TABLE OF CONTENTS SIP is an application-layer control protocol that enables one to establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility users can maintain a single externally-visible identifier regardless of their network location. SIP works with both IPv4 and IPv6.
SIP supports five facets related to establishing and terminating multimedia communications:
User location determination of the end system to be used for communication;
User availability determination of the willingness of the called party to engage in communications;
User capabilities determination of the media and media parameters to be used;
Session setup "ringing", establishment of session parameters at both called and calling party; and,
Session management including transfer and termination of sessions, modifying session parameters, and invoking services.
SIP is not a vertically-integrated communications system. Rather, SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture. Typically, these architectures include protocols such as the Real-Time Transport Protocol (RTP) (RFC 1889) for transporting real-time data and providing QoS feedback, the Real-Time Streaming Protocol (RTSP) (RFC 2326) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) (RFC 3015) for controlling gateways to the Public Switched Telephone Network (PSTN), and, the Session Description Protocol (SDP) (RFC 2327) for describing multimedia sessions. Therefore, SIP should be used in conjunction...
