Session Initiation Protocol
Voice-over-IP communications protocol / From Wikipedia, the free encyclopedia
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The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications.[1] SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).[2]
Communication protocol | |
Abbreviation | SIP |
---|---|
Purpose | Internet telephony |
Introduction | March 1999; 25 years ago (1999-03) |
OSI layer | Application layer (Layer 7) |
Port(s) | 5060, 5061 |
RFC(s) | 2543, 3261 |
The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).[3] A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message.
SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP).