Weekly github digest (WebRTC WG specifications)

Issues
------
* w3c/webrtc-pc (+3/-1/💬31)
  3 issues created:
  - relay-first as an option for RTCIceTransportPolicy (by vr000m)
    https://github.com/w3c/webrtc-pc/issues/1658
  - [[SctpTransport]] slot needs to be defined and initialized (by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1653
  - RTCPeerConnection.close should only refer to one RTCSctpTransport (by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1652

  14 issues received 31 new comments:
  - #1446 RTCSctpTransport.maxMessageSize 0 case (9 by adam-be, lgrahl, nils-ohlmeier)
    https://github.com/w3c/webrtc-pc/issues/1446
  - #1295 Section 11: Examples (3 by adam-be, Jxck, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1295
  - #1617 Adopt "test as you commit" policy (3 by foolip, youennf)
    https://github.com/w3c/webrtc-pc/issues/1617
  - #1651 JSEP references are out dated (2 by fluffy, vivienlacourba)
    https://github.com/w3c/webrtc-pc/issues/1651
  - #1644 Adding more values to RTCIceTransportPolicy Enum (2 by jianjunz, nils-ohlmeier)
    https://github.com/w3c/webrtc-pc/issues/1644
  - #1613 Stats & isolated streams (2 by dontcallmedom, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1613
  - #1646 Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs (2 by soareschen, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1646
  - #1619 canInsertDTMF transitions not specified (2 by fluffy, aboba)
    https://github.com/w3c/webrtc-pc/issues/1619
  - #1635 Need for Initial Bitrate by the Application/RtpSender? (1 by fluffy)
    https://github.com/w3c/webrtc-pc/issues/1635
  - #942 Meta: auto-publish changes to the spec (1 by foolip)
    https://github.com/w3c/webrtc-pc/issues/942
  - #1271 Inconsistencies in Asynchronous Task Queueing (1 by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1271
  - #1658 relay-first as an option for RTCIceTransportPolicy (1 by fippo)
    https://github.com/w3c/webrtc-pc/issues/1658
  - #1533 Clarify whether RTCRtpContributingSource members are live. (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1533
  - #1288 Section 6.4: Datachannel Garbage Collection  (1 by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1288

  1 issues closed:
  - Section 6.4: Datachannel Garbage Collection  https://github.com/w3c/webrtc-pc/issues/1288

* w3c/webrtc-stats (+1/-1/💬8)
  1 issues created:
  - Add stat for inputAudioLevel, before the audio filter (by huibk)
    https://github.com/w3c/webrtc-stats/issues/271

  5 issues received 8 new comments:
  - #235 Is keeping stats around a memory problem? (2 by dontcallmedom, jan-ivar)
    https://github.com/w3c/webrtc-stats/issues/235
  - #271 Add stat for inputAudioLevel, before the audio filter (2 by alvestrand, huibk)
    https://github.com/w3c/webrtc-stats/issues/271
  - #255 What is fractionLost for a local incoming media stream? (2 by vr000m)
    https://github.com/w3c/webrtc-stats/issues/255
  - #231 We need "sender" and "receiver" stats, not "track" stats (1 by jan-ivar)
    https://github.com/w3c/webrtc-stats/issues/231
  - #223 jitterBufferDelay vs playoutDelay (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/223

  1 issues closed:
  - jitterBufferDelay vs playoutDelay https://github.com/w3c/webrtc-stats/issues/223



Pull requests
-------------
* w3c/webrtc-pc (+6/-6/💬15)
  6 pull requests submitted:
  - fix ref to webidl  (by fluffy)
    https://github.com/w3c/webrtc-pc/pull/1660
  - Let javascript set different priorities for bitrate and DSCP markings. (by pthatcherg)
    https://github.com/w3c/webrtc-pc/pull/1659
  - Document "test as you commit" policy in CONTRIBUTING.md (by foolip)
    https://github.com/w3c/webrtc-pc/pull/1657
  - RTCSctpTransport: Specify special cases for maxMessageSize (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1656
  - Use [[SctpTransport]] slot in RTCPeerConnection.close (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1655
  - Add steps to create an RTCSctpTransport (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1654

  7 pull requests received 15 new comments:
  - #1657 Document "test as you commit" policy in CONTRIBUTING.md (5 by foolip, adam-be)
    https://github.com/w3c/webrtc-pc/pull/1657
  - #1151 Prepare status of the document for CR publication (3 by dontcallmedom, alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1151
  - #1650 Specify how RTCSctpTransport.maxMessageSize gets its value (2 by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1650
  - #1647 Replace setDirection() with writable direction attribute (2 by alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1647
  - #1632 Adding relativeBitrate parameter to RTCRtpEncodingParameters. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1632
  - #1608 Validate protocol string in IdP operations (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1608
  - #1656 RTCSctpTransport: Specify special cases for maxMessageSize (1 by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1656

  6 pull requests merged:
  - Prepare status of the document for CR publication
    https://github.com/w3c/webrtc-pc/pull/1151
  - Validate protocol string in IdP operations
    https://github.com/w3c/webrtc-pc/pull/1608
  - Use [[SctpTransport]] slot in RTCPeerConnection.close
    https://github.com/w3c/webrtc-pc/pull/1655
  - Add steps to create an RTCSctpTransport
    https://github.com/w3c/webrtc-pc/pull/1654
  - Specify how RTCSctpTransport.maxMessageSize gets its value
    https://github.com/w3c/webrtc-pc/pull/1650
  - Replace setDirection() with writable direction attribute
    https://github.com/w3c/webrtc-pc/pull/1647

* w3c/webrtc-stats (+2/-0/💬2)
  2 pull requests submitted:
  - Pivot from "track" to "sender" and "receiver" stats. (by jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/273
  - Split RTCMediaStreamTrackStats into four dictionaries. (by jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/272

  2 pull requests received 2 new comments:
  - #259 Adding "networkType" field to RTCIceCandidateStats. (1 by garyliu33)
    https://github.com/w3c/webrtc-stats/pull/259
  - #243 Added Guidelines for getStats() results caching (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/243


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats

Received on Tuesday, 7 November 2017 17:00:38 UTC