Weekly github digest (WebRTC WG specifications)

Issues
------
* w3c/webrtc-pc (+6/-11/💬79)
  6 issues created:
  - Possibly racy replaceTrack() (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1728
  - WebRTC bypass CSP connect-src policies (by murillo128)
    https://github.com/w3c/webrtc-pc/issues/1727
  - canTrickleIceCandidiates question (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1721
  - "a=msid" line should contain sender/receiver IDs, not track IDs (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1718
  - Should we require a reference from RTPSender to RTCPeerConnection? (by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1712
  - enqueue an operation: is executing async? (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1711

  29 issues received 79 new comments:
  - #1718 "a=msid" line should contain sender/receiver IDs, not track IDs (19 by stefhak, henbos, ibc, fippo, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1718
  - #1662 addTransceiver woes (10 by taylor-b, fippo, jan-ivar, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1662
  - #1562 How to handle removing and re-adding remote streams/tracks - possible ID collisions? (6 by henbos, taylor-b, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1562
  - #1721 canTrickleIceCandidiates question (6 by fippo, taylor-b, cdh4u, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1721
  - #1709 example 14: render before verifying the remote fingerprint? (4 by nils-ohlmeier, aboba, alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1709
  - #1711 enqueue an operation: is executing async? (3 by fippo, jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1711
  - #1712 Should we require a reference from RTPSender to RTCPeerConnection? (3 by fippo, aboba, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1712
  - #1420 At risk text in wrong location (2 by aboba, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1420
  - #1677 replaceTrack and removeTrack: Synchronous? (2 by aboba, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1677
  - #1690 RTCRtpContributingSource.timestamp needs a clearer definition (2 by jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1690
  - #1694 RTCCertificate Interface should (or should not) be backed up. (2 by steely-glint, aboba)
    https://github.com/w3c/webrtc-pc/issues/1694
  - #1697 replaceTrack: Never negotiate when replacing an ended track? (2 by adam-be, aboba)
    https://github.com/w3c/webrtc-pc/issues/1697
  - #1240 editorial: media api introduction (2 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1240
  - #1112 Terminology around "setting" attributes may be incorrect (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1112
  - #1689  Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1689
  - #1691 When to fire events triggered by setRemoteDescription. (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1691
  - #1699 Data channel closing procedure (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1699
  - #1700 Maximum message size slightly incorrect (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1700
  - #1445 Standard should provide some guidance for when an application should perform an ICE restart (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1445
  - #1705 specify legacy onaddstream? (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1705
  - #1707 onmute then onunmute can fire before negotiation completes (1 by docfaraday)
    https://github.com/w3c/webrtc-pc/issues/1707
  - #1708 Example 14 never sends any media (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1708
  - #1454 Using setConfiguration() to add application certificates to an RTCPeerConnection post-construction? (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1454
  - #1342 Consider using generic algorithm (1 by stefhak)
    https://github.com/w3c/webrtc-pc/issues/1342
  - #1728 Possibly racy replaceTrack() (1 by henbos)
    https://github.com/w3c/webrtc-pc/issues/1728
  - #1345 Make promise rejection/enqueing consistent (1 by stefhak)
    https://github.com/w3c/webrtc-pc/issues/1345
  - #1625 RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1625
  - #230 Add support for WebRTC Data Channel in Workers (1 by lgrahl)
    https://github.com/w3c/webrtc-pc/issues/230
  - #1651 JSEP references are out dated (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1651

  11 issues closed:
  - RTCIceCandidate and candidate string parsing requirements https://github.com/w3c/webrtc-pc/issues/1224
  - Maximum message size slightly incorrect https://github.com/w3c/webrtc-pc/issues/1700
  - replaceTrack: Never negotiate when replacing an ended track? https://github.com/w3c/webrtc-pc/issues/1697
  - enqueue an operation: is executing async? https://github.com/w3c/webrtc-pc/issues/1711
  - addTrack can reuse a sender whose transceiver is stopped https://github.com/w3c/webrtc-pc/issues/1494
  - At risk text in wrong location https://github.com/w3c/webrtc-pc/issues/1420
  - How to handle removing and re-adding remote streams/tracks - possible ID collisions? https://github.com/w3c/webrtc-pc/issues/1562
  - Ordering of stream "addtrack"/"removetrack" events vs. "track" event https://github.com/w3c/webrtc-pc/issues/1599
  - JSEP references are out dated https://github.com/w3c/webrtc-pc/issues/1651
  - replaceTrack and removeTrack: Synchronous? https://github.com/w3c/webrtc-pc/issues/1677
  - Consider using generic algorithm https://github.com/w3c/webrtc-pc/issues/1342

* w3c/webrtc-stats (+1/-2/💬13)
  1 issues created:
  - WiFi Stats. (by karthikbr82)
    https://github.com/w3c/webrtc-stats/issues/289

  6 issues received 13 new comments:
  - #238 Add stat to reflect the redundancy of FEC/RED data (5 by henbos, minyuel, vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/238
  - #289 WiFi Stats. (4 by vr000m, alvestrand, aboba)
    https://github.com/w3c/webrtc-stats/issues/289
  - #240 Stats for Audio network adaptation (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/240
  - #241 Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/241
  - #222 Audio/Video sync follow-up (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/222
  - #287 packetsDuplicated is not reported in an RTCP Report (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/287

  2 issues closed:
  - Stats for Audio network adaptation https://github.com/w3c/webrtc-stats/issues/240
  - Audio/Video sync follow-up https://github.com/w3c/webrtc-stats/issues/222

* w3c/webrtc-charter (+3/-0/💬0)
  3 issues created:
  - Add language on enabling Workers to use WebRTC constructs (by alvestrand)
    https://github.com/w3c/webrtc-charter/issues/20
  - Add language on "Using the Stream API to access data channels" (by alvestrand)
    https://github.com/w3c/webrtc-charter/issues/19
  - Revise IETF liaison text (by aboba)
    https://github.com/w3c/webrtc-charter/issues/18



Pull requests
-------------
* w3c/webrtc-pc (+13/-4/💬11)
  13 pull requests submitted:
  - Define JSEP terms in terminology section (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1726
  - replaceTrack "negotiation needed" clarification (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1725
  - Add detail to RTCDtlsTransportState descriptions (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1724
  - Whether KeyingMaterial internal slot can be stored and retrieved (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1723
  - Underlying data transport explanation (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1722
  - Defer SRD add/remove tracks until right before firing track events. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1720
  - Define pc.getTransceivers()) et al to be in insertion order. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1719
  - addTrack cannot reuse a sender whose transceiver is stopped (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1717
  - Remove reference from RTPSender to RTCPeerConnection (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1716
  - Move operations queue definition paragraph to right section (editorial) (by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1715
  - Validation of reordered readonly parameters in setParameters (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1714
  - Maximum message size slightly incorrect (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1713
  - replaceTrack: Never negotiate when replacing an ended track? (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1710

  5 pull requests received 11 new comments:
  - #1698 RTPRtcSender: provided properties to get related MediaStreamTrackId and media type (7 by mparis, dontcallmedom, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1698
  - #1696 replaceTrack: Clarify how the UA determines if negotiation is needed (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1696
  - #1713 Maximum message size slightly incorrect (1 by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1713
  - #1722 Underlying data transport explanation (1 by lgrahl)
    https://github.com/w3c/webrtc-pc/pull/1722
  - #1723 Whether KeyingMaterial internal slot can be stored and retrieved (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1723

  4 pull requests merged:
  - Check that transceiver is not stopped before using.
    https://github.com/w3c/webrtc-pc/pull/1701
  - replaceTrack: Never negotiate when replacing an ended track?
    https://github.com/w3c/webrtc-pc/pull/1710
  - Maximum message size slightly incorrect
    https://github.com/w3c/webrtc-pc/pull/1713
  - Move operations queue definition paragraph to right section (editorial)
    https://github.com/w3c/webrtc-pc/pull/1715

* w3c/webrtc-stats (+2/-0/💬2)
  2 pull requests submitted:
  - Introduction to RTP stream statistics (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/291
  - Adds per-DSCP packet counters to RTP streams (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/290

  2 pull requests received 2 new comments:
  - #290 Adds per-DSCP packet counters to RTP streams (1 by vr000m)
    https://github.com/w3c/webrtc-stats/pull/290
  - #273 Pivot from "track" to "sender" and "receiver" stats. (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/273


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

Received on Tuesday, 9 January 2018 17:01:20 UTC