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Abbreviations | p. ix |
Glossary | p. xxi |
Preface | p. xxix |
Multimedia Over Packet | p. 1 |
Transporting voice, fax, and video over a packet network | p. 1 |
A Darwinian view of voice transport | p. 1 |
Voice and video over IP with RTP and RTCP | p. 5 |
Encoding media streams | p. 16 |
Codecs | p. 16 |
DTMF | p. 39 |
Fax | p. 40 |
H.323: Packet-based Multimedia Communications Systems | p. 49 |
Introduction | p. 49 |
Understanding H.323 | p. 50 |
Development of the standard | p. 52 |
Relation between H.323 and H.245 versions, H.323 annexes, and related specifications | p. 55 |
Where to find the documentation | p. 57 |
H.323 step by step | p. 58 |
The 'hello world case': simple voice call from terminal A to terminal B | p. 58 |
A more complex case: calling a public phone from the Internet using a gatekeeper | p. 72 |
The gatekeeper-routed model | p. 79 |
H.323 calls across multiple zones or administrative domains | p. 86 |
Optimizing and enhancing H.323 | p. 95 |
Issues in H.323v1 | p. 95 |
The 'early H.245' procedure | p. 99 |
The 'fast-connect' procedure | p. 99 |
H.245 tunneling | p. 103 |
Reverting to normal operation | p. 106 |
Using RAS properly and only when required | p. 106 |
Conferencing with H.323 | p. 108 |
The MCU conference bridge, MC and MP subsystems | p. 108 |
Creating or joining a conference | p. 109 |
H.332 | p. 113 |
Directories and numbering | p. 114 |
Introduction | p. 114 |
Contacting an email alias with H.323 and the DNS | p. 115 |
E164 numbers and IP telephony | p. 116 |
H.323 security | p. 124 |
Typical deployment cases | p. 124 |
H.235 | p. 131 |
Supplementary services | p. 148 |
Supplementary services using H.450 | p. 148 |
Proper use of H.450 supplementary services, future directions for implementation of supplementary services | p. 154 |
Future work on H.323 | p. 155 |
The Session Initiation Protocol (SIP) | p. 159 |
The origin and purpose of SIP | p. 159 |
From RFC 2543 to RFC 3261 | p. 163 |
From RFC 3261 to 3GPP, 3GPP2 and TISPAN | p. 166 |
Overview of a simple SIP call | p. 167 |
Basic call scenario | p. 167 |
Syntax of SIP messages | p. 169 |
Call handling services with SIP | p. 219 |
Location and registration | p. 220 |
The proxy function, back to back user agents | p. 230 |
Some common services | p. 242 |
Multiparty conferencing | p. 244 |
SIP security | p. 250 |
Media security | p. 250 |
Message exchange security | p. 251 |
Instant messaging (IM) and presence | p. 254 |
Common profile for instant messaging (CPIM) | p. 255 |
RFC 3265, Specific Event Notification | p. 260 |
RFC 3428: SIP extensions for instant messaging | p. 266 |
The 3GPP IP Multimedia Subsystem (IMS) Architecture | p. 269 |
Introduction | p. 269 |
Centralized value added services platforms on Switched telephone networks: the 'tromboning' issue | p. 269 |
The 'Intelligent Network' (IN) | p. 270 |
How VoIP solves the 'tromboning' issue. The value added services architecture of 3GPP IMS | p. 270 |
The IMS architecture is ideal for mobile Networks … but not only | p. 273 |
Overview of the IMS architecture | p. 274 |
Registration | p. 274 |
SIP session establishment in an IMS environment | p. 275 |
A few remarks on the IMS architecture | p. 278 |
The IMS CSCFs | p. 279 |
The Proxy-CSCF | p. 279 |
The Serving-CSCF (S-CSCF) and Application Servers (AS) | p. 282 |
The Media Resource Function (MRF) | p. 286 |
The full picture: 3GPP release 8, TISPAN | p. 288 |
The packet core domain: the evolved packet system | p. 289 |
The IMS domain | p. 299 |
Summary of SIP extensions required in an IMS network | p. 311 |
The Media Gateway to Media Controller Protocol (MGCP) | p. 313 |
Introduction: why MGCP? | p. 313 |
Stimulus protocols | p. 313 |
Decomposed gateways | p. 315 |
Some history | p. 317 |
MGCP 1.0 | p. 318 |
The MGCP connection model | p. 321 |
The protocol | p. 323 |
Handling of fax | p. 350 |
Extensions for phone user interface control | p. 354 |
Sample MGCP call flows | p. 358 |
Call set-up | p. 358 |
DTMF tones | p. 364 |
Call release | p. 364 |
The future of MGCP | p. 365 |
Advanced Topics: Call Redirection | p. 367 |
Call redirection in VoIP networks | p. 367 |
Call transfer, call forward, call deflection | p. 367 |
Summary of major issues | p. 368 |
Reference network configurations in the PSTN | p. 371 |
Reference network configurations with VoIP | p. 374 |
How to signal call transfer? | p. 387 |
VoIP call redirection and call routing | p. 388 |
Conclusion | p. 390 |
Advanced Topics: NAT Traversal | p. 393 |
Introduction to Network Address Translation | p. 393 |
One-to-one NAT | p. 393 |
NAPT | p. 394 |
Issues with NAT and NAPT | p. 396 |
Workarounds for VoIP when the network cannot be controlled | p. 398 |
Ringing the proper phone | p. 398 |
Using port forwarding to solve the wrong media address problem | p. 399 |
STUN | p. 399 |
Other proposals: COMEDIA and TURN | p. 402 |
Recommended network design for service providers | p. 404 |
Avoid NAT in the customer premises for VoIP | p. 405 |
Media proxies | p. 412 |
Security considerations | p. 415 |
Conclusion | p. 416 |
Annex | p. 417 |
Index | p. 427 |
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