Session Initiation Protocol (SIP) is the Internet Engineering Task Force’s (IETF‘s) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more end points.SIp has been defined in RFC 2543.
SIP is designed to address the functions of signaling and session management within an IP packet telephony network. Signaling allows call information to be carried across networks and Session management provides the ability to control the attributes of a call end to end.
H.323 is a standard approved by the International Telecommunication Union (ITU) and defines the protocols to provide audio-visual communication sessions on any packet network. H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.H.323 is widely implemented by voice and videoconferencing equipment manufacturers and used within various Internet real-time applications like netmeeting etc.
The individual protocols used under the umbrella of H.323 include:
- 225.0 for call signalling;
- 931, a protocol borrowed from ISDN, also used for call signalling;
- 245 for negotiating audio/video channel parameters;
- 235 for security and authentication;
- RTP, the Real Time Protocol defined by IETF, used to transmit audio/video streams;
- 450.x for additional services like call transfer, call diversion, etc.
Below table enlists the comparison of both protocols –
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