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== Meetings ==
== Meetings ==
* [[February 6 - February 7 2013]] in Boston, USA
* [[October 29 - October 30 2012]] in Lyon, France
* [[October 29 - October 30 2012]] in Lyon, France
* [[June 11 2012]] in Stockholm, Sweden
* [[June 11 2012]] in Stockholm, Sweden

Revision as of 07:30, 29 January 2013

Web Real-Time Communications Working Group

This is the Wiki of the W3C Real-Time Communications Working Group. The wiki is the main point of entry for participants of the group as well as for people interesting in the day-to-day life of the group. It is used to collect drafts, ideas and logistics information. Higher-level information about the Web Real-Time Communications Workin Group is available on the main home page of the group.


In addition, there are teleconferences at varying intervals between these. Background materials can sometimes be found on the Teleconferences page.

Minutes from meetings and teleconferences are linked from the official WG page.

On-going work

API document

Editors' draft

Use cases and requirements

The Editor's draft of Use cases and requirements currently merely refers to the equivalent IETF document. In the IETF document use cases and requirements are listed. New use cases that have been proposed but not yet decided are also listed (with links).

There are however some items that should be discussed more within the context of current use cases and requirements (mostly taken from this mail):

  • QoS. Unclear to a large extent at this time. Skip for first version?
  • The use case (4.2.5) of using SIP between service providers must be sorted out to know what "SIP" really means and furthermore if all webrtc functionality would be available or if there are certain things you can't do. Also F24 should probably be updated, along with a new API requirement (text proposed from the mail "Web API MUST NOT prevent two webapps that happen to choose to peer with SIP from peering."). Also, the current A18 is vague and it can be discussed how much this is an API question
  • What of more "advanced" audio media processing should remain in webrtc, and what should be moved to the Audio WG? Will have implications on F13 and F15 and A14.
  • A more advanced IVR use case, or at least more detailed should perhaps replace the text in 4.3.2
  • Codec requirements.....
  • Section 7.2 (Security): should these requirements be moved into the F-req part?
  • More work needed on data channels (unreliable, reliable, file-transfer) in terms of actual use and of requirements.

Useful Resources



  • Tracker is used to track actions within the group. We're only using the ACTION section.
  • Buganizer is available for tracking issues and bugs.

The Tracker usage guidelines (Work In Progress) gives details on how and when to use these tools.


The group uses IRC to take minutes and exchange info during calls and face-to-face meetings:

Other Useful W3C Resources