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The Internet Engineering Task Force (IETF) is a large open international community of network designers, operators, vendors, and researchers concerned with the evolution of the Internet architecture and the smooth operation of the Internet. It is open to any interested individual. The Session Initiation Protocol (SIP) working group is chartered to continue the development of SIP, currently specified as proposed standard RFC 2543.

The Session Initiation Protocol Project INvestiGation (SIPPING) working group is chartered to document the use of SIP for several applications related to telephony and multimedia, and to develop requirements for any extensions to SIP needed for those applications.

The Multiparty MUltimedia SessIon Control (MMUSIC) Working Group was chartered to develop protocols to support Internet teleconferencing and multimedia communications.

This working group focuses on the application of the Session Initiation Protocol (SIP, RFC 3261) to the suite of services collectively known as instant messaging and presence (IMP).

The Session Initiation Protocol (SIP) research and developments of the Columbia University, Computer of Science Department, organised by the Professor Henning Schulzrinne.

The SIP Center is a portal for the commercial development of the Session Initiation Protocol. Serving both the SIP community and the wider industry.

The SIP Forum's mission is to advance the adoption of products and services based on the Session Initiation Protocol.

The ITU, headquartered in Geneva, Switzerland is an international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services.

Two standards have recently emerged for signalling and control for Internet Telephony. One is ITU Recommendation H.323 and the other the IETF Session Initian Protocol (SIP).

The IETF standards are interoperable with the ITU-T standards on the voice transport level because ITU-T incorporated IETF's  RTP protocol in its H.323 umbrella standard. However, different signaling protocols are proposed by both institutions: ITU-T uses the H.323 standard ("Visual Telephone Systems and Equipment for Local Areas Networks which Provide a Non-guaranteed Quality of Service") whereas IETF pushes the SIP signaling. Currently, there are many contraversial discussions and predictions on which approach will gain greater popularity.

This is a collection of H.323/SIP comparisons, listed in roughly chronological order, that range from accurate and up-to-date to egregious and obsolete. Responses that were originally sent as private email have been edited slightly.

SIP is, more or less, equivalent to the Q.931 and H.225 components of H.323. These protocols are responsible for call setup and call signalling. Consequently, both SIP and H.323 can be used as signalling protocols in IP networks.

As a manufacturer of telephone test equipment, we have to evaluate the potential market for both H.323 and SIP. H.323 is the more mature of the two, but problems may arise due to lack of flexibility.  SIP is currently less defined, but has greater scalability which could ease internet application integration. Which protocol will win out in the end?  It is still too early to tell, but our unbiassed analysis will help you decide which protocol best suites your application.

Vovida is a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments.

SIP Express Router (ser) is a high-performance, configurable, free SIP (RFC3261) server. It can act as SIP registrar, proxy or redirect server.

Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

The online site of the TMCnet Internet Telephony Magazine.

The online site of the CMP Media Communications Convergence Magazine.

The Authority on VoIP, Call Centers, CRM, and Telecom.

The Voiceglobe users and visitors forum for VoIP and Internet Telephony.

Exchange ideas on VoIP and Videoconferencing at …

Forum for discussion of Voice over IP performance issues and problems.

A user’s guide to the VoIP revolution