Session Initiation Protocol

The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE). The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE). 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). 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 is 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). SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996. The protocol was standardized as .mw-parser-output cite.citation{font-style:inherit}.mw-parser-output .citation q{quotes:''''''''''''}.mw-parser-output .citation .cs1-lock-free a{background:url('//upload.wikimedia.org/wikipedia/commons/thumb/6/65/Lock-green.svg/9px-Lock-green.svg.png')no-repeat;background-position:right .1em center}.mw-parser-output .citation .cs1-lock-limited a,.mw-parser-output .citation .cs1-lock-registration a{background:url('//upload.wikimedia.org/wikipedia/commons/thumb/d/d6/Lock-gray-alt-2.svg/9px-Lock-gray-alt-2.svg.png')no-repeat;background-position:right .1em center}.mw-parser-output .citation .cs1-lock-subscription a{background:url('//upload.wikimedia.org/wikipedia/commons/thumb/a/aa/Lock-red-alt-2.svg/9px-Lock-red-alt-2.svg.png')no-repeat;background-position:right .1em center}.mw-parser-output .cs1-subscription,.mw-parser-output .cs1-registration{color:#555}.mw-parser-output .cs1-subscription span,.mw-parser-output .cs1-registration span{border-bottom:1px dotted;cursor:help}.mw-parser-output .cs1-ws-icon a{background:url('//upload.wikimedia.org/wikipedia/commons/thumb/4/4c/Wikisource-logo.svg/12px-Wikisource-logo.svg.png')no-repeat;background-position:right .1em center}.mw-parser-output code.cs1-code{color:inherit;background:inherit;border:inherit;padding:inherit}.mw-parser-output .cs1-hidden-error{display:none;font-size:100%}.mw-parser-output .cs1-visible-error{font-size:100%}.mw-parser-output .cs1-maint{display:none;color:#33aa33;margin-left:0.3em}.mw-parser-output .cs1-subscription,.mw-parser-output .cs1-registration,.mw-parser-output .cs1-format{font-size:95%}.mw-parser-output .cs1-kern-left,.mw-parser-output .cs1-kern-wl-left{padding-left:0.2em}.mw-parser-output .cs1-kern-right,.mw-parser-output .cs1-kern-wl-right{padding-right:0.2em}RFC 2543 in 1999. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular networks. In June 2002 the specification was revised in RFC 3261 and various extensions and clarifications have been published since. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network (PSTN) with a vision of supporting new multimedia applications. It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. SIP has been standardized primarily by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU). SIP is only involved for the signaling operations of a media communication session and is primarily used to set up and terminate voice or video calls. SIP can be used to establish two-party (unicast) or multiparty (multicast) sessions. It also allows modification of existing calls. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. SIP works in conjunction with several other protocols that specify the media format and coding and that carry the media once the call is set up. For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol. Voice and video media streams are typically carried between the terminals using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). Every resource of a SIP network, such as user agents, call routers, and voicemail boxes, are identified by a Uniform Resource Identifier (URI). The syntax of the URI follows the general standard syntax also used in Web services and e-mail. The URI scheme used for SIP is sip and a typical SIP URI has the form sip:username@domainname or sip:username@hostport, where domainname requires DNS SRV records to locate the servers for SIP domain while hostport can be an IP address or a fully qualified domain name of the host and port. If secure transmission is required, the scheme sips is used.