Voice on IP: Internet, fixed and mobile

The second part of this dossier describes the main standards involved in supporting Voice over IP. Firstly, call control protocols and associated signaling are presented with H.323 and SIP V1, then MGCP and Megaco/H.248 introduced with the "next generation networks" (NGN) and finally SIP V2 on which the IMS domain is based. Secondly, source coding and the main codecs associated with VoIP are introduced, whether standardized by the International Telecommunication Union (G.729), 3GPP (AMR) or IETF (iLBC). Thirdly, the protocols associated with the transmission of digitized voice frames (RTP/RTCP, UDP/UDP-Lite, IP) are described. Fourthly, mechanisms for header compression (ROHC), signaling (SigComp) and user data are introduced. Fifthly, call control (HTTP Digest, IPsec, TLS) and communication security (SRTP), client authentication (RADIUS and DIAMETER) and firewall and NAT traversal are explained. Finally, protocols (RSVP, COPS, IP ToS, IP precedence, IntServ and DiffServ) and mechanisms (VAD, UEP/UED and MOS) for quality of service applied to VoIP calls are covered.