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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 Working Group is available on the main home page of the group.

See also the wiki of the Media Capture Task Force.


* August 29 2018 Virtual Interim (online) CANCELLED

Background materials can sometimes be found on the Teleconferences page.


Any resolution taken in a face-to-face meeting or teleconference is to be considered provisional until 10 working days after the publication of the resolution in draft minutes sent to the working groups mailing list. If no objections are raised on the mailing list within that time, the resolution will be considered to have consensus as a resolution of the Working Group.

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.

Discussion on app control over transport related things

Transport Control

Stats interface and stats values


Discussion on what is missing to build "real" services

Missing for real services

Migration to Github for spec repositories

The WG has agreed that we would rather have the specs on github in repositories hosted under the W3C account than on other version control systems or sites. The page below tracks decisions and work on this.

Github migration

NV Documents

List of future NV Documents (for tracking): NV Documents

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