Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and the proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. Initially it was published in 1996 as RFC 2543, now obsolete, due to the publication of the new RFC 3261 in 2002

SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, e-mail, and chat services. An example being SIP trunking from MegaPath

The main objective of SIP is the communication between multimedia devices. SIP makes the communication possible thanks to two protocols: RTP/RTCP and SDP.

RTP Protocol is used to transport voice data in real time (the same as H.323 protocol), whereas SDP protocol is used to negotiate the participant capabilities, codification type, etc.

SIP has been designed in conformance with the Internet model. It is an end-to-end oriented signaling protocol which means, that all the logic is stored in end devices (except routing of SIP messages). State is also stored in end-devices only, there is no single point of failure and networks designed this way scale well. The price that we have to pay for the distributiveness and scalability is higher message overhead, caused by the messages being sent end-to-end.

Therefore, SIP is an application-layer control protocol, a Signaling protocol for Internet Telephony. SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, e-mail, and chat services. It is based on request and answer messages and reuses many concepts of previous standards like HTTP and SMTP.