IP Telephony -- Contents -- Abbreviations -- Glossary -- Preface -- 1 Multimedia Over Packet -- 1.1 Transporting voice, fax, and video over a packet network -- 1.1.1 A Darwinian view of voice transport -- 1.1.2 Voice and video over IP with RTP and RTCP -- 1.2 Encoding media streams -- 1.2.1 Codecs -- 1.2.2 DTMF -- 1.2.3 Fax -- 2 H.323: Packet-based Multimedia Communications Systems -- 2.1 Introduction -- 2.1.1 Understanding H.323 -- 2.1.2 Development of the standard -- 2.1.3 Relation between H.323 and H.245 versions, H.323 annexes, and related specifications -- 2.1.4 Where to find the documentation -- 2.2 H.323 step by step -- 2.2.1 The 'hello world case': simple voice call from terminal A to terminal B -- 2.2.2 A more complex case: calling a public phone from the Internet using a gatekeeper -- 2.2.3 The gatekeeper-routed model -- 2.2.4 H.323 calls across multiple zones or administrative domains -- 2.3 Optimizing and enhancing H.323 -- 2.3.1 Issues in H.323v1 -- 2.3.2 The 'early H.245' procedure -- 2.3.3 The 'fast-connect' procedure -- 2.3.4 H.245 tunneling -- 2.3.5 Reverting to normal operation -- 2.3.6 Using RAS properly and only when required -- 2.4 Conferencing with H.323 -- 2.4.1 The MCU conference bridge, MC and MP subsystems -- 2.4.2 Creating or joining a conference -- 2.4.3 H.332 -- 2.5 Directories and numbering -- 2.5.1 Introduction -- 2.5.2 Contacting an email alias with H.323 and the DNS -- 2.5.3 E164 numbers and IP telephony -- 2.6 H.323 security -- 2.6.1 Typical deployment cases -- 2.6.2 H.235 -- 2.7 Supplementary services -- 2.7.1 Supplementary services using H.450 -- 2.7.2 Proper use of H.450 supplementary services, future directions for implementation of supplementary services -- 2.8 Future work on H.323 -- 3 The Session Initiation Protocol (SIP) -- 3.1 The origin and purpose of SIP -- 3.1.1 From RFC 2543 to RFC 3261.

3.1.2 From RFC 3261 to 3GPP, 3GPP2 and TISPAN -- 3.2 Overview of a simple SIP call -- 3.2.1 Basic call scenario -- 3.2.2 Syntax of SIP messages -- 3.3 Call handling services with SIP -- 3.3.1 Location and registration -- 3.3.2 The proxy function, back to back user agents -- 3.3.3 Some common services -- 3.3.4 Multiparty conferencing -- 3.4 SIP security -- 3.4.1 Media security -- 3.4.2 Message exchange security -- 3.5 Instant messaging (IM) and presence -- 3.5.1 Common profile for instant messaging (CPIM) -- 3.5.2 RFC 3265, Specific Event Notification -- 3.5.3 RFC 3428: SIP extensions for instant messaging -- 4 The 3GPP IP Multimedia Subsystem (IMS) Architecture -- 4.1 Introduction -- 4.1.1 Centralized value added services platforms on switched telephone networks: the 'tromboning' issue -- 4.1.2 The 'Intelligent Network' (IN) -- 4.1.3 How VoIP solves the 'tromboning' issue. The value added services architecture of 3GPP IMS -- 4.1.4 The IMS architecture is ideal for mobile networks . . . but not only -- 4.2 Overview of the IMS architecture -- 4.2.1 Registration -- 4.2.2 SIP session establishment in an IMS environment -- 4.2.3 A few remarks on the IMS architecture -- 4.3 The IMS CSCFs -- 4.3.1 The Proxy-CSCF -- 4.3.2 The Serving-CSCF (S-CSCF) and Application Servers (AS) -- 4.3.3 The Media Resource Function (MRF) -- 4.4 The full picture: 3GPP release 8, TISPAN -- 4.4.1 The packet core domain: the evolved packet system -- 4.4.2 The IMS domain -- 4.4.3 Summary of SIP extensions required in an IMS network -- 5 The Media Gateway to Media Controller Protocol (MGCP) -- 5.1 Introduction: why MGCP? -- 5.1.1 Stimulus protocols -- 5.1.2 Decomposed gateways -- 5.1.3 Some history -- 5.2 MGCP 1.0 -- 5.2.1 The MGCP connection model -- 5.2.2 The protocol -- 5.2.3 Handling of fax -- 5.2.4 Extensions for phone user interface control -- 5.3 Sample MGCP call flows.

5.3.1 Call set-up -- 5.3.2 DTMF tones -- 5.3.3 Call release -- 5.4 The future of MGCP -- 6 Advanced Topics: Call Redirection -- 6.1 Call redirection in VoIP networks -- 6.1.1 Call transfer, call forward, call deflection -- 6.1.2 Summary of major issues -- 6.1.3 Reference network configurations in the PSTN -- 6.1.4 Reference network configurations with VoIP -- 6.1.5 How to signal call transfer? -- 6.1.6 VoIP call redirection and call routing -- 6.1.7 Conclusion -- 7 Advanced Topics: NAT Traversal -- 7.1 Introduction to Network Address Translation -- 7.1.1 One-to-one NAT -- 7.1.2 NAPT -- 7.1.3 Issues with NAT and NAPT -- 7.2 Workarounds for VoIP when the network cannot be controlled -- 7.2.1 Ringing the proper phone -- 7.2.2 Using port forwarding to solve the wrong media address problem -- 7.2.3 STUN -- 7.2.4 Other proposals: COMEDIA and TURN -- 7.3 Recommended network design for service providers -- 7.3.1 Avoid NAT in the customer premises for VoIP -- 7.3.2 Media proxies -- 7.3.3 Security considerations -- 7.4 Conclusion -- Annex -- Index.