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

Tuesday, 30 April 2019

• Re: [mediacapture-main] Clarify that enumerateDevices must not expose devices that a given context cannot use through getUserMedia (#549) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Says to count datachannel label & protocol in bytes, but they're USVString (#2191) Lennart Grahl via GitHub
• Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106) Martin Thomson via GitHub
• Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106) youennf via GitHub
• [webrtc-pc] Says to count datachannel label & protocol in bytes, but they're USVString (#2191) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106) Martin Thomson via GitHub

Monday, 29 April 2019

• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) youennf via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106) youennf via GitHub
• [mediacapture-screen-share] Pull Request: Add user activation check when calling getDisplayMedia youennf via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) youennf via GitHub
• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) Jan-Ivar Bruaroey via GitHub
• [webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#434) henbos via GitHub
• [mediacapture-main] Pull Request: #577 make echo cancellation scope more explicit Jai Gupta via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub
• Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) Martin Thomson via GitHub

Sunday, 28 April 2019

• [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105) guest271314 via GitHub

Saturday, 27 April 2019

• [mediacapture-fromelement] MediaStreamTrack does not dispatch "ended" event at Chromium when src of HTMLMediaElement is changed, Firefox does dispatch "ended" event; which implementation is correct? (#78) guest271314 via GitHub
• Re: [mediacapture-main] Travis is failing git checkout with "reference is not a tree" (#581) Harald Alvestrand via GitHub

Friday, 26 April 2019

• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) henbos via GitHub
• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) henbos via GitHub
• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) Chen Xing via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Johannes Kron via GitHub
• [webrtc-pc] Pull Request: Clarify `RTCRtpContributingSource` and `RTCRtpSynchronizationSource`. Chen Xing via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• Re: [mediacapture-main] echo cancellation scope (#577) henbos via GitHub
• Re: [mediacapture-main] echo cancellation scope (#577) Jai Gupta via GitHub
• Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433) henbos via GitHub
• Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) Johannes Kron via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• [mediacapture-main] Travis is failing git checkout with "reference is not a tree" (#581) henbos via GitHub
• [mediacapture-main] new commits pushed by henbos henbos via GitHub
• Closed: [mediacapture-main] Remove overconstrained event (Why do we have overconstrained event?) (#573) henbos via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (#576) henbos via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (#576) Armando \(armax00\) via GitHub
• [mediacapture-main] Pull Request: Just testing... ignore this henbos via GitHub
• [mediacapture-main] Pull Request: Remove "onoverconstrained" re-upload Armando \(armax00\) via GitHub

Thursday, 25 April 2019

• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) guest271314 via GitHub
• Re: [webrtc-pc] Define time order of results from getContributingSources() et al (#2189) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Define time order of results from getContributingSources() et al (#2189) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (#576) Armando \(armax00\) via GitHub
• Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) na-g via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Lennart Grahl via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) na-g via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Dominique Hazael-Massieux via GitHub
• [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
• Re: [mediacapture-main] echo cancellation scope (#577) Harald Alvestrand via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (#576) Dominique Hazael-Massieux via GitHub
• Closed: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) Bernard Aboba via GitHub
• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) Bernard Aboba via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (#576) Harald Alvestrand via GitHub
• Closed: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) youennf via GitHub
• Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) youennf via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] Clarify update negotiation needed (#2159) Bernard Aboba via GitHub
• Closed: [webrtc-pc] Mark getDefaultIceServers() as feature at risk. (#2185) Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] Add rtpTimestamp to RTCRtpContributingSource (#2178) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] Add rtpTimestamp to RTCRtpContributingSource (#2178) henbos via GitHub
• Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• [webrtc-pc] Pull Request: Define which constraints are applicable in WebRTC henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• Re: [webrtc-pc] Constrainable properties on remote tracks are under-specified (#2121) henbos via GitHub
• [mediacapture-main] Add note on applicability of constraints to tracks in other specs (#578) henbos via GitHub
• Re: [mediacapture-main] echo cancellation scope (#577) henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• [webrtc-pc] Pull Request: Clarify a step in "update negotiation needed" henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) henbos via GitHub
• Re: [webrtc-pc] Simulcast stats (#2116) henbos via GitHub
• Closed: [webrtc-pc] Simulcast stats (#2116) henbos via GitHub
• Closed: [webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103) henbos via GitHub
• Re: [webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103) henbos via GitHub
• [webrtc-pc] Pull Request: Adds a note clarifying removeTrack() henbos via GitHub
• Re: [webrtc-pc] how to use datachannel bufferedamountlow threshold callback? (#1979) Lennart Grahl via GitHub
• Re: [webrtc-pc] Should we remove getDefaultIceServers? (#2023) henbos via GitHub
• [webrtc-pc] Mark getDefaultIceServers() as feature at risk. (#2185) henbos via GitHub
• Re: [webrtc-pc] Should we remove getDefaultIceServers? (#2023) henbos via GitHub
• [webrtc-pc] Pull Request: Mark getDefaultIceServers() as feature at risk henbos via GitHub
• Closed: [webrtc-pc] how to use datachannel bufferedamountlow threshold callback? (#1979) henbos via GitHub
• Re: [webrtc-pc] how to use datachannel bufferedamountlow threshold callback? (#1979) henbos via GitHub
• Re: [webrtc-pc] RTCRtpReceiver.getStreams() and RTCRtpSender.getStreams() (#1921) henbos via GitHub
• Closed: [webrtc-pc] RTCRtpReceiver.getStreams() and RTCRtpSender.getStreams() (#1921) henbos via GitHub
• Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183) henbos via GitHub
• Closed: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183) henbos via GitHub
• Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183) henbos via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) Lennart Grahl via GitHub
• Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183) henbos via GitHub
• Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152) henbos via GitHub
• [webrtc-pc] There are redundant steps in setCodecPreferences (#2183) henbos via GitHub
• Re: [webrtc-pc] Make sure prflx remote candidates do not leak information when exposed by getSelectedCandidatePair (#2182) youennf via GitHub
• Re: [webrtc-pc] Cap expires to 365 days at most (#2181) youennf via GitHub
• [webrtc-pc] Pull Request: Make sure prflx remote candidates do not leak information when exposed by getSelectedCandidatePair youennf via GitHub
• [webrtc-pc] Pull Request: Cap expires to 365 days at most youennf via GitHub

Wednesday, 24 April 2019

• Re: [webrtc-pc] RTCIceTransport.getRemoteCandidates() does not return prflx candidates (#2124) youennf via GitHub
• Closed: [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Clarify "multiple sources of media stitched together" Bernard Aboba via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) Lennart Grahl via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) Lennart Grahl via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) Lennart Grahl via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#2179) Lennart Grahl via GitHub
• [webrtc-pc] Allow to import existing certificate (#2179) ibsusu via GitHub
• Re: [webrtc-pc] Allow to import existing certificate (#1853) ibsusu via GitHub
• [webrtc-pc] Pull Request: Add rtpTimestamp to RTCRtpContributingSource Johannes Kron via GitHub
• [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177) Johannes Kron via GitHub
• Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433) Philipp Hancke via GitHub
• Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433) henbos via GitHub
• [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433) henbos via GitHub
• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Philipp Hancke via GitHub
• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Lennart Grahl via GitHub
• Re: [mediacapture-fromelement] Should tracks captured from a media element fire "ended" when ending? (#77) Martin Thomson via GitHub
• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Bernard Aboba via GitHub

Tuesday, 23 April 2019

• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Justin Uberti via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• [mediacapture-main] echo cancellation scope (#577) Jai Gupta via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) armax00

Monday, 22 April 2019

• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Lennart Grahl via GitHub
• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Bernard Aboba via GitHub
• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Lennart Grahl via GitHub

Sunday, 21 April 2019

• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Bernard Aboba via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) guest271314 via GitHub
• Re: [webrtc-pc] Should DataChannels have a getStats() method, should they be usable as selector to the getStats algorithm? (#2174) Harald Alvestrand via GitHub

Saturday, 20 April 2019

• Re: [webrtc-pc] No procedure for the ICE failed state (#2004) Lennart Grahl via GitHub

Thursday, 18 April 2019

• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) na-g via GitHub
• Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) Bernard Aboba via GitHub
• Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Jan-Ivar Bruaroey via GitHub
• [mediacapture-fromelement] Should tracks captured from a media element fire "ended" when ending? (#77) Andreas Pehrson via GitHub
• Re: [mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34) Andreas Pehrson via GitHub

Wednesday, 17 April 2019

• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) wssjwl via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) wssjwl via GitHub
• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Add stopping state (transceiver.stop() queues to stable). Rename old unsafe stop to reject() (#2168) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Spec note on stop() underestimates BUNDLE problem. (#2176) henbos via GitHub
• Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432) Varun Singh via GitHub
• Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432) henbos via GitHub
• [webrtc-pc] Spec note on stop() underestimates BUNDLE problem. (#2176) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430) henbos via GitHub
• Closed: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429) henbos via GitHub
• Re: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429) Varun Singh via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175) Lennart Grahl via GitHub
• [webrtc-pc] Pull Request: Permission API for receive-only media and data use cases Lennart Grahl via GitHub
• Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432) Varun Singh via GitHub
• [webrtc-pc] Should DataChannels have a getStats() method, should they be usable as selector to the getStats algorithm? (#2174) na-g via GitHub

Tuesday, 16 April 2019

• Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432) na-g via GitHub
• Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430) Varun Singh via GitHub
• [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432) Varun Singh via GitHub
• Closed: [webrtc-stats] Additional Audio/Video Quality Metrics (#431) Bernard Aboba via GitHub
• Re: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429) Jan-Ivar Bruaroey via GitHub

Monday, 15 April 2019

• [webrtc-pc] Timing of setRemoteDescription's identity validation is unclear (#2173) Jan-Ivar Bruaroey via GitHub
• [webrtc-stats] Additional Audio/Video Quality Metrics (#431) Bernard Aboba via GitHub
• [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430) Bernard Aboba via GitHub
• [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429) Bernard Aboba via GitHub
• [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172) Chen Xing via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) armax00

Sunday, 14 April 2019

• Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4) guest271314 via GitHub
• Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4) guest271314 via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) guest271314 via GitHub
• Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171) Philipp Hancke via GitHub

Saturday, 13 April 2019

• [webrtc-pc] Clarify "multiple sources of media **stitched together**" at Note describing replaceTrack (#2171) guest271314 via GitHub

Friday, 12 April 2019

• Re: [webrtc-pc] Simulcast: Which layer gets dropped first? (#2080) Bernard Aboba via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) wssjwl via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub

Thursday, 11 April 2019

• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) wssjwl via GitHub
• Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026) Harald Alvestrand via GitHub
• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) guest271314 via GitHub
• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) guest271314 via GitHub
• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) guest271314 via GitHub
• Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) jesup via GitHub
• Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) youennf via GitHub
• Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170) Lennart Grahl via GitHub
• [webrtc-pc] data channel default binaryType value is 'blob' (#2170) youennf via GitHub
• Closed: [mediacapture-main] Spec does no handle fingerprinting related to exposing non default capture devices (#559) youennf via GitHub
• Re: [mediacapture-main] Spec does no handle fingerprinting related to exposing non default capture devices (#559) youennf via GitHub
• Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575) Harald Alvestrand via GitHub
• Re: [mediacapture-main] enumerateDevices is exposing devices labels for origins that are granted access once (#563) youennf via GitHub
• Closed: [mediacapture-main] enumerateDevices is exposing devices labels for origins that are granted access once (#563) youennf via GitHub
• Re: [mediacapture-main] What constraint name should be exposed in case of a getUserMedia query with multiple failing constraints (#562) youennf via GitHub
• Closed: [mediacapture-main] What constraint name should be exposed in case of a getUserMedia query with multiple failing constraints (#562) youennf via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [mediacapture-main] enumerateDevices can be used to track user devices in background pages (#561) youennf via GitHub
• Re: [mediacapture-main] enumerateDevices can be used to track user devices in background pages (#561) youennf via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Bernard Aboba via GitHub
• Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167) Bernard Aboba via GitHub
• Re: [webrtc-pc] Missing specification on how to assign bandwidth between encodings and/or drop simulcast layers (#2141) Harald Alvestrand via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub

Wednesday, 10 April 2019

• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] RTCAudioHandlerStats: Signal/Noise Ratio (#383) Bernard Aboba via GitHub
• [webrtc-stats] Pull Request: Audio Noise metric Bernard Aboba via GitHub
• Re: [webrtc-pc] Add pc.restartIce() method. (#2169) henbos via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) henbos via GitHub
• [webrtc-pc] new commits pushed by henbos henbos via GitHub
• Closed: [webrtc-pc] Remove stopped transceivers from getTransceivers() (#2092) henbos via GitHub
• Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152) henbos via GitHub
• Re: [webrtc-pc] Remove stopped transceivers after negotiation (#2160) henbos via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Philipp Hancke via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427) henbos via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Philipp Hancke via GitHub
• [webrtc-pc] Pull Request: Add pc.restartIce() method. Jan-Ivar Bruaroey via GitHub

Tuesday, 9 April 2019

• [webrtc-pc] Pull Request: Add stopping state (transceiver.stop() queues to stable). Rename old unsafe stop to reject() Jan-Ivar Bruaroey via GitHub
• [webrtc-stats] Request to add neteq waiting time to stats report (#427) wssjwl via GitHub
• Re: [webrtc-stats] RTCRemoteInboundRtpStreamStats: total round trip time (#426) Varun Singh via GitHub
• [webrtc-stats] RTCRemoteInboundRtpStreamStats: total round trip time (#426) henbos via GitHub
• Re: [webrtc-pc] Clarify or fix racy peerIdentity validation failure (#2148) guest271314 via GitHub
• [webrtc-stats] Byte/packet counter consistency (retransmittedPacketsSent is unsigned long long) (#425) henbos via GitHub

Monday, 8 April 2019

• Re: [webrtc-stats] Receiver attachment stats missing (#424) Varun Singh via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Philipp Hancke via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Receiver attachment stats missing (#424) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Philipp Hancke via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Harald Alvestrand via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Receiver attachment stats missing (#424) henbos via GitHub
• Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167) Philipp Hancke via GitHub
• Re: [mediacapture-main] Should we remove track attributes "remote" and "readonly"? (#320) guest271314 via GitHub

Sunday, 7 April 2019

• [webrtc-pc] ICE restart deserves a first-class API. (#2167) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Better API for rollback that's glare-proof (#2166) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] A simpler non-glare-prone setLocalDescription() (#2165) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Silence OperationError in addIceCandidate from rollback (#2164) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4) guest271314 via GitHub
• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub
• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub

Saturday, 6 April 2019

• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Lennart Grahl via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub

Friday, 5 April 2019

• [webrtc-pc] Pull Request: Restore firing of track event when there are no streams (editorial mistake) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-stats] Receiver attachment stats missing (#424) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Lennart Grahl via GitHub
• Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149) henbos via GitHub
• Re: [webrtc-stats] Receiver attachment stats missing (#424) henbos via GitHub
• Closed: [webrtc-stats] References to undefined RTCReceiverAudioTrackAttachmentStats and RTCReceiverVideoTrackAttachmentStats (#414) henbos via GitHub
• Re: [webrtc-stats] References to undefined RTCReceiverAudioTrackAttachmentStats and RTCReceiverVideoTrackAttachmentStats (#414) henbos via GitHub
• [webrtc-stats] Receiver attachment stats missing (#424) henbos via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) armax00
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) guidou via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) armax00

Thursday, 4 April 2019

• [webrtc-pc] Pull Request: Fix prose indicating reasons why sender may be missing in removeTrack (Editorial) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161) Dominique Hazael-Massieux via GitHub
• [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161) Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Too many test links in the spec, some redundant (#2161) Jan-Ivar Bruaroey via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) guidou via GitHub
• [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Does direction matter for rejected m= sections? (#1812) Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Does direction matter for rejected m= sections? (#1812) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Set the configuration is unnecessarily checking for some members being set (#2142) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Remove stopped transceivers after negotiation (#2160) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Add in-line references to relevant test cases (#2156) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Add in-line references to relevant test cases (#2156) Harald Alvestrand via GitHub
• Closed: [webrtc-pc] simulcast offer doesn't work with addTrack. An omission (#2147) 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
• [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] Add video latency description (#2110) henbos via GitHub
• Re: [webrtc-pc] Add in-line references to relevant test cases (#2156) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Add in-line references to relevant test cases (#2156) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Remove stopped transceivers from the set of transceivers (#2102) henbos via GitHub
• [webrtc-pc] Pull Request: Remove stopped transceivers after negotiation henbos via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Add a "panel test" to track implementation status (#346) Varun Singh via GitHub
• Re: [webrtc-pc] Attach simulcast offer sendEncodings to suitable (addTrack) transceiver. (#2155) Jan-Ivar Bruaroey via GitHub
• Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149) henbos via GitHub
• Re: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2139) henbos via GitHub
• Closed: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2139) henbos via GitHub
• Re: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2138) henbos via GitHub
• Re: [mediacapture-depth] Update editors metadata (#172) Anssi Kostiainen via GitHub
• [mediacapture-depth] Pull Request: Update editors metadata Anssi Kostiainen via GitHub
• Closed: [webrtc-stats] RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent (#418) henbos via GitHub
• [webrtc-stats] new commits pushed by henbos henbos via GitHub
• Re: [webrtc-stats] targetBitrate should not be "measured over 1 second window" (#423) henbos via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416) Harald Alvestrand via GitHub
• [webrtc-stats] targetBitrate should not be "measured over 1 second window" (#423) henbos via GitHub
• [webrtc-pc] Clarify update negotiation needed (#2159) henbos via GitHub
• Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576) Harald Alvestrand via GitHub
• [mediacapture-main] Pull Request: Fix for #573 armax00
• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub

Wednesday, 3 April 2019

• Re: [webrtc-stats] Update targetBitrate and add totalTargetBytesEncoded (#417) Varun Singh via GitHub
• Re: [webrtc-stats] References to undefined RTCReceiverAudioTrackAttachmentStats and RTCReceiverVideoTrackAttachmentStats (#414) Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add retransmittedPacketsSent and retransmittedBytesSent (#419) Varun Singh via GitHub
• [webrtc-stats] Capture vs encode frameWidth, frameHeight and framesPerSecond (#422) henbos via GitHub
• Re: [webrtc-stats] totalPacketSendDelay added (#421) Varun Singh via GitHub
• [webrtc-stats] Pull Request: totalPacketSendDelay added henbos via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Add stats for accelerating/decelarating playout speed (#407) Varun Singh via GitHub
• Re: [webrtc-stats] Add insertedSamplesForDeceleration and removedSamplesForAcceleration (#408) Varun Singh via GitHub
• [webrtc-stats] Stats for the delay of packets being sent (#420) henbos via GitHub
• [webrtc-stats] Pull Request: Add retransmittedPacketsSent and retransmittedBytesSent henbos via GitHub
• [webrtc-stats] RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent (#418) henbos via GitHub
• [webrtc-stats] Pull Request: Update targetBitrate and add totalTargetBytesEncoded henbos via GitHub
• Re: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416) henbos via GitHub
• Re: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416) Harald Alvestrand via GitHub
• [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416) henbos via GitHub
• Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149) henbos via GitHub

Tuesday, 2 April 2019

• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Michael Tüxen via GitHub
• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub
• Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147) guest271314 via GitHub
• Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149) Philip Jägenstedt via GitHub
• Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157) Lennart Grahl via GitHub
• Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Attach simulcast offer sendEncodings to suitable (addTrack) transceiver. (#2155) Harald Alvestrand via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Lennart Grahl via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub
• Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157) Lennart Grahl via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Lennart Grahl via GitHub
• Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub
• [webrtc-pc] DataChannel max value for "id" before connecting? (#2158) Harald Alvestrand via GitHub
• [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157) Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add insertedSamplesForDeceleration and removedSamplesForAcceleration (#408) henbos via GitHub
• Re: [webrtc-stats] Add stats for accelerating/decelarating playout speed (#407) henbos via GitHub
• Re: [mediacapture-screen-share] Issue 38 - Cursor constraint (#58) guest271314 via GitHub

Monday, 1 April 2019

• [mediacapture-image] Clarify applicability of MediaTrackSupportedConstraints (#206) Reilly Grant via GitHub
• Re: [webrtc-stats] Add stats for accelerating/decelarating playout speed (#407) Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Pull Request: Add in-line references to relevant test cases Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Pull Request: Attach simulcast offer sendEncodings to suitable (addTrack) transceiver. Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Pull Request: Fix link to description of DataChannel send() Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Pull Request: Mark 'hold functionality' section as informative Dominique Hazael-Massieux via GitHub
• [webrtc-pc] new commits pushed by jan-ivar Jan-Ivar Bruaroey via GitHub
• [webrtc-pc] Pull Request: Remove redundant steps in setCodecParameters Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Pull Request: Fix name of MTI stats in RTCDataChannelStats Dominique Hazael-Massieux via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Re: [webrtc-stats] Add insertedSamplesForDeceleration and removedSamplesForAcceleration (#408) Varun Singh via GitHub
• Re: [webrtc-stats] Add insertedSamplesForDeceleration and removedSamplesForAcceleration (#408) Varun Singh via GitHub

Last message date: Tuesday, 30 April 2019 18:58:30 UTC