- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Tue, 14 Mar 2017 17:00:42 +0000
- To: public-webrtc@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