ir network locations. SIP support five facets of establishing and terminating multimedia communi- cations: user location (the determination of the end system to be used for commu- nication), user availability (the determination of the willingness of the called party to engage in communications), user capabilities (the determination of the media and media parameters to be used), session setup (the establishment of session parameters at both the called and calling parties), and session management (the transfer and termination of sessions, modification of session parameters, and invo- cation of services). SIP can be used with other IETF protocols to build a complete multimedia architecture. For example, it can be used with RTP for transporting real-time data and providing QoS feedback, with Real-Time Streaming Protocol (RTSP) for con- trolling delivery of streaming media, with MGCP for controlling gateways to the PSTN, or with Session Description Protocol (SDP) for describing multimedia ses- sions. Although SIP should be used in conjunction with other protocols to provide complete services to users, the basic functionality and operation of SIP do not depend on any other protocols. SIP, which works with both IPv4 and IPv6, operates as follows for Internet tele- phony sessions: Callers and callees are identified by SIP addresses. When making a SIP call, a caller locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their locations with SIP servers; SIP addresses can be embedded in Web pages and, therefore, can be integrated as part of powerful applications such as click-to-talk. The purpose of SIP is just to make the communication possible (see Figure 9.6). The communication itself must be achieved by another means and possibly another protocol. As mentioned earlier, two protocols that are most often used alon