serving-CSCF (S-CSCF) and Application Servers (AS) 282

4.3.3 The Media Resource Function (MRF) 286

4.4 The full picture: 3GPP release 8, TISPAN 288

4.4.1 The packet core domain: the evolved packet system 289

4.4.2 The IMS domain 299

4.4.3 Summary of SIP extensions required in an IMS network 311

5 The Media Gateway to Media Controller Protocol (MGCP) 313

5.1 Introduction: why MGCP" 313

5.1.1 Stimulus protocols 313

5.1.2 Decomposed gateways 315

5.1.3 Some history 317

5.2 MGCP 1.0 318

5.2.1 The MGCP connection model 321

5.2.2 The protocol 323

5.2.3 Handling of fax 350

5.2.4 Extensions for phone user interface control 354

5.3 Sample MGCP call flows 358

5.3.1 Call set-up 358

5.3.2 DTMF tones 364

5.3.3 Call release 364

5.4 The future of MGCP 365

6 Advanced Topics: Call Redirection 367

6.1 Call redirection in VoIP networks 367

6.1.1 Call transfer, call forward, call deflection 367

6.1.2 Summary of major issues 368

6.1.3 Reference network configurations in the PSTN 371

6.1.4 Reference network configurations with VoIP 374

6.1.5 How to signal call transfer" 387

6.1.6 VoIP call redirection and call routing 388

6.1.7 Conclusion 390

7 Advanced Topics: NAT Traversal 393

7.1 Introduction to Network Address Translation 393

7.1.1 One-to-one NAT 393

7.1.2 NAPT 394

7.1.3 Issues with NAT and NAPT 396

7.2 Workarounds for VoIP when the network cannot be controlled 398

7.2.1 Ringing the proper phone 398

7.2.2 Using port forwarding to solve the wrong media address problem 399

7.2.3 STUN 399

7.2.4 Other proposals: COMEDIA and TURN 402

7.3 Recommended network design for service providers 404

7.3.1 Avoid NAT in the customer premises for VoIP 405

7.3.2 Media proxies 412

7.3.3 Security considerations 415

7.4 Conclusion 416

Annex 417

Index 427

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