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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.

SIPNOC 2017 Concludes, Was a Resounding Success!

SIPNOC 2017 was the seventh iteration of the Forum’s highly successful “SIP Network Operators Conference”.

The SIPNOC conferences attract leading technical and operations personnel from the global carrier community and have earned high praise from attendees for their educational, non-commercial and technical content that focuses on the real-world challenges operators face when deploying SIP services in global IP networks. SIPNOC attendees include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives. While the international carrier and service provider community is the lynch-pin of the SIPNOC conferences, industry stakeholders involved in major SIP initiatives such as network equipment vendors, government agency representatives, large enterprise network operators and academic research organizations have also attended.

For more information, please visit the archived SIPNOC 2017 event webpage.

For those interested, the presentations from SIPNOC 2017 are now available for download from the documents section of the website.


Federal Communications Commission Approves Notice of Inquiry (NOI) on the SHAKEN Call Authentication Framework Developed by the SIP Forum/ATIS NNI Task Force and the Publication of New Joint ATIS/SIP Forum SHAKEN Standard

ATIS and the SIP Forum have announced the publication of a new Joint ATIS/SIP Forum Standard, Signature-based Handling of Asserted information using toKENs (SHAKEN): Governance Model and Certificate Management that expands the SHAKEN framework, introducing a governance model and defining X.509 certificate management procedures. Certificate management provides mechanisms for validation of a certificate and verification of the associated digital signature, allowing for the identification of illegitimate use of national telecommunications infrastructure.

We have also announced that the United States Federal Communications Commission (FCC) has officially approved a Notice of Inquiry (NOI) on this new standard.

Read the full text of this announcement.

The SHAKEN (Signature-based Handling of Asserted information using toKENs) specification is a major advancement in industry efforts to mitigate unwanted robocalls and caller ID spoofing. Developed to efficiently implement the IETF’s STIR (Secure Telephony Identity Revisited) standard, SHAKEN defines a signature to verify the calling number and specifies how it will be transported in SIP “on the wire.” The SHAKEN framework provides guidance for service providers to implement STIR. Together, STIR/SHAKEN will offer a practical mechanism to provide verified information about the calling party as well as the origin of the call — what is known as “attestation” — for the first time in the network. Giving service providers the tools needed to sign and verify calling numbers makes it possible for consumers to know, before answering, that the calls they receive are from legitimate parties.

Read the full text of the earlier SHAKEN announcement.

The ATIS/SIP Forum NNI Task Force previously completed the first standardized IP-based network-to-network interconnection (NNI) with consensus across North American Service providers. This accomplishment enables a major objective identified in the United States National Broadband Plan, to ensure that all service connections between providers occur at the Internet Protocol (IP) level. It also helps the industry advance a major business objective of achieving the interconnection needed to reliably deliver a range of exciting new IP-based services. This work is the product of the ATIS/SIP Forum Joint Task Force, which is at the forefront of conducting an industry effort to evaluate the current state of IP-IP interconnection, identify problem areas, and provide detailed protocol-level specifications to advance solutions.

Those completed documents provide a detailed, protocol-level IP-NNI specification for one service – voice. From this initial profile, further specifications can more easily be developed. The two new ratified documents are (1) IP Interconnection Profile, which describes a reference architecture and specifications for both the protocol and media as it appears “on-the-wire” at interconnect points; and (2) IP Interconnection Routing Report, which documents mechanisms for identifying the preferred IP interconnection point for a given phone number.

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

For more information about the NNI Task Force and Charter, and to obtain copies of the completed, ratified specifications, please visit the NNI Task Force Introduction Page.

SIPconnect 2.0

SIP Forum Ratifies the SIPconnect Technical Recommendation Version 2.0!

The SIP Forum reached an important milestone by formally ratifying Version 2.0 of the SIPconnect Technical Recommendation on November 28, 2016, with the unanimous approval of the SIP Forum Board of Directors.

The formal adoption by the SIP Forum Board of the SIPconnect 2.0 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.

The SIPconnect 2.0 Technical Recommendation is a profile of the Session Initiation Protocol (SIP) and related media aspects that enables direct connectivity between a SIP-enabled Service Provider Network and a SIP-enabled Enterprise Network. It specifies the minimal set of IETF and ITU-T standards that must be supported, provides precise guidance in the areas where the standards leave multiple implementation options, and specifies a minimal set of capabilities that should be supported by the Service Provider and Enterprise Networks.

SIPconnect 2.0 effectively extends SIPconnect 1.1. Where SIPconnect 1.0, and 1.1 focused primarily on basic network registration, identity/privacy management, call originations, call terminations, and advanced services, the 2.0 version adds additional guidance on Security, Emergency Calling, and IPv6.

Where appropriate, recommendations from SIPconnect 1.1 have been left unchanged, although some modifications to prior recommendations have been made based on experience and feedback gathered through adoption of SIPconnect 1.1 in the industry.

View the full text of the SIP connect 2.0 announcement.

View or download the ratified SIPconnect Technical Recommendation Version 2.0.

View more information about the SIPconnect 2.0 Task Group charter.

SIPconnect Certified

SIP Forum Launches SIPconnect Certification Testing Program

We are proud to announce the launch of the SIPconnect Certification Testing Program, a unique certification testing program that includes a new certification test suite and test platform, as well as an associated “SIPconnect Certified” logo program that will serve as the official “seal of certification” for companies products and services that have successfully passed the certification test and officially achieved conformance with the SIPconnect specification.

The result of three years of planning and development, the new SIPconnect Certification Testing Program will be hosted by the UNH-IOL, a well-known independent, third-party laboratory dedicated to broad-based testing and standards conformance services for networking industries. The program will be available to IP communications equipment vendors, including IP-PBX and SBC vendors, and service providers who are offering SIP trunking services.

For more information, please read SIPconnect Certification Program Overview webpage.

SIP Forum 2017 Annual General Meeting

The SIP Forum 2017 General Meeting was held on Monday, 18 December 2017.

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.

View more information about the meeting, including the full text of the 2017 Annual General Meeting Notice.

SIP Forum’s SIP over IPv6 Task Group Achieves Milestone with the Publication of “Interoperability Impacts of IPv6 Interworking with Existing IPv4 SIP Implementations”

The continued proliferation of IPv6 infrastructure deployments has resulted in more IPv6 Session Initiation Protocol (SIP) User Agents (UAs) being turned up on networks around the world. Considering the large installed base of IPv4 SIP UAs deployed prior to the deployment of IPv6, it is a well-known fact that not all IPv4 SIP UAs have taken into account all possible IPv4 SIP-to-IPv6 SIP interoperability considerations at the time of their development.

The SIP Forum has announced that the SIP over IPv6 Task Group has achieved a significant milestone in its mission to improve the interoperability of IPv6 Interworking with IPv4 implementations with the publication of a new document entitled “Interoperability Impacts of IPv6 Interworking with Existing IPv4 SIP Implementations“.

View the full text of the announcement.

The SIP Forum’s SIP over IPv6 Task Group (IPv6) was formed to address key deployment and interoperability issues in the telecommunications industry’s migration to SIP over IPv6. The task group, which includes 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.


SIPit 32 Post-Event Info

SIPit 32 was held at the University of New Hampshire Interoperability Laboratory September 12-16, 2016. A Summary of test results from the event will be available soon.

In addition to the usual thorough testing, SIPit 32 focused on new evolutions in SIP Security such as the mechanisms defined in the STIR working group. The event also conducted tests designed to inform the new SIPBRANDY effort in the IETF.

Read the full SIPit 32 announcement.

View the results of past SIPit events in the SIPit section of the SIP Forum website.


SIP Forum and BITKOM Partner in the Ongoing Development of SIPconnect

The SIP Forum announced an ongoing alliance with BITKOM, the Federal Association for Information Technology, Telecommunications and New Media in Germany, in the development of updates to the SIP Forum’s SIPconnect Technical Specification (currently SIPconnect 2.0).

The formal endorsement of SIPconnect by BITKOM provides valuable technical input and peer review by one of the European Union’s most prominent technical associations, and will help ensure that the next versions of SIPconnect meet 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.


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.

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. 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.

The FoIP Task Group has developed a task group charter for viewing.

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

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

The SIP Forum’s 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 marked a significant milestone for the SIP Forum and its UA Configuration Task Group, at the time led by Chairman John Elwell, former Head of Standardization Strategy at Siemens Enterprise Communications GmbH. The UA Configuration Task Group had been working on the procedure since 2009, and designed the standard in a manner that addressed 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.

View the full text of the announcement.

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


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.

Recent News

Upcoming Events with a SIP and IP Communications Technology Focus

The SIP Forum is a proud co-sponsor of the following events. Please click the banners below for more information:

  1. Enterprise Connect Orlando 2018