摘要：The Session Initiation Protocol (SIP), defined in RFC 3261 [6], is an application level signaling protocol for setting up, modifying, and terminating real-time sessions between participants over an IP data network. SIP can support any type of single-media or multi-media session, including teleconferencing.SIP is just one component in the set of protocols and services needed to support multimedia exchanges over the Internet. SIP is the signaling protocol that enables one party to place a call to another party and to negotiate the parameters of a multimedia session. The actual audio, video, or other multimedia content is exchanged between session participants using an appropriate transport protocol. In many cases, the transport protocol to use is the Real-Time Transport Protocol (RTP). Directory access and lookup protocols are also needed.The key driving force behind SIP is to enable Internet telephony, also referred to as Voice over IP (VoIP). There is wide industry acceptance that SIP will be the standard IP signaling mechanism for voice and multimedia calling services. Further, as older Private Branch Exchanges (PBXs) and network switches are phased out, industry is moving toward a voice networking model that is SIP signaled, IP based, and packet switched, not only in the wide area but also on the customer premises [2, 3].
