SIP

Session Initiation Protocol (SIP) supports the set up and tear down of media sessions, including audio teleconferencing, videoconferencing, and unicast streaming. SIP is defined in the Internet Engineering Task Force (IETF) as RFC 3261, which defines an application-level signaling protocol between end-points over an IP data network. While SIP does not actually contain the data itself, it was traditionally used hand-in-hand with the G.7xx audio codecs as well as Real-time Transport Protocol (RTP) or Real-time Streaming Protocol (RTSP) to exchange actual audio, video, or other multimedia content between participants. Recent updates include RFC 8217 and RFC 7463, the latter of which allows for “shared appearances of an Address of Record (AOR) since SIP does not have the concept of lines” when a multi-line or bridged-lines participation is initiated.