WebRTC Task Group Charter Print

 


WebRTC Task Group Charter


 

 

Mailing list:

webrtc "at" sipforum.org -- [Subscribe] [Archive]
SIP Forum Technical Working Group Director

Spencer Dawkins -- spencer "at" wonderhamster "dot" org (Huawei)
Task Group Co-Chairs:

Victor Pascual Avila - victor.pascual.avila "at" gmail "dot" com
Gonzalo Salgueiro -- gsalguei "at" cisco "dot" com (Cisco Systems)
Task Group Chief Document Editor:

TBA
Meetings:

 

Meetings take place on a schedule established by the task group. You can stay informed about upcoming meetings by subscribing to the webrtc email list (see link above).

 


 

Problem Statement

 

The W3C WebRTC and IETF RTCWeb Working Groups are defining standards for Web-based real-time communication, to enable Web browsers to communicate using audio and video over RTP/RTCP based on a Javascript/ECMAScript API.

One clear and immediate use-case for WebRTC is enabling communication with existing SIP-based deployments, whether they are Enterprise or Service Provider SIP domains. In order to achieve interoperability, however, interworking functions may need to be performed in both the signaling and media planes. The definition of such interworking functions, and whether if or when they are needed, will not be addressed by the IETF or W3C, and is more appropriate for the SIP Forum.

 


 

WebRTC Task Group Charter

 

The objective of the WebRTC-SIP Task Group is to enable deployment of WebRTC for SIP-based domains. This may entail creating new SIP Forum Recommendations, Reference Architecture Documents, Certifications, and/or White Papers. The initial focus of the Task Group is to determine what the needs are for successful interoperability of WebRTC-to-SIP deployments, in terms of such new deliverables for the SIP Forum.

The focus of the work covers the following deployment scenarios:

  • An Enterprise WebRTC domain communicating with the same Enterprise's SIP domain, for audio and video services.
  • An Enterprise WebRTC domain communicating with a Service Provider's SIP domain using SIPconnect, for audio and video services.
  • A Service Provider WebRTC domain communicating with the same Service Provider's SIP domain, for audio and video services.
  • A Service Provider WebRTC domain communicating with an Enterprise's SIP domain using SIPconnect, for audio and video services.
  • Two WebRTC domains communicating with each other using SIP between the domains.

The scope of the work includes the following topic areas:

  • Interoperability of the communication session initiation, modification, and termination.
  • Interoperability of audio and video media.
  • Interoperability of identity information of the parties in the communication session.
  • Security considerations, vulnerabilities, and best practices.

 


 

Deliverables and Milestones

 

Goals for documents agreed for publication are:

  • July 2013: Completion of evaluation of needed work items.
  • August 2013: Re-charter with new work items, based on above evaluation.

 


 

Participation in the Task Group

 

The SIP Forum has appointed Victor Pascual Avila, Oracle, and Gonzalo Salgueiro, Cisco Systems, as the co-chairs for this task group to coordinate the creation of the various activities of the group. The various activities are intended to be open to all members—full, academic and participant—as determined by the Task Groups and the Working Group Chairs.

 


 

Operations

 

The WebRTC Task Group will establish various activities as required to support its activities. The specific meeting and operational schedule for each activity will be established by the chair.

 


 

WebRTC Task Group Document Repository

 

A special WebRTC TG Document repository, which will contain documentation relevant to the work at hand, is located HERE.

 


 

SIP Forum Announces New WebRTC Interoperability Testing Event

 

RTCWeb-it 2, hosted by TMC and scheduled for November 18, 2013 -- on the Monday before the WebRTC Conference and Expo -- is the second WebRTC interoperability testing event produced by the SIP Forum. This special event allows 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 will bring 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.

Like the SIPit Interoperability testing events, RTCWeb-it 2 will be a technology test event, and not product or application showcase, nor a marketing/networking event for general industry participants.

Registration for qualified participants is now open, and is located at www.regonline.com/rtcwebit2

 


 

For More Information

 

For more information, please contact Marc Robins, SIP Forum Managing Director, at marc "dot" robins "at" sipforum.org or call +1-203-829-6307.

 

 

SIP FORUM MEMBER

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