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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 14 Mar 2017 17:00:42 +0000
To: public-webrtc@w3.org
Message-Id: <E1cnpoQ-0006ks-4L@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+6/-12/💬22)
  6 issues created:
  - Mark Identity as feature at risk? (by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/1074
  - Need to specify which members of the encodings in "sendEncodings" are actually used (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1073
  - Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1065
  - Attaching the same track to multiple RTCRtpSenders: spec contradiction. (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1064
  - Section 10.3: Freeing resources for incoming stream (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1063
  - Section 10.3:  (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1062

  12 issues closed:
  - Section 5.6: Generation of candidates https://github.com/w3c/webrtc-pc/issues/1050
  - Attaching the same track to multiple RTCRtpSenders: spec contradiction. https://github.com/w3c/webrtc-pc/issues/1064
  - Codecs supporting multiple clock rates/packetization-mode https://github.com/w3c/webrtc-pc/issues/1040
  - Section 13: Connecting event https://github.com/w3c/webrtc-pc/issues/1043
  - Section 10.3: mute signal  https://github.com/w3c/webrtc-pc/issues/1051
  - Section 10.3: Issue 5 https://github.com/w3c/webrtc-pc/issues/1052
  - "Throw a FooError" steps not written in a consistent manner https://github.com/w3c/webrtc-pc/issues/845
  - Range checking needed for iceCandidatePoolSize https://github.com/w3c/webrtc-pc/issues/1048
  - Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP https://github.com/w3c/webrtc-pc/issues/1049
  - RTCDataChannelInit.id should use [EnforceRange] https://github.com/w3c/webrtc-pc/issues/1046
  - sdpFmtpLine isn't very convenient https://github.com/w3c/webrtc-pc/issues/1022
  - Section 10.3:  https://github.com/w3c/webrtc-pc/issues/1062

  8 issues received 22 new comments:
  - #1063 Section 10.3: Freeing resources for incoming stream (6 by Pehrsons, aboba, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1063
  - #1053 Section 11.6: Issue 6 (5 by fippo, aboba, alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1053
  - #1051 Section 10.3: mute signal  (4 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/1051
  - #1052 Section 10.3: Issue 5 (2 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/1052
  - #1065 Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy (2 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/1065
  - #1064 Attaching the same track to multiple RTCRtpSenders: spec contradiction. (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1064
  - #881 Align getAlgorithm return value with Web Crypto (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/881
  - #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 (+0/-10/💬25)
  10 issues closed:
  - RTCPeerConnection.getStats: What to do with 'selector' argument? https://github.com/w3c/webrtc-stats/issues/116
  - Make sure timing and responsibility for stat object creation is clear https://github.com/w3c/webrtc-stats/issues/120
  - Consider making (aggregate) stats more accessible https://github.com/w3c/webrtc-stats/issues/119
  - Expose protocol used to communicate with TURN server https://github.com/w3c/webrtc-stats/issues/83
  - Consider putting roundTripTime on RTCInboundStreamStats https://github.com/w3c/webrtc-stats/issues/78
  - There is no ICE "cancelled" state https://github.com/w3c/webrtc-stats/issues/66
  - Unclear if the request encompasses consent checks or not https://github.com/w3c/webrtc-stats/issues/171
  - RTCDataChannelStats should reference RTCTransportStats https://github.com/w3c/webrtc-stats/issues/140
  - Not clear how to differentiate between received connectivity checks and consent requests https://github.com/w3c/webrtc-stats/issues/96
  - Need descriptions for `label`, `protocol` and `state` members of `RTCDataChannelStats` https://github.com/w3c/webrtc-stats/issues/111

  14 issues received 25 new comments:
  - #163 RTCIceCandidatePairStats.writable/readable: redundant? (4 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/163
  - #180 RoundTripTime not defined when the underlying stream can't calculate it (4 by henbos, vr000m, jesup, jan-ivar)
    https://github.com/w3c/webrtc-stats/issues/180
  - #160 Stat for adaptation reason (3 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/160
  - #171 Unclear if the request encompasses consent checks or not (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/171
  - #178 Packets or frames discarded on send? (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/178
  - #150 Stat for how much time it takes to encode video (2 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/150
  - #161 Definitions from MSE need re-targeting (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/161
  - #144 Stat for how many adaptation changes occur for a video track (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/144
  - #152 Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/152
  - #116 RTCPeerConnection.getStats: What to do with 'selector' argument? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/116
  - #119 Consider making (aggregate) stats more accessible (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/119
  - #120 Make sure timing and responsibility for stat object creation is clear (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/120
  - #154 Stat for retransmitted bytes (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/154
  - #153 Stat for likelihood of echo (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/153



Pull requests
-------------
* w3c/webrtc-pc (+7/-12/💬31)
  7 pull requests submitted:
  - Remove Section 11.6 note (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1072
  - Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1071
  - Update Call Flow Browser to Browser (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1070
  - Freeing resources for incoming stream (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1069
  - Remove Section 12.2.1.1 errorDetailEnum (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1068
  - Mark getAlgorithm method at risk (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1067
  - RtpSender and track (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1066

  12 pull requests merged:
  - Add stats selection algorithm based on sender or receiver of selector.
    https://github.com/w3c/webrtc-pc/pull/1030
  - Switching to new, consistent terminology when talking about exceptions.
    https://github.com/w3c/webrtc-pc/pull/1056
  - RtpSender and track
    https://github.com/w3c/webrtc-pc/pull/1066
  - Remove connecting event from Event summary
    https://github.com/w3c/webrtc-pc/pull/1061
  - ended event fired on the track
    https://github.com/w3c/webrtc-pc/pull/1060
  - mute signal
    https://github.com/w3c/webrtc-pc/pull/1059
  - Generation of candidates
    https://github.com/w3c/webrtc-pc/pull/1058
  - Add clockrate, channels, sdpFmtpLine to codec capability
    https://github.com/w3c/webrtc-pc/pull/1057
  - Changing iceCandidatePoolSize to an octet and adding EnforceRange.
    https://github.com/w3c/webrtc-pc/pull/1055
  - Throw InvalidModificationError if changing pool size after SLD.
    https://github.com/w3c/webrtc-pc/pull/1054
  - Adding "[EnforceRange]" to RTCDataChannelInit.id.
    https://github.com/w3c/webrtc-pc/pull/1047
  - Clarifying exactly what "sdpFmtpLine" represents.
    https://github.com/w3c/webrtc-pc/pull/1045

  17 pull requests received 31 new comments:
  - #1033 "Hybrid" OAuth solution. (5 by taylor-b, alvestrand, misi, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1033
  - #1055 Changing iceCandidatePoolSize to an octet and adding EnforceRange. (4 by jan-ivar, aboba)
    https://github.com/w3c/webrtc-pc/pull/1055
  - #1054 Throw InvalidModificationError if changing pool size after SLD. (3 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/pull/1054
  - #1071 Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. (3 by taylor-b, aboba, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1071
  - #1030 Add stats selection algorithm based on sender or receiver of selector. (2 by jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1030
  - #1056 Switching to new, consistent terminology when talking about exceptions. (2 by dontcallmedom, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1056
  - #1066 RtpSender and track (2 by alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1066
  - #1047 Adding "[EnforceRange]" to RTCDataChannelInit.id. (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1047
  - #1058 Generation of candidates (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1058
  - #1059 mute signal (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1059
  - #1060 ended event fired on the track (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1060
  - #1061 Remove connecting event from Event summary (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1061
  - #1067 Mark getAlgorithm method at risk (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1067
  - #1068 Remove Section 12.2.1.1 errorDetailEnum (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1068
  - #1069 Freeing resources for incoming stream (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1069
  - #1070 Update Call Flow Browser to Browser (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1070
  - #1072 Remove Section 11.6 note (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1072

* w3c/webrtc-stats (+1/-4/💬5)
  1 pull requests submitted:
  - Add section on obsoleted stats. (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/181

  4 pull requests merged:
  - Adding remoteTimestamp to RTCRtpStreamStats.
    https://github.com/w3c/webrtc-stats/pull/164
  - Add link from datachannel to transport
    https://github.com/w3c/webrtc-stats/pull/176
  - Remove separation of received consent and connectivity requests.
    https://github.com/w3c/webrtc-stats/pull/165
  - Adds definitions for RTCDataChannelStats members.
    https://github.com/w3c/webrtc-stats/pull/179

  3 pull requests received 5 new comments:
  - #164 Adding remoteTimestamp to RTCRtpStreamStats. (2 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/164
  - #181 Add section on obsoleted stats. (2 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/181
  - #176 Add link from datachannel to transport (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/176


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 14 March 2017 17:00:50 UTC

This archive was generated by hypermail 2.3.1 : Monday, 23 October 2017 15:19:50 UTC