Re: [mediacapture-main] Clarify that enumerateDevices must not expose devices that a given context cannot use through getUserMedia (#549)
[webrtc-pc] Says to count datachannel label & protocol in bytes, but they're USVString (#2191)
[mediacapture-screen-share] Pull Request: Add user activation check when calling getDisplayMedia
- Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106)
- Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106)
- Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106)
- Re: [mediacapture-screen-share] Add user activation check when calling getDisplayMedia (#106)
[webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#434)
[mediacapture-main] Pull Request: #577 make echo cancellation scope more explicit
[mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
- Re: [mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)
[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)
[webrtc-pc] Pull Request: Clarify `RTCRtpContributingSource` and `RTCRtpSynchronizationSource`.
[mediacapture-main] Travis is failing git checkout with "reference is not a tree" (#581)
[mediacapture-main] new commits pushed by henbos
Closed: [mediacapture-main] Remove overconstrained event (Why do we have overconstrained event?) (#573)
[mediacapture-main] Pull Request: Just testing... ignore this
[mediacapture-main] Pull Request: Remove "onoverconstrained" re-upload
[webrtc-pc] Define time order of results from getContributingSources() et al (#2189)
[webrtc-pc] new commits pushed by dontcallmedom
Closed: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] Pull Request: Define which constraints are applicable in WebRTC
Re: [webrtc-pc] Constrainable properties on remote tracks are under-specified (#2121)
[mediacapture-main] Add note on applicability of constraints to tracks in other specs (#578)
[webrtc-pc] Pull Request: Clarify a step in "update negotiation needed"
Re: [webrtc-pc] Simulcast stats (#2116)
Closed: [webrtc-pc] Simulcast stats (#2116)
Closed: [webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103)
Re: [webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103)
[webrtc-pc] Pull Request: Adds a note clarifying removeTrack()
[webrtc-pc] Mark getDefaultIceServers() as feature at risk. (#2185)
Re: [webrtc-pc] Should we remove getDefaultIceServers? (#2023)
[webrtc-pc] Pull Request: Mark getDefaultIceServers() as feature at risk
Closed: [webrtc-pc] how to use datachannel bufferedamountlow threshold callback? (#1979)
Re: [webrtc-pc] how to use datachannel bufferedamountlow threshold callback? (#1979)
Re: [webrtc-pc] RTCRtpReceiver.getStreams() and RTCRtpSender.getStreams() (#1921)
Closed: [webrtc-pc] RTCRtpReceiver.getStreams() and RTCRtpSender.getStreams() (#1921)
[webrtc-pc] There are redundant steps in setCodecPreferences (#2183)
- Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183)
- Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183)
- Closed: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183)
- Re: [webrtc-pc] There are redundant steps in setCodecPreferences (#2183)
[webrtc-pc] Pull Request: Make sure prflx remote candidates do not leak information when exposed by getSelectedCandidatePair
[webrtc-pc] Pull Request: Cap expires to 365 days at most
Re: [webrtc-pc] RTCIceTransport.getRemoteCandidates() does not return prflx candidates (#2124)
[webrtc-pc] Pull Request: Clarify "multiple sources of media stitched together"
[webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
- Closed: [webrtc-pc] Allow to import existing certificate (#2179)
- Re: [webrtc-pc] Allow to import existing certificate (#2179)
Re: [webrtc-pc] Allow to import existing certificate (#1853)
[webrtc-pc] Pull Request: Add rtpTimestamp to RTCRtpContributingSource
- Re: [webrtc-pc] Add rtpTimestamp to RTCRtpContributingSource (#2178)
- Re: [webrtc-pc] Add rtpTimestamp to RTCRtpContributingSource (#2178)
[webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
- Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
- Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
- Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
- Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
- Re: [webrtc-pc] Add RTP timestamp to RTCRtpContributingSource (#2177)
[webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433)
- Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433)
- Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433)
- Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433)
- Re: [webrtc-stats] The spec has incorrect assumptions abound SR and RR (#433)
[mediacapture-main] echo cancellation scope (#577)
- Re: [mediacapture-main] echo cancellation scope (#577)
- Re: [mediacapture-main] echo cancellation scope (#577)
- Re: [mediacapture-main] echo cancellation scope (#577)
- Re: [mediacapture-main] echo cancellation scope (#577)
Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
- Re: [webrtc-pc] No procedure for the ICE failed state (#2004)
[mediacapture-fromelement] Should tracks captured from a media element fire "ended" when ending? (#77)
Re: [mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34)
Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150)
- Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150)
- Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150)
- Re: [webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150)
[webrtc-pc] Spec note on stop() underestimates BUNDLE problem. (#2176)
[webrtc-pc] Pull Request: Permission API for receive-only media and data use cases
- Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175)
- Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175)
- Re: [webrtc-pc] Permission API for receive-only media and data use cases (#2175)
[webrtc-pc] Should DataChannels have a getStats() method, should they be usable as selector to the getStats algorithm? (#2174)
[webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432)
- Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432)
- Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432)
- Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432)
- Re: [webrtc-stats] OPUS -- Exposing SILK or CELT mode in getStats (#432)
[webrtc-pc] Timing of setRemoteDescription's identity validation is unclear (#2173)
[webrtc-stats] Additional Audio/Video Quality Metrics (#431)
[webrtc-stats] RTCVideo*Stats: Video Device Errors (#430)
- Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430)
- Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430)
- Re: [webrtc-stats] RTCVideo*Stats: Video Device Errors (#430)
[webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429)
- Re: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429)
- Re: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429)
- Closed: [webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429)
[webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
- Re: [webrtc-pc] Clarify the definition of "playout" for `RTCRtpContributingSource`. (#2172)
[webrtc-pc] Clarify "multiple sources of media **stitched together**" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
- Re: [webrtc-pc] Clarify "multiple sources of media stitched together" at Note describing replaceTrack (#2171)
Re: [webrtc-pc] Simulcast: Which layer gets dropped first? (#2080)
Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026)
- Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026)
- Re: [webrtc-pc] Should WPT webrtc folder be renamed to webrtc-pc (#2026)
[webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Re: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
- Closed: [webrtc-pc] data channel default binaryType value is 'blob' (#2170)
Closed: [mediacapture-main] Spec does no handle fingerprinting related to exposing non default capture devices (#559)
Re: [mediacapture-main] Spec does no handle fingerprinting related to exposing non default capture devices (#559)
Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
- Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
- Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
- Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
- Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
- Re: [mediacapture-main] Proposal: DecodeConcatVideoData (input multiple files or streams) => Output: single webm file (#575)
Re: [mediacapture-main] enumerateDevices is exposing devices labels for origins that are granted access once (#563)
Closed: [mediacapture-main] enumerateDevices is exposing devices labels for origins that are granted access once (#563)
Re: [mediacapture-main] What constraint name should be exposed in case of a getUserMedia query with multiple failing constraints (#562)
Closed: [mediacapture-main] What constraint name should be exposed in case of a getUserMedia query with multiple failing constraints (#562)
[webrtc-pc] new commits pushed by aboba
Closed: [mediacapture-main] enumerateDevices can be used to track user devices in background pages (#561)
Re: [mediacapture-main] enumerateDevices can be used to track user devices in background pages (#561)
[webrtc-pc] new commits pushed by aboba
Re: [webrtc-pc] Missing specification on how to assign bandwidth between encodings and/or drop simulcast layers (#2141)
Re: [webrtc-stats] RTCAudioHandlerStats: Signal/Noise Ratio (#383)
[webrtc-stats] Pull Request: Audio Noise metric
[webrtc-pc] new commits pushed by henbos
Closed: [webrtc-pc] Remove stopped transceivers from getTransceivers() (#2092)
[webrtc-pc] Pull Request: Add pc.restartIce() method.
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
- Re: [webrtc-pc] Add pc.restartIce() method. (#2169)
[webrtc-pc] Pull Request: Add stopping state (transceiver.stop() queues to stable). Rename old unsafe stop to reject()
[webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
- Re: [webrtc-stats] Request to add neteq waiting time to stats report (#427)
[webrtc-stats] RTCRemoteInboundRtpStreamStats: total round trip time (#426)
Re: [webrtc-pc] Clarify or fix racy peerIdentity validation failure (#2148)
[webrtc-stats] Byte/packet counter consistency (retransmittedPacketsSent is unsigned long long) (#425)
Re: [mediacapture-main] Should we remove track attributes "remote" and "readonly"? (#320)
[webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] ICE restart deserves a first-class API. (#2167)
- Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167)
- Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167)
- Re: [webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167)
[webrtc-pc] Better API for rollback that's glare-proof (#2166)
[webrtc-pc] A simpler non-glare-prone setLocalDescription() (#2165)
[webrtc-pc] Silence OperationError in addIceCandidate from rollback (#2164)
Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4)
- Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4)
- Re: [mediacapture-record] MediaRecorder needs to define effect of adding / removing tracks in its input MediaStream (#4)
[webrtc-pc] Pull Request: Restore firing of track event when there are no streams (editorial mistake)
Closed: [webrtc-stats] References to undefined RTCReceiverAudioTrackAttachmentStats and RTCReceiverVideoTrackAttachmentStats (#414)
[webrtc-stats] Receiver attachment stats missing (#424)
- Re: [webrtc-stats] Receiver attachment stats missing (#424)
- Re: [webrtc-stats] Receiver attachment stats missing (#424)
- Re: [webrtc-stats] Receiver attachment stats missing (#424)
- Re: [webrtc-stats] Receiver attachment stats missing (#424)
- Re: [webrtc-stats] Receiver attachment stats missing (#424)
[webrtc-pc] Pull Request: Fix prose indicating reasons why sender may be missing in removeTrack (Editorial)
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] Too many test links in the spec, some redundant (#2161)
- Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161)
- Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161)
- Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161)
- Re: [webrtc-pc] Too many test links in the spec, some redundant (#2161)
[webrtc-pc] new commits pushed by dontcallmedom
Re: [webrtc-pc] Does direction matter for rejected m= sections? (#1812)
Closed: [webrtc-pc] Does direction matter for rejected m= sections? (#1812)
Re: [webrtc-pc] Set the configuration is unnecessarily checking for some members being set (#2142)
Closed: [webrtc-pc] simulcast offer doesn't work with addTrack. An omission (#2147)
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by alvestrand
Re: [webrtc-pc] Add video latency description (#2110)
Re: [webrtc-pc] Remove stopped transceivers from the set of transceivers (#2102)
[webrtc-pc] Pull Request: Remove stopped transceivers after negotiation
- Re: [webrtc-pc] Remove stopped transceivers after negotiation (#2160)
- Re: [webrtc-pc] Remove stopped transceivers after negotiation (#2160)
[webrtc-stats] new commits pushed by vr000m
Closed: [webrtc-stats] Add a "panel test" to track implementation status (#346)
Re: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2139)
Closed: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2139)
Re: [webrtc-pc] Add setTargetJitterBufferDelay method to RTCRtpReceiver (#2138)
[mediacapture-depth] Pull Request: Update editors metadata
[webrtc-stats] new commits pushed by henbos
[webrtc-stats] new commits pushed by alvestrand
[webrtc-stats] targetBitrate should not be "measured over 1 second window" (#423)
[webrtc-pc] Clarify update negotiation needed (#2159)
[mediacapture-main] Pull Request: Fix for #573
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (Fix for #573) (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (#576)
- Re: [mediacapture-main] Remove "onoverconstrained" (#576)
Re: [webrtc-stats] References to undefined RTCReceiverAudioTrackAttachmentStats and RTCReceiverVideoTrackAttachmentStats (#414)
[webrtc-stats] Capture vs encode frameWidth, frameHeight and framesPerSecond (#422)
[webrtc-stats] Pull Request: totalPacketSendDelay added
[webrtc-stats] new commits pushed by vr000m
Closed: [webrtc-stats] Add stats for accelerating/decelarating playout speed (#407)
[webrtc-stats] Stats for the delay of packets being sent (#420)
[webrtc-stats] Pull Request: Add retransmittedPacketsSent and retransmittedBytesSent
[webrtc-stats] RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent (#418)
[webrtc-stats] Pull Request: Update targetBitrate and add totalTargetBytesEncoded
[webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416)
- Re: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416)
- Re: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416)
- Closed: [webrtc-stats] Rethinking RTCOutboundRtpStreamStats.targetBitrate (#416)
Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
- Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
- Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
- Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
- Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
- Re: [mediacapture-record] Proposal: Specify ability to pause and resume between adding and removing MediaStreamTracks to an active MediaStream (#147)
Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149)
- Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149)
- Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149)
- Re: [webrtc-pc] RTCRtpReceiver.jitterBufferDelayHint attribute added. (#2149)
[webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
- Re: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)
[webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157)
- Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157)
- Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157)
- Re: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157)
Re: [mediacapture-screen-share] Issue 38 - Cursor constraint (#58)
[mediacapture-image] Clarify applicability of MediaTrackSupportedConstraints (#206)
Re: [webrtc-stats] Add stats for accelerating/decelarating playout speed (#407)
[webrtc-pc] Pull Request: Add in-line references to relevant test cases
- Re: [webrtc-pc] Add in-line references to relevant test cases (#2156)
- Re: [webrtc-pc] Add in-line references to relevant test cases (#2156)
- Re: [webrtc-pc] Add in-line references to relevant test cases (#2156)
- Re: [webrtc-pc] Add in-line references to relevant test cases (#2156)
[webrtc-pc] Pull Request: Attach simulcast offer sendEncodings to suitable (addTrack) transceiver.
- Re: [webrtc-pc] Attach simulcast offer sendEncodings to suitable (addTrack) transceiver. (#2155)
- Re: [webrtc-pc] Attach simulcast offer sendEncodings to suitable (addTrack) transceiver. (#2155)
[webrtc-pc] Pull Request: Fix link to description of DataChannel send()
[webrtc-pc] Pull Request: Mark 'hold functionality' section as informative
[webrtc-pc] new commits pushed by jan-ivar
[webrtc-pc] Pull Request: Remove redundant steps in setCodecParameters
- Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152)
- Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152)
- Re: [webrtc-pc] Remove redundant steps in setCodecParameters (#2152)