public-webrtc-logs@w3.org from June 2019 by date

Saturday, 29 June 2019

• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub

Friday, 28 June 2019

• Re: [webrtc-pc] Add reject() + make stop() safe. New stopping state affects createOffer only. (#2220) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) guest271314 via GitHub
• [webrtc-pc] Pull Request: Add reject() + make stop() safe. New stopping state affects createOffer only. Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Clarify qualityLimitationReason when limited by multiple reasons (#440) henbos via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) Andreas Pehrson via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) guest271314 via GitHub

Thursday, 27 June 2019

• Re: [webrtc-stats] Clarify qualityLimitationReason when limited by multiple reasons (#440) Varun Singh via GitHub
• Re: [webrtc-stats] Correlating RTCInboundRtpStreamStats with simulcast streams (#393) Varun Singh via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) guest271314 via GitHub
• [webrtc-pc] new commits pushed by jan-ivar Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] new commits pushed by henbos henbos via GitHub
• Closed: [webrtc-pc] Example 11 expects event.streams[0] from streamless tracks (#2217) henbos via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) Andreas Pehrson via GitHub
• Re: [webrtc-pc] Don't rely on event.streams in Example 11. (#2218) stefan hakansson via GitHub
• Re: [webrtc-pc] Read [[LastCreatedOffer/Answer]] at the right time in SLD (#2209) henbos via GitHub
• Re: [mediacapture-record] Define behavior for all bitsPerSecond attributes (#175) Andreas Pehrson via GitHub
• [mediacapture-record] Pull Request: Define behavior for all bitsPerSecond attributes Andreas Pehrson via GitHub
• [mediacapture-record] Pull Request: (Editorial) Make timeslice unsigned everywhere Andreas Pehrson via GitHub
• [mediacapture-main] Pull Request: Reference to WHATWG HTML LS caribouW3 via GitHub

Wednesday, 26 June 2019

• Re: [webrtc-pc] Encode and Decode error event (#2219) Bernard Aboba via GitHub
• Re: [webrtc-pc] Encode and Decode error event (#2219) Bernard Aboba via GitHub
• Re: [webrtc-stats] audioLevel can be removed from "track"|"receiver"|"sender" stats (#391) henbos via GitHub
• [webrtc-stats] Clarify that remote-[in/out]bound-rtp.ssrc refers to the localId's ssrc (#454) henbos via GitHub
• Re: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) karthikbr82 via GitHub
• Re: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) henbos via GitHub
• Re: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) henbos via GitHub
• Closed: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) henbos via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#434) Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) Varun Singh via GitHub
• [webrtc-stats] Add remoteSource to "media-source" (#453) henbos via GitHub
• Re: [webrtc-stats] Add receiving "track" stats to the obsolete section (#452) Varun Singh via GitHub
• [webrtc-stats] Add receiving "track" stats to the obsolete section (#452) henbos via GitHub
• Re: [webrtc-stats] Deprecate remoteSource (#442) Varun Singh via GitHub
• Re: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) Varun Singh via GitHub
• Re: [webrtc-stats] audioLevel can be removed from "track"|"receiver"|"sender" stats (#391) Varun Singh via GitHub
• Re: [webrtc-stats] Move audio levels to "media-source" and "receiver" (#451) Varun Singh via GitHub
• Re: [webrtc-stats] Move audio levels to "media-source" and "receiver" (#451) henbos via GitHub
• Re: [webrtc-stats] Move audio levels to "media-source" and "receiver" (#451) henbos via GitHub
• Re: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) Varun Singh via GitHub
• Re: [webrtc-stats] Replace RTCVideo{Sender, Receiver}Stats::keyFrames{Sent, Received} with RTC{Out, In}boundRtpStreamStats::keyFrames{En, De}coded (#447) Varun Singh via GitHub
• [webrtc-stats] Pull Request: Move audio levels to "media-source" and "receiver" henbos via GitHub
• Re: [webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#450) Varun Singh via GitHub
• Re: [webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#450) Varun Singh via GitHub
• Re: [webrtc-stats] Replace RTCVideo{Sender, Receiver}Stats::keyFrames{Sent, Received} with RTC{Out, In}boundRtpStreamStats::keyFrames{En, De}coded (#447) Rasmus Brandt via GitHub
• [webrtc-pc] Encode and Decode error event (#2219) Johannes Kron via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Rasmus Brandt via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Rasmus Brandt via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) henbos via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) henbos via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Harald Alvestrand via GitHub
• Re: [webrtc-stats] audioLevel can be removed from "track"|"receiver"|"sender" stats (#391) henbos via GitHub
• Re: [webrtc-stats] audioLevel can be removed from "track"|"receiver"|"sender" stats (#391) henbos via GitHub
• Re: [webrtc-stats] Outbound audio levels (#449) henbos via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Rasmus Brandt via GitHub
• [webrtc-stats] Pull Request: Add RTCInboundRtpStreamStats.totalDecodeTime Johannes Kron via GitHub
• [webrtc-stats] Outbound audio levels (#449) henbos via GitHub

Tuesday, 25 June 2019

• Re: [mediacapture-record] What are the default bitrate attribute values? (#169) Jan-Ivar Bruaroey via GitHub
• [webrtc-stats] Audio samples and channels (#448) henbos via GitHub

Monday, 24 June 2019

• [webrtc-pc] Pull Request: Don't rely on event.streams in Example 11. Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Example 11 expects event.streams[0] from streamless tracks (#2217) Jan-Ivar Bruaroey via GitHub
• [mediacapture-output] Normative reference to HTML 5.2 should be changed to HTML LS (#82) Chris Lilley via GitHub
• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub
• Closed: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) henbos via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) henbos via GitHub
• Re: [mediacapture-record] Clarification needed for mime type handling (#170) guest271314 via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) guest271314 via GitHub
• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub
• [webrtc-pc] Pull Request: Use WHATWG references for HTML Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) henbos via GitHub

Sunday, 23 June 2019

• Re: [mediacapture-record] Clarification needed for mime type handling (#170) guest271314 via GitHub
• Re: [mediacapture-record] Input width, height MUST be recorded and playable (#172) guest271314 via GitHub
• [mediacapture-record] Input video track width and height MUST be recorded and playable (#173) guest271314 via GitHub
• [mediacapture-record] Pull Request: Input width, height MUST be recorded and playable guest271314 via GitHub

Saturday, 22 June 2019

• Re: [webrtc-pc] Order of multiple RTCDataChannel.send calls in case of blobs (#2215) Lennart Grahl via GitHub

Friday, 21 June 2019

• Re: [webrtc-pc] Order of multiple RTCDataChannel.send calls in case of blobs (#2215) youennf via GitHub
• [webrtc-pc] Order of multiple RTCDataChannel.send calls in case of blobs (#2215) youennf via GitHub
• Re: [mediacapture-screen-share] Screenshot (still image) capability (#107) guest271314 via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) guest271314 via GitHub

Thursday, 20 June 2019

• Re: [mediacapture-screen-share] Screenshot (still image) capability (#107) guest271314 via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) guest271314 via GitHub
• Re: [mediacapture-screen-share] Should the permissions UI be the first captured frame? (#108) guest271314 via GitHub
• Re: [mediacapture-record] Do not discard data in case track set change (#162) Andreas Pehrson via GitHub
• Re: [mediacapture-record] Clarify MediaRecorder's mimeType handling (#171) Andreas Pehrson via GitHub
• [mediacapture-record] Pull Request: Clarify MediaRecorder's mimeType handling Andreas Pehrson via GitHub
• Re: [mediacapture-screen-share] Screenshot (still image) capability (#107) Armando \(armax00\) via GitHub
• Re: [mediacapture-screen-share] Should the permissions UI be the first captured frame? (#108) henbos via GitHub
• Closed: [mediacapture-screen-share] Ask for user gesture to call getDisplayMedia (#104) Bernard Aboba via GitHub
• Re: [mediacapture-screen-share] Screenshot (still image) capability (#107) henbos via GitHub
• Re: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) henbos via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] RTCIceTransport role before start() called (#2210) Bernard Aboba via GitHub
• [webrtc-pc] Specify when RTCIceRole is updated (#2214) henbos via GitHub
• Re: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) henbos via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Rasmus Brandt via GitHub
• Re: [webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447) Rasmus Brandt via GitHub
• Re: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439) Harald Alvestrand via GitHub
• [webrtc-stats] Pull Request: WIP: Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded Rasmus Brandt via GitHub
• Closed: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446) henbos via GitHub
• Re: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446) henbos via GitHub
• Re: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446) Jan-Ivar Bruaroey via GitHub

Wednesday, 19 June 2019

• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub
• Re: [mediacapture-record] What are the default bitrate attribute values? (#169) Andreas Pehrson via GitHub
• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub
• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub

Tuesday, 18 June 2019

• Re: [mediacapture-record] Clarification needed for mime type handling (#170) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-record] Why is start 5.3 in the specification? (#168) guest271314 via GitHub
• Re: [mediacapture-record] Do not discard data in case track set change (#162) Andreas Pehrson via GitHub

Monday, 17 June 2019

• [mediacapture-record] Clarification needed for mime type handling (#170) Andreas Pehrson via GitHub
• Re: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446) henbos via GitHub
• Re: [webrtc-stats] Detecting Video Playback glitches (#443) ilnik via GitHub
• [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213) guest271314 via GitHub

Saturday, 15 June 2019

• Re: [webrtc-pc] Add support for WebRTC Data Channel in Workers (#230) Nuri via GitHub

Friday, 14 June 2019

• [webrtc-pc] Pull Request: Add setRemoteDescription(description, {rollback: true}) option. Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446) Varun Singh via GitHub
• Re: [webrtc-stats] RTCInboundRtpStreamStats.fractionLost (#446) henbos via GitHub
• [webrtc-stats] RTCInboundRtpStreamStats.fractionLost (#446) henbos via GitHub
• Re: [webrtc-stats] Type stunserverconnection should be stun-server-connection (#444) Varun Singh via GitHub
• Re: [webrtc-stats] Detecting glitches (#443) Varun Singh via GitHub
• Re: [webrtc-stats] Clarify qualityLimitationReason when limited by multiple reasons (#440) henbos via GitHub
• [webrtc-pc] Pull Request: Define an "unknown" value for RTCIceRole Harald Alvestrand via GitHub
• Re: [webrtc-stats] Clarify qualityLimitationReason when limited by multiple reasons (#440) Varun Singh via GitHub
• [webrtc-pc] RTCIceTransport role before start() called (#2210) Harald Alvestrand via GitHub
• Re: [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) henbos via GitHub
• Re: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) henbos via GitHub
• Re: [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) karthikbr82 via GitHub
• Re: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) Varun Singh via GitHub
• Re: [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) Varun Singh via GitHub
• Re: [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) henbos via GitHub
• Re: [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) Varun Singh via GitHub
• Re: [webrtc-pc] Add a security/privacy note about remote SDP (#2193) guest271314 via GitHub
• Re: [webrtc-pc] Add a security/privacy note about remote SDP (#2193) guest271314 via GitHub
• Re: [webrtc-pc] Add a security/privacy note about remote SDP (#2193) guest271314 via GitHub
• [webrtc-pc] Pull Request: Read [[LastCreatedOffer/Answer]] at the right time in SLD Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] SLD without {sdp} reads [[LastCreatedOffer/Answer]] at the wrong time (#2208) Jan-Ivar Bruaroey via GitHub

Thursday, 13 June 2019

• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) henbos via GitHub
• Closed: [webrtc-pc] Add a security/privacy note about remote SDP (#2193) henbos via GitHub
• [webrtc-pc] new commits pushed by henbos henbos via GitHub
• Re: [webrtc-pc] Add security note about remote SDP (#2201) henbos via GitHub
• [mediacapture-main] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [mediacapture-main] Clarify that enumerateDevices must not expose devices that a given context cannot use through getUserMedia (#549) Harald Alvestrand via GitHub
• [mediacapture-main] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] Remove [WEBIDL-1] reference in favor of [WEBIDL] (#2207) henbos via GitHub
• [webrtc-pc] new commits pushed by henbos henbos via GitHub
• Closed: [webrtc-pc] Update WebIDL reference in Terminology (#2091) henbos via GitHub
• Re: [webrtc-pc] Make DOMException references cite [WEBIDL] (#2200) henbos via GitHub
• Re: [webrtc-pc] Make DOMException references cite [WEBIDL] (#2200) Dominique Hazael-Massieux via GitHub
• Re: [mediacapture-record] What are the default bitrate attribute values? (#169) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Make DOMException references cite [WEBIDL] (#2200) henbos via GitHub
• Re: [webrtc-pc] Add security note about remote SDP (#2201) henbos via GitHub
• Re: [webrtc-stats] Type stunserverconnection should be stun-server-connection (#444) Philipp Hancke via GitHub
• [webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445) henbos via GitHub
• [webrtc-stats] Type stunserverconnection should be stun-server-connection (#444) henbos via GitHub
• Re: [mediacapture-record] What are the default bitrate attribute values? (#169) Harald Alvestrand via GitHub

Wednesday, 12 June 2019

• Re: [webrtc-stats] Correlating RTCInboundRtpStreamStats with simulcast streams (#393) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Correlating RTCInboundRtpStreamStats with simulcast streams (#393) Jan-Ivar Bruaroey via GitHub
• [mediacapture-record] What are the default bitrate attribute values? (#169) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Detecting glitches (#443) ilnik via GitHub
• Re: [webrtc-stats] Detecting glitches (#443) henbos via GitHub
• Re: [webrtc-stats] Detecting glitches (#443) ilnik via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Harald Alvestrand via GitHub

Tuesday, 11 June 2019

• Re: [mediacapture-main] Interpret the empty string as if the constraint is not specified (#595) youennf via GitHub
• Re: [mediacapture-main] Interpret the empty string as if the constraint is not specified (#595) Harald Alvestrand via GitHub
• [webrtc-stats] Detecting glitches (#443) henbos via GitHub
• Re: [mediacapture-main] MediaTrackSettings lacks support of channel layout (#605) Armando \(armax00\) via GitHub
• Re: [mediacapture-main] MediaTrackSettings lacks support of channel layout (#605) Armando \(armax00\) via GitHub
• Closed: [mediacapture-main] MediaTrackSettings lacks support of channel layout (#605) Armando \(armax00\) via GitHub
• [mediacapture-main] MediaTrackSettings lacks support of channel layout (#605) henrikand via GitHub
• Re: [webrtc-pc] RTCDataChannel.bufferedAmount value (#1823) Iñaki Baz Castillo via GitHub
• Re: [mediacapture-fromelement] Don't run the steps for MediaStreamTrack.stop() when ending a track (#81) Martin Thomson via GitHub
• Closed: [mediacapture-fromelement] Should tracks captured from a media element fire "ended" when ending? (#77) Martin Thomson via GitHub

Monday, 10 June 2019

• Re: [mediacapture-main] When an HTMLMediaElement has srcObject set to a MediaStream where the MediaStreamTrack is muted the HTMLMediaElement muted attribute should be set to muted to reflect muted input and muted output (#583) guest271314 via GitHub

Sunday, 9 June 2019

• Re: [mediacapture-main] Specify a way for WebDriver to add/remove/setup capture devices (#554) guest271314 via GitHub

Saturday, 8 June 2019

• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) guest271314 via GitHub
• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) guest271314 via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Lennart Grahl via GitHub
• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub

Friday, 7 June 2019

• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) youennf via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) youennf via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) youennf via GitHub
• Re: [mediacapture-main] Clarify language around element attributes after stream removal. (#600) guidou via GitHub
• Re: [mediacapture-record] Why is start 5.3 in the specification? (#168) guest271314 via GitHub

Thursday, 6 June 2019

• [mediacapture-main] oldList definition in enumerateDevices is unclear (#604) youennf via GitHub
• Re: [mediacapture-record] Why is start 5.3 in the specification? (#168) guest271314 via GitHub
• Re: [mediacapture-record] Why is start 5.3 in the specification? (#168) Harald Alvestrand via GitHub
• Re: [mediacapture-main] Should deviceId be partitioned by top level origin and document origin (#598) Harald Alvestrand via GitHub
• [mediacapture-main] Pull Request: Filter enumerateDevices by feature policy. Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Pull Request: Fix outdated information in mandatory stats list. Jan-Ivar Bruaroey via GitHub
• [webrtc-stats] Pull Request: Deprecate remoteSource Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399) Jan-Ivar Bruaroey via GitHub
• [mediacapture-record] Why is start 5.3 in the specification? (#168) guest271314 via GitHub
• Re: [mediacapture-main] Should deviceId be partitioned by top level origin and document origin (#598) Harald Alvestrand via GitHub
• Closed: [mediacapture-main] Echo Cancellation (#597) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-main] Echo Cancellation (#597) Jan-Ivar Bruaroey via GitHub
• [mediacapture-main] new commits pushed by aboba Bernard Aboba via GitHub
• [mediacapture-main] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [mediacapture-main] Clarification needed on HTMLMediaElement attributes that carry over (#599) Harald Alvestrand via GitHub
• Re: [mediacapture-main] Clarify language around element attributes after stream removal. (#600) Harald Alvestrand via GitHub
• Re: [mediacapture-main] Recommend UUID for deviceId and groupId (#591) Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Setting active attribute to false triggers BYE? (#2202) Bernard Aboba via GitHub
• Closed: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157) Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Make DOMException references cite [WEBIDL] (#2200) Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: Specify that "id" is ignored when negotiated=false. Harald Alvestrand via GitHub
• Closed: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub
• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: Clarify that setting active to false does not send RTCP BYE Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Ensure icecandidate events happen after SLD(offer). Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Setting active attribute to false triggers BYE? (#2202) Bernard Aboba via GitHub

Wednesday, 5 June 2019

• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) guest271314 via GitHub
• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) Andreas Pehrson via GitHub
• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) guest271314 via GitHub
• [webrtc-pc] Pull Request: Add security note about remote SDP henbos via GitHub
• [webrtc-pc] Pull Request: Make DOMException references cite [WEBIDL] henbos via GitHub
• Re: [webrtc-pc] Define which constraints are applicable in WebRTC (#2188) henbos via GitHub
• Re: [webrtc-pc] Define which constraints are applicable in WebRTC (#2188) henbos via GitHub
• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) Andreas Pehrson via GitHub
• [mediacapture-image] Extend getPhotoSettings() to include additional EXIF data (#209) Joe Palmer via GitHub

Tuesday, 4 June 2019

• Re: [mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80) guest271314 via GitHub
• Closed: [mediacapture-main] MediaStream in HTMLMediaElement: Does it have a first frame? (#555) Andreas Pehrson via GitHub
• Re: [mediacapture-main] MediaStream in HTMLMediaElement: Does it have a first frame? (#555) Andreas Pehrson via GitHub
• Re: [mediacapture-main] Clarify language around element attributes after stream removal. (#600) Andreas Pehrson via GitHub

Monday, 3 June 2019

• Re: [mediacapture-main] Clarify language around element attributes after stream removal. (#600) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-main] Clarification needed on HTMLMediaElement attributes that carry over (#599) Jan-Ivar Bruaroey via GitHub
• [mediacapture-main] Pull Request: Create FUNDING.yml W-wA via GitHub
• [mediacapture-main] Pull Request: Add definition to track-ending algorithm Andreas Pehrson via GitHub
• Re: [mediacapture-main] Clarification needed on HTMLMediaElement attributes that carry over (#599) Andreas Pehrson via GitHub
• Re: [webrtc-pc] Clarify/strengthen timing invariants of candidates not firing before SLD(offer) (#2199) Philipp Hancke via GitHub
• Re: [webrtc-pc] Clarify/strengthen timing invariants of candidates not firing before SLD(offer) (#2199) Harald Alvestrand via GitHub

Sunday, 2 June 2019

• Re: [mediacapture-screen-share] Remove See Work Flow and bad links (#109) Dominique Hazael-Massieux via GitHub
• [mediacapture-screen-share] Pull Request: Remove See Work Flow and bad links guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub

Last message date: Saturday, 29 June 2019 00:05:21 UTC