|VoIP & SIP Innovations
|IETF The Internet Engineering Task Force: SIP Working Group|
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.
|IETF: Session Initiation Proposal Investigation (SIPPING) Working Group|
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.
|IETF: Multiparty Multimedia Session Control (MMUSIC) Working Group|
The Multiparty MUltimedia SessIon Control (MMUSIC) Working Group was chartered to develop protocols to support Internet teleconferencing and multimedia communications.
|IETF: SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)|
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).
|Columbia University Department of Computer Science SIP|
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|
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.
|ITU International Telecommunication Union|
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.
|A Comparison of SIP and H.323 for Internet Telephony|
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).
|SIP versus H.323|
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.
|Comparisons between H.323 and SIP|
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 and H.323|
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.
|SIP Vs. H.323 - A Comparison|
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.
|Open Source VoIP Products
Vovida is a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments.
|SER SIP Express Router|
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.
|IP Telephony Magazines
|Internet Telephony Magazine|
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.
|The Daily Payload|
Exchange ideas on VoIP and Videoconferencing at
Forum for discussion of Voice over IP performance issues and problems.