sion initiation protocol (SIP) is a key standard in the IP telephony architecture, backed up by for example IETF and 3GPP. See Figure 14.6. H.323 is an alternative. In this book SIP gets most focus. Both SIP and H.323 are signalling and control standards for Internet telephony. H.323 is an ITU recommendation, and SIP is developed by IETF. Both protocols provide a similar set of services, but SIP is often considered simpler with less logical components. H.323 is an ITU standard. In H.323 architecture, each client belongs to a zone, and there is a gatekeeper in each zone. All the clients of a zone are registered to its gatekeeper. The gatekeeper provides address translation (allowing aliases to be used), admission control and bandwidth control. 14.4.1 SESSION INITIATION PROTOCOL (SIP) The IETF Multiparty Multimedia Session Control (MMUSIC) working group has specified an IP telephony architecture based on SIP. The two basic protocols are the session Understanding Changing Telecommunications - Building a Successful Telecom Business description protocol (SDP), which is specified in RFC 2327, for channel and terminal capability handling, and the session initiation protocol (SIP) for registration, admission, status and Q.931 messages. SIP uses a client server model similar to the HTTP. It is used in conjunction with other protocols such as SDP, RTP and RSVP. SIP can set up all types of sessions, e.g. voice, video, text or combinations, and manage any of these types of sessions, regardless of the media type. SIP is a direct competitor to H.323. An IP telephone call is considered a kind multimedia session in which voice is exchanged between the parties. The voice path is independent of whether SIP or H.323 is used for signalling. SIP is independent of the transport layer and only requires an unreliable datagram service, as it provides its own reliability mechanism. It is typically transported over UDP or TCP, but could also use frame relay, ATM AAL5 and X.25. SIP supports personal mobility and negotiation of the capabilit