SIP is primarily used only for setting up, modifying, tearing down the media session (voice, video,..) between 2 (unicast) or more parties (multicast).
SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build a complete multimedia architecture. Typically, these architectures will include protocols such as the Real-time Transport Protocol (RTP) for transporting real-time data and providing QoS feedback, the Real-Time streaming protocol (RTSP) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) for describing multimedia sessions. Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.
SIP fits in to Application Layer of TCP/IP Model, which is designed to work independent of the underlying transport layer protocol. SIP is a text based protocol incorporating many elements of HTTP, SMTP.
Outline for functionalities of SIP:
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;
Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.
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