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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 28 Mar 2017 17:00:29 +0000
To: public-webrtc@w3.org
Message-Id: <E1csuTt-000850-GN@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+4/-5/💬28)
  4 issues created:
  - editorial: Some links to setLocalDescription points to the legacy extensions (by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1095
  - Sender/Receiver.rtcpTransport:  feature at risk? (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1093
  - DTLS failures (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1092
  - When exactly is an SSRC RTCRtpContributingSource object updated? (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1091

  5 issues closed:
  - Section 11.6: Issue 6 https://github.com/w3c/webrtc-pc/issues/1053
  - Handling RTX in RTCRtpCodecCapabilities https://github.com/w3c/webrtc-pc/issues/1079
  - Don't fire events on a closed peer connection https://github.com/w3c/webrtc-pc/issues/1020
  - Candidate from onicecandidate event and addIceCandidate are incompatible https://github.com/w3c/webrtc-pc/issues/1077
  - Align getAlgorithm return value with Web Crypto https://github.com/w3c/webrtc-pc/issues/881

  13 issues received 28 new comments:
  - #1086 Make legacy API optional to implement (6 by foolip, alvestrand, youennf)
    https://github.com/w3c/webrtc-pc/issues/1086
  - #1090 When should RTCRtpContributingSource#audioLevel be null? (4 by taylor-b, alvestrand, aboba)
    https://github.com/w3c/webrtc-pc/issues/1090
  - #1091 When exactly is an SSRC RTCRtpContributingSource object updated? (4 by taylor-b, alvestrand, aboba)
    https://github.com/w3c/webrtc-pc/issues/1091
  - #1092 DTLS failures (3 by taylor-b, aboba, rshpount)
    https://github.com/w3c/webrtc-pc/issues/1092
  - #881 Align getAlgorithm return value with Web Crypto (2 by foolip, aboba)
    https://github.com/w3c/webrtc-pc/issues/881
  - #1077 Candidate from onicecandidate event and addIceCandidate are incompatible (2 by lgrahl, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1077
  - #1093 Sender/Receiver.rtcpTransport:  feature at risk? (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1093
  - #1085 RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout (1 by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1085
  - #1073 Need to specify which members of the encodings in "sendEncodings" are actually used (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1073
  - #1074 Mark Identity as feature at risk? (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1074
  - #1044 Section 12.2.1.1: enum errorDetail definition (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1044
  - #1084 Note At-Risk features at front of document (1 by burnburn)
    https://github.com/w3c/webrtc-pc/issues/1084
  - #1021 get/setParameters does not have a parameter for packetization interval (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1021

* w3c/webrtc-stats (+2/-1/💬17)
  2 issues created:
  - example 8.2: calculating fraction lost vs fractionLost stat (by fippo)
    https://github.com/w3c/webrtc-stats/issues/190
  - Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. (by jan-ivar)
    https://github.com/w3c/webrtc-stats/issues/189

  1 issues closed:
  - Timestamp in the getStats https://github.com/w3c/webrtc-stats/issues/134

  4 issues received 17 new comments:
  - #189 Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. (11 by taylor-b, jesup, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/189
  - #97 Remove availableIncomingBitrate? (3 by vr000m, aboba)
    https://github.com/w3c/webrtc-stats/issues/97
  - #183 Stats report for RTCRtpContributingSource objects (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/183
  - #151 Stat for audio playout delay (1 by aboba)
    https://github.com/w3c/webrtc-stats/issues/151



Pull requests
-------------
* w3c/webrtc-pc (+3/-4/💬3)
  3 pull requests submitted:
  - RTP/RTCP non-mux: feature at risk (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1097
  - Remove last use of 'set of receivers' (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1096
  - Check RTCPeerConnection isClosed slot before running queued tasks (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1094

  4 pull requests merged:
  - Handling RTX in RTCRtpCodecCapabilities
    https://github.com/w3c/webrtc-pc/pull/1082
  - Check RTCPeerConnection isClosed slot before running queued tasks
    https://github.com/w3c/webrtc-pc/pull/1094
  - Make "candidate" non-nullable in addIceCandidate parameter table.
    https://github.com/w3c/webrtc-pc/pull/1088
  - Update call flow in Section 11.6
    https://github.com/w3c/webrtc-pc/pull/1087

  3 pull requests received 3 new comments:
  - #1088 Make "candidate" non-nullable in addIceCandidate parameter table. (1 by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1088
  - #1094 Check RTCPeerConnection isClosed slot before running queued tasks (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1094
  - #1082 Handling RTX in RTCRtpCodecCapabilities (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1082

* w3c/webrtc-stats (+3/-0/💬10)
  3 pull requests submitted:
  - rtt undefined when no RTCP RR (by vr000m)
    https://github.com/w3c/webrtc-stats/pull/188
  - RTCMediaStreamTrackStats members to keep track of audio and video sync. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/187
  - Change log, and make tidy (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/186

  6 pull requests received 10 new comments:
  - #188 rtt undefined when no RTCP RR (3 by vr000m, jesup)
    https://github.com/w3c/webrtc-stats/pull/188
  - #182 Bandwidth estimations again (Issue 97 redux) (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/182
  - #184 Add RTCOutboundRTPStreamStats.totalEncodeTime (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/184
  - #185 Added RTCInboundRTPStreamStats sample counters. (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/185
  - #186 Change log, and make tidy (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/186
  - #187 RTCMediaStreamTrackStats members to keep track of audio and video sync. (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/187


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 28 March 2017 17:00:38 UTC

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