Asterisk

SIP Connection

SIP Connection Diagram

Calls Per Line: Unlimited
Requirements High speed internet connection

 

The Session Initiation Protocol (SIP) is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams.

The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games.

SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP is primarily used in setting up and tearing down voice or video calls. It has also found applications in messaging applications, such as instant messaging, and event subscription and notification.

There are a large number of SIP-related Internet Engineering Task Force (IETF) documents (Request for Comments) that define behavior for such applications. The voice and video stream communications in SIP applications are carried over another application protocol, the Real-time Transport Protocol (RTP). Parameters (port numbers, protocols, codecs) for these media streams are defined and negotiated using the Session Description Protocol (SDP) which is transported in the SIP packet body.


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