Session Initiation Protocol (SIP) |
Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences). A session is considered as an exchange of data between an association of participants, such as Internet telephony calls and videotelephony. SIP is able to support multicast conferences with more than two participants. Participants can be invited to already existing sessions. Media can be added to (and removed from) an existing session.
SIP supports personal mobility - users can maintain a single externally visible identifier regardless of their network location. SIP transparently supports name mapping and redirection services, to support this mobility.
SIP supports five facets of establishing and terminating multimedia communications:
- User location: determination of the end system to be used for communication;
- User availability: determination of the willingness of the called party to engage in communications;
- User capabilities: determination of the media and media parameters to be used;
- Session setup: "ringing", establishment of session parameters at both called and calling party;
- Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.
SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other Internet protocols to build a complete multimedia architecture. However, the basic functionality and operation of SIP does not depend on any of these protocols.
SIP is described in RFC 3261.
The Session Initiation Protocol is a signaling protocol that enables the Voice Over Internet Protocol (VoIP) by defining the messages sent between endpoints and managing the actual elements of a call. SIP supports voice calls, video conferencing, instant messaging, and media distribution. Check this blog post for in-depth information on SIP architecture, messages, and methods!
SIP is just one method of deploying VoIP; its primary benefit is the fact that it provides a direct connection between private or local telephone systems (private branch exchanges, or PBX) and the public telephone network. This way, individuals and businesses don't need a legacy telephone line in order to connect.
Other VoIP deployment methods include the Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), and Session Description Protocol (SDP).
An SIP trunk is the interconnection between two domains of the Unified Communications network. By creating these interconnections, SIP trunking allows us to partition the network into public and private domains. Public domains are generally managed by an internet telephone service provider, while private domains are connected to someone's personal server. Internet telephone service provider use SIP trunking to securely deliver telephone and streaming media services to users equipped with private branch exchange.
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.[1] 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).
You must be logged in to post a comment.