IP telephony : packet-based multimedia communications systems Upplaga 1

Foreword Jeff Pulver Preface SECTION 1 The application layer, IP telephony protocols H.323 and a general background on IP telephony A little history Where to find documentation From RTP to H.323: a quick tour Transporting voice over a packet network A Darwinian view of voice transport Voice and video over IP with RTP and RTCP H.323 step by step The “hello world” case: simple voice call from terminal A to terminal B A more complex case: calling a public phone from the Internet H.323 across multiple domains Advanced topics Faster procedures Conferencing with H.323 Directories and numbering H.323 security: H.235 Media streams Codecs DTMF Fax Supplementary services using H.450 H.450.1 H.450.2: call transfer H.450.3: call diversion The future of H.450 Future work on H.323 The session initiation protocol (SIP) The origin and purpose of SIP Overview of a simple SIP call SIP messages Session description syntax, SDP Advanced services with SIP SIP entities User location and mobility Multiparty conferencing Configuring network-based call handling Billing SIP calls SIP security Media security SIP firewalls SIP and H.323 What SIP does and H.323 does not What H.323 does and SIP does not H.323 to SIP gateways Conclusion on the future of SIP and its relation to H.323 Media gateway to media controller protocols (MGCP) Introduction Which protocol? What about the requirements? A new architecture for IP telephony? What is MGCP? MGCP commands Protocols at work Scenario 1 Scenario 2 The H.323 case SECTION 2 Voice technology background Voice quality Introduction Reference connection Echo in a telephone network Talker echo, listener echo Hybrid echo Acoustic echo How to limit echo Delay in a telephone network Influence of the operating system Influence of the jitter buffer policy on delay Influence of the codec, frame grouping and redundancy Consequence for measuring end-to-end delay Acceptability of a phone call with echo and delay Consequences for an IP telephony network The approach of ETSI TIPHON Other requirements for gateways and transcoding equipment Voice coding Introduction Transmitted bandwidth, sampling and quantization Some basic tools for digital signal processing The A or u law ITU-T 64 kbit/s G.711 Specifications and subjective quality of speech coders ACR subjective test or what is the MOS Speech and auditory properties Speech production Auditory perception used for speech and audio bitrate reduction Quantization and coders Adaptive quantizers Differential (and predictive) quantization Vector quantization Entropy coding Waveform coders: the ADPCM ITU-T G.726 Wideband speech coding using waveform-type coder Speech coding techniques Hybrids and analysis by synthesis speech coders The GSM Full Rate RPE-LTP speech coder Codebook excited linear predictive (CELP) coders The ITU-T 8 kbit/s CS-ACELP G.729 The ITU-T G.723.1 Discontinuous transmission and comfort noise generation The low delay CELP coding scheme: ITU-T G.728 Partial conclusion on speech coding techniques and their near future Remarks applicable to VolP telephone gateways The electrical echo canceller Best effort Non-standardization SECTION 3 The network Quality of service What is QoS? Describing a stream Queuing techniques for QoS Signaling QoS requirements RSVP Scaling issues with RSVP Classes of service in the backbone RSVP to Diffserv mapping Improving QoS in the best-effort class Issues with slow links Conclusion Network dimensioning Simple compressed voice flow model Voice coders Model for N simultaneous conversations using the same coder Loss rate and dimensioning Packet or frame loss? Multiple coders Network dedicated to IP telephony Is it necessary? Network dimensioning Merging data communications and voice communications on a common IP backbone Prioritization of voice flows Impact on end-to-end delay Multipoint communications Audio multipoint conferences Multipoint video conferencing Modeling call seizures Introduction to the Erlang model Model for a limited set of servers and calls are rejected if no server is available Calls per second Conclusion Multicast routing Introduction When to use multicast routing A real-time technology Network efficiency The multicast framework Multicast address, multicast group Multicast on Ethernet Group membership protocol Controlling scope in multicast applications Scope versus initial TTL TTL threshold Administrative scoping Building the multicast delivery tree Flooding and spanning tree Shared trees Source-based trees Multicast routing protocols DVMRPv3 Other protocols Core-based trees Security issues in IP multicast Unauthorized listening Unauthorized sending and denial of service attacks Firewalls The MBONE Routing protocols Inter-domain multicast routing Interoperational between domains running different protocols BGMP Conclusion on multicast inter-domain routing Multicast caveats Multicasting on non-broadcast media Flooding Common issues Address allocation MBONE applications Video conferencing with RTP on multicast networks SDR: session directory VIC and VAT Reliable multicast Appendix: well known multicast addresses References Glossary Index