- 1 Web Real-Time Communications Working Group
- 1.1 Meetings
- 1.2 On-going work
- 1.3 Useful Resources
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.
- November 11 - November 12 2013 in Shenzen, China
- February 6 - February 7 2013 in Boston, USA
- October 29 - October 30 2012 in Lyon, France
- June 11 2012 in Stockholm, Sweden
- February 1 2012 Feb 1st, Mountain View, CA
- October 31 - November 1 2011 in Santa Clara, California, USA
- July 23 2011 in Quebec City, Canada
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.
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.
Stats interface and stats values
Discussion on what is missing to build "real" services
- Archives of the email@example.com mailing-list are publicly available. This mailing-list should be used for all technical discussions within the group.
- Archives of the firstname.lastname@example.org mailing-list are available to W3C Members only. Use of this second list should be restricted to administrative purposes (e.g. sending regrets or F2F logistics)
- 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:
- IRC info: #webrtc on irc.w3.org, port 6665
- You may use the W3C Web client, specifying a username and channel #webrtc
- You may find more info on IRC usage at W3C