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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 31 Jan 2017 17:00:40 +0000
To: public-webrtc@w3.org
Message-Id: <E1cYbnM-0001cb-Of@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+1/-4/💬22)
  1 issues created:
  - RTCDataChannel close event not defined (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1009

  4 issues closed:
  - RTCDataChannel CloseEvent not defined 
https://github.com/w3c/webrtc-pc/issues/1009
  - NetworkError event is not defined and might not be needed 
https://github.com/w3c/webrtc-pc/issues/526
  - Specify when a data channel's ID is assigned, and what the `id` 
attribute returns when no ID is assigned. 
https://github.com/w3c/webrtc-pc/issues/795
  - When is an RTCSctpTransport created and destroyed? 
https://github.com/w3c/webrtc-pc/issues/979

  9 issues received 22 new comments:
  - #763 Handling of simulcast errors (6 by taylor-b, alvestrand, 
aboba)
    https://github.com/w3c/webrtc-pc/issues/763
  - #795 Specify when a data channel's ID is assigned, and what the 
`id` attribute returns when no ID is assigned. (5 by taylor-b, 
alvestrand)
    https://github.com/w3c/webrtc-pc/issues/795
  - #964 Handing SDP with more than one identity  (2 by alvestrand, 
stefhak)
    https://github.com/w3c/webrtc-pc/issues/964
  - #526 NetworkError event is not defined and might not be needed (2 
by aboba)
    https://github.com/w3c/webrtc-pc/issues/526
  - #979 When is an RTCSctpTransport created and destroyed? (2 by 
taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/979
  - #714 STUN/TURN OAuth token auth parameter passing (2 by misi, 
alvestrand)
    https://github.com/w3c/webrtc-pc/issues/714
  - #997 RTCRtcpMuxPolicy of "negotiate" should be marked as an 
"at-risk" feature (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/997
  - #687 Clarify reasoning behind and mitigation of privacy issues 
(PING review) (1 by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/687
  - #671 Processing remote MediaStreamTracks without MediaStreams info
 (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/671

* w3c/webrtc-stats (+6/-1/💬11)
  6 issues created:
  - Stat for audio acceleration rate? (by henbos)
    https://github.com/w3c/webrtc-stats/issues/146
  - RTCMediaStreamTrackStats.audioLevel... input vs output? (by 
henbos)
    https://github.com/w3c/webrtc-stats/issues/145
  - Stat for how many video adaption changes (by henbos)
    https://github.com/w3c/webrtc-stats/issues/144
  - RTCCodecStats.implementation... per-codec, per-stream? (by henbos)
    https://github.com/w3c/webrtc-stats/issues/143
  - Unclear which framerate "framePerSecond" represents (by taylor-b)
    https://github.com/w3c/webrtc-stats/issues/141
  - RTCDataChannelStats should reference RTCTransportStats (by henbos)
    https://github.com/w3c/webrtc-stats/issues/140

  1 issues closed:
  - RTCMediaStreamTrackStats.audioLevel... input vs output? 
https://github.com/w3c/webrtc-stats/issues/145

  4 issues received 11 new comments:
  - #141 Unclear which framerate "framePerSecond" represents (6 by 
gunnarhm, taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/141
  - #143 RTCCodecStats.implementation... per-codec, per-stream? (3 by 
henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/143
  - #96 Not clear how to differentiate between received connectivity 
checks and consent requests (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/96
  - #145 RTCMediaStreamTrackStats.audioLevel... input vs output? (1 by
 henbos)
    https://github.com/w3c/webrtc-stats/issues/145



Pull requests
-------------
* w3c/webrtc-pc (+13/-4/💬10)
  13 pull requests submitted:
  - More complete example (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1012
  - Eliminate NetworkError (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1011
  - RTCDataChannelCloseEvent definition (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1010
  - Updating RTCDataChannel.close and onclose EventHandler (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1008
  - Freeze version of respec-w3c-common.js (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1007
  - errorCode required in RTCPeerConnectionIceErrorEventInit (by 
aboba)
    https://github.com/w3c/webrtc-pc/pull/1006
  - Add offerToReceive* as legacy extensions (by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1005
  - currentRemoteDescription.sdp need not match 
setRemoteDescription().sdp (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1004
  - Change SetLocalDescription to require unchanged offer/answer 
string (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1003
  - Event when a transceiver is stopped via remote action (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1002
  - Effect of a BYE on RtpReceiver.track (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1001
  - Separated auth dictionaries for STUN/TURN (by misi)
    https://github.com/w3c/webrtc-pc/pull/1000
  - transceiver.stop() sends a BYE (by aboba)
    https://github.com/w3c/webrtc-pc/pull/999

  4 pull requests merged:
  - Event when a transceiver is stopped via remote action
    https://github.com/w3c/webrtc-pc/pull/1002
  - transceiver.stop() sends a BYE
    https://github.com/w3c/webrtc-pc/pull/999
  - Describe when an RTCSctpTransport is created/set to null.
    https://github.com/w3c/webrtc-pc/pull/996
  - Changing "non-null" to "missing" to match IDL terminology.
    https://github.com/w3c/webrtc-pc/pull/994

  7 pull requests received 10 new comments:
  - #1004 currentRemoteDescription.sdp need not match 
remoteDescription.sdp (3 by fippo, alvestrand, aboba)
    https://github.com/w3c/webrtc-pc/pull/1004
  - #1001 Effect of a BYE on RtpReceiver.track (2 by alvestrand, 
aboba)
    https://github.com/w3c/webrtc-pc/pull/1001
  - #1002 Event when a transceiver is stopped via remote action (1 by 
stefhak)
    https://github.com/w3c/webrtc-pc/pull/1002
  - #971 Clarify wording on TypeError from addIceCandidate. (1 by 
alvestrand)
    https://github.com/w3c/webrtc-pc/pull/971
  - #1005 Add offerToReceive* as legacy extensions (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1005
  - #996 Describe when an RTCSctpTransport is created/set to null. (1 
by aboba)
    https://github.com/w3c/webrtc-pc/pull/996
  - #988 Add RTCOfferOptions.reofferOptions. (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/988

* w3c/webrtc-stats (+1/-0/💬2)
  1 pull requests submitted:
  - Rename RTCRtpMediaStreamStats.trackId (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/142

  2 pull requests received 2 new comments:
  - #139 Define terminology for "stats object" et al. (1 by 
alvestrand)
    https://github.com/w3c/webrtc-stats/pull/139
  - #142 Rename RTCRtpMediaStreamStats.trackId (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/142


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 31 January 2017 17:00:47 UTC

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