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The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions. The most common applications of SIP are in Internet telephony for voice and video calls, as well as instant messaging all over Internet Protocol (IP) networks. The protocol defines the messages that are sent between endpoints, which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP is an application layer protocol designed to be independent of the underlying transport layer. It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). SIP works in conjunction with several other application layer protocols that identify and carry the session media. Media identification and negotiation is achieved with the Session Description Protocol (SDP). For the transmission of media streams (voice, video) SIP typically employs the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages, the protocol may be encrypted with Transport Layer Security (TLS). ==History== SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996. The protocol was standardized as RFC 2543 in 1999 (SIP 1.0). 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 systems. , the latest version (SIP 2.0) of the specification is RFC 3261, published in June 2002, with extensions and clarifications since then.〔(【引用サイトリンク】url=https://datatracker.ietf.org/doc/search/?name=SIP&rfcs=on&sort=date )〕 The U.S. National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public-domain Java implementation〔(【引用サイトリンク】title=JAIN SIP project )〕 that serves as a reference implementation for the standard. The implementation can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects. It supports RFC 3261 in full and a number of extension RFCs including RFC 6665 (event notification) and RFC 3262 (reliable provisional responses). While originally developed based on voice applications, the protocol was envisioned and supports a diverse array of applications, including video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.〔(【引用サイトリンク】url=http://www.networkworld.com/article/2332980/lan-wan/what-is-sip-.html )〕〔(【引用サイトリンク】url=http://www.voip-info.org/wiki/view/SIP )〕〔(【引用サイトリンク】url=https://tools.ietf.org/html/rfc3261 )〕〔(【引用サイトリンク】title=Session Initiation Protocol (SIP) )〕 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Session Initiation Protocol」の詳細全文を読む スポンサード リンク
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