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WELCOME TO THE WEBSITE OF THE

 

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The SIP Forum is an industry association with members from the leading IP communications companies. Its mission: To advance the adoption and interoperability of IP communications products and services based on SIP.

The Forum promotes SIP as the technology of choice for the control of real-time multimedia communication sessions throughout the Internet, corporate networks, and wireless networks.

The Forum directs technical activities aimed at achieving high levels of product interoperability, provides information on the benefits and capabilities of SIP, and highlights successful applications and deployments.

Each of our Working Groups and Task Groups has their own mailing list, many of which are open to individual, "Participant" members. Please feel free to join them, or to join the general "discussion" mailing list.

The SIP Forum is currently engaged in the following technical activities -- each of which has its own Task Group that includes a unique mailing list, chairperson(s) and group of contributors:

The Forum is open to individual "Participant" members, Academic/Institutional Members, and to corporate "Full Members". Individual "Participant" and Academic/Institutional membership is free.

To view our current Full Member Listing, click HERE. To view our Academic Member Listing, please click HERE.

You can find out more about membership here.

Need more information or logos? Visit our Press Room HERE.

 


 

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SIP Forum/ATIS NNI Joint Task Force Draft Documents Published for Comment

The IP-Network to Network Interface (NNI) Joint Task Force, a cooperative effort between the ATIS and the SIP Forum, is seeking public comment on two draft documents:

1. IP Interconnection Profile. The IP-NNI Profile Specification defines a reference architecture and specifications for both the protocol and media as it appears “on-the-wire” at interconnect points. The specifications reference commonly used IETF, 3GPP, and other related industry specifications and identify protocol extensions and capability information needed for all-IP telephony peering.

2. IP Interconnection Routing Report. The IP-NNI Routing Technical Report documents mechanisms for identifying the preferred IP interconnection point for a given TN. This report presents multiple views of current IP interconnection mechanisms based on aggregate PSTN constructs, interim solutions based on all-IP routing using a per-TN registry, and a consideration of hybrid approaches across both mechanisms during the transition to all-IP.

Access the documents HERE (access is public).

Providing Feedback: Written input, technical feedback only, is requested by October 29, 2014. However, in light of the short timeframe, input will be accepted until December 1, 2014. Submit comments to the NNI Task Force via its mailing list.

To do this, individuals must first be registered as a SIP Forum Participant Member (Free) here: http://www.sipforum.org/component/option,com_advanced_registration/task,register/.

When completed, a subscription link to the NNI Task Force mailing list can be found on the NNI Task Force Charter webpage.

ATIS and the SIP Forum formed the Joint Task Force to fully specify an IP communications network-to-network interface (NNI) between North American service providers. The Joint Task Force will help enable the goal, identified in the United States National Broadband Plan, of ensuring all service interconnection between providers occur at the Internet Protocol (IP) level. By enabling ubiquitous IP-based interconnection, the new Task Force supports wide-scale availability of IP-based voice services, and lays the groundwork for ubiquitous advanced real-time communications such as high-definition voice, point-to-point video calling, and multimedia text across wireless, wireline and cable providers.

Read the full text of the ATIS and SIP Forum IP-NNI Task Force announcement.

 


 

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SIP Forum Announces the Launch of SIPconnect 2.0!

Work has begun on SIPconnect 2.0!

The new SIPconnect 2.0 task group will produce a SIP Forum SIPconnect 2.0 recommendation that defines a common set of implementation rules for implementers who desire to interface an IP PBX with a SIP-enabled service provider. The recommendation(s) will specify which standards must be supported, provide precise guidance in the areas where the standards leave multiple options, supplement functionality gaps in existing protocols, and identify a baseline set of features that should be common to an IP PBX to service provider interface.

SIPconnect 2.0 will support a variety of new applications and technologies, including IPV6 and IPV4/6 Dual Stack components within the reference architecture, secure media (SRTP), video-enabled devices, and emergency calling (NG911/NG112) and the transport of location information.

For more information about the new and updated SIPconnect 2.0 charter, please visit HERE.

 


 

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SIPNOC 2014 Presentations Now Available

SIPNOC 2014 -- the fourth annual "SIP Network Operators Conference" -- concluded on June 12, 2014, and the presentations from the event are now available for viewing and downloading from the SIP Forum website HERE. Please note that you must be a registered SIP Forum participant member and logged into your account in order to get access!.

Topics explored at SIPNOC 2014 included WebRTC, codec handling, SIP trunking design and deployment, the SIP Forum’s various technical initiatives, Fax over Internet protocol (FoIP) interworking, implementing SIP with IPv6, toll fraud detection and prevention, security best-practices, HD Voice deployment, testing methodologies, applications development, call routing and peering, troubleshooting and monitoring, emergency services, SIP market research findings, standards developments, important regulatory and policy developments, and much more.

For more information about SIPNOC 2014, please visit the archived event webpage HERE.

 


 

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SIPit 31 Results Now Available

SIPit 31 was held in Nice, France, September 29-October 3, 2014, and was hosted by ETSI.

For more information about the SIPit events, please visit www.sipit.net.

 


 

SIP Forum 2013 Annual General Meeting Results

The SIP Forum 2013 General Meeting was held on Monday, 09 December 2013.

In the meeting, a review of the financial operations was conducted, as well as a review of the operational and technical activities that have occurred over the past 12 months since the last General Meeting. In addition, votes were cast for new board members and for one other voting issue.

For more information about the meeting, including the voting results, the full text of the 2013 Annual General Meeting Notice, and the presentations given during the meeting, please click HERE.

 


 

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SIP Forum Reports Results From RTCWeb-it 2 Interoperability Testing Event

RTCWeb-it 2, hosted by TMC, was the second WebRTC interoperability testing event produced by the SIP Forum, and was held in Santa Clara, California, Monday, November 18, 2013. This special event allowed WebRTC implementers above and below the WebRTC API to work together with the specification authors to improve their code and the specifications.

RTCWeb-it 2 brought together the browser developers, and other implementers working on the code that makes the WebRTC API work; Application developers that are using the WebRTC API; Gateway developers, particularly those building systems that bridge WebRTC and SIP; and the editors of several of the specifications being developed.

RTCWeb-it 2 was a very successful test session, as most of the scenarios that were exercised "just worked" the first time. There were two browser implementations, and one application suite at the event. The scenarios tested included a rich combination of NAT and firewall restricted network paths, ensuring that the expected media and data channel path was taken in each scenario. (This included forcing the browsers to communicate only through a Turn server.)

One observation from the tests is that implementations cannot rely on streams being available until onAddStream is called. This may or may not happen before the setRemoteDescription success callback.

We found a few deployed applications that are currently changing their behavior based on the User-Agent string, and have cost themselves starting to automatically work cross-browser as the browsers are being updated. New application developers should isolate such decisions. (It would have been nice to disable these checks to see if cross-browser DataChannels worked with the browser code.)

Having a rich application suite available helped drive effective tests, and provided very useful feedback for the application developer. All application developers should consider attending the next event to get the quick feedback this kind of testing environment provides.

 


 

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SIP Forum and BITKOM Announce New Partnership in the Ongoing Development of SIPconnect

The SIP Forum announced a new alliance with BITKOM, the Federal Association for Information Technology, Telecommunications and New Media in Germany, in the development of the next update to the SIP Forum’s SIPconnect 1.1 Technical Specification.

The formal endorsement of SIPconnect by BITKOM will provide valuable technical input and peer review by one of the European Union’s most prominent technical associations, and help ensure that the next version of SIPconnect meets the requirements of BITKOM members. As part of the agreement, the two organizations will pool requirements as they work towards an update of their respective SIP trunking specifications.

Read the full text of the SIP Forum and BITKOM Announcement.

 


 

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SIP Forum’s Fax over IP Task Group Achieves Milestone with the Publication of IETF RFC 6913

The SIP Forum announced its Fax over IP (FoIP) Task Group has achieved a significant milestone in its mission to improve international IP fax transport services with the publication of RFC 6913 – a new Internet Engineering Task Force (IETF) RFC that introduces a new “sip.fax” media feature tag that aims to enable the intelligent routing of International faxes and greatly improve the reliability of International faxing services.

RFC 6913, co-authored by David Hanes, Kevin Fleming and Gonzalo Salgueiro, defines and registers with IANA a new "fax" media feature tag for use with the Session Initiation Protocol (SIP). Currently, fax calls are indistinguishable from voice calls at call initiation. Consequently, fax calls can be routed to SIP user agents that are not fax capable. A "fax" media feature tag implemented in conjunction with caller preferences allows for early advertisement of fax capabilities and consequently, more intelligent fax call routing.

Click here to see the full text of the RFC 6913 announcement.

As part of the work leading up to this new RFC, the Fax-over-IP Interoperability Task Group published a T.38 SIP-SDP subgroup Problem Statement that details a number of SDP offer/answer interoperability issues found while implementing and connecting T.38 compliant endpoints together, primarily over the SIP signaling mechanism. Click HERE to view or download this document.

In addition, the Forum previously published an official task group problem statement, a document that details the various interoperability issues that currently plague FoIP services. This document is available for download HERE.

The FoIP Task Group has developed a task group charter, which you can view HERE.

To view task group documents and other related materials, please visit the FoIP Task Group document repository HERE.

 


 

SIP Forum Forms New SIP over IPv6 Task Group

The SIP Forum has announced the formation of a new SIP over IPv6 Task Group (IPv6) to address key deployment and interoperability issues in the telecommunications industry’s migration to SIP over IPv6. The new task group, which will enlist key stakeholders from the service provider, application developer and equipment communities, has developed and ratified a charter with the mandate to identify issues with SIP over IPv6 and assess the impact of transition technologies and dual stack devices on existing SIP networks.

For more information, please visit the SIP over IPv6 Task Group Charter webpage.

 


 

SIP Forum Names Spencer Dawkins Technical Director

The SIP Forum announced today the selection of Spencer Dawkins as its new Technical Director. A longtime member of the SIP Forum and major contributor to Forum work, Dawkins brings to this role about a decade of leadership experience helping to advance SIP and SIP-related technologies within industry and international standards bodies.

As Technical Director of the Forum’s Technical Working Group, Mr. Dawkins will work with SIP Forum management to translate strategic priorities from the SIP Forum Board of Directors into action, developing these directives into key SIP Forum technical initiatives for implementation by SIP Forum task groups. Mr. Dawkins will work closely with each group including the SIPconnect Task Group, new SIPconnect Interoperability Certification Task Group (SC-IT), User Agent Configuration Task Group, Fax over IP Task Group and the Test Event Working Group (SIPit).

For the full text of this announcement, please click HERE.

 


 

SIPconnect Technical Recommendation

 

SIP Forum Ratifies the SIPconnect Technical Recommendation Version 1.1!

The SIP Forum reached an important milestone by formally ratifying version 1.1 of the SIPconnect Technical Recommendation on March 1, 2011, with the unanimous approval of the SIP Forum Board of Directors.

The formal adoption by the SIP Forum Board of the SIPconnect 1.1 Technical Recommendation is based on recognition that the recommendation has been through credible peer review, including broad membership and significant community review, that it is stable and is well-understood, and that it is believed to have resolved known design choices.

For more information, please click HERE to read the full text of the announcement or visit the main SIPconnect page HERE.

To view or download the ratified SIPconnect Technical Recommendation Version 1.1, please click HERE.

The SIP Forum has created a repository of archived SIPconnect 1.1 documents, which includes submissions from Avaya, Broadsoft, CableLabs, Cbeyond, MetaSwitch, Microsoft and Siemens, and contains various draft iterations, including v.00 through v.27 documents.

Please click here for the SIPconnect 1.1 Document repository.

 


 

SIP Forum UA Configuration Recommendation Ratified and Published as RFC 6011 by the IETF

The SIP Forum has announced that its User Agent Configuration Recommendation for the locating, retrieving and maintaining of SIP User Agents has been ratified and published as RFC 6011 by the Internet Engineering Task Force (IETF).

The publishing of the recommendation as RFC 6011 marks a significant milestone for the SIP Forum, the leading IP communications industry association, and its UA Configuration Task Group, led by Chairman John Elwell, the Head of Standardization Strategy at Siemens Enterprise Communications GmbH. The UA Configuration Task Group has been working on the procedure since 2009, and has designed the standard in a manner that addresses the needs of end users and services providers, as well as both small businesses and large enterprises deploying SIP-enabled endpoints.

Now published as RFC 6011 by the IETF’s Internet Engineering Steering Group (IESG) after a public review process by the IETF community, the User Agent Configuration Recommendation (UA Config) sets a standard procedure for how a SIP User Agent locates, retrieves and maintains current configuration information for a given SIP Service Provider. It requires that each User Agent, the configuration agent at the service provider and network infrastructure meet such requirements to ensure communication.

To view the full text of the announcement, please click HERE.

To view the published text of RFC 6011, please visit http://tools.ietf.org/search/rfc6011.

 


 

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