public-webrtc-logs@w3.org from January 2018 by date

Wednesday, 31 January 2018

• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• [webrtc-stats] Pull Request: tidy checks, missed in latest PRs. Varun Singh via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Respec issue: Locked to too old version? Harald Alvestrand via GitHub
• Closed: [webrtc-stats] averageRTCPInterval should be averageRtcpInterval Varun Singh via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Varun Singh via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-stats] Work through implications of simulcast on the receiver side henbos via GitHub
• Re: [webrtc-stats] Work through implications of simulcast on the receiver side henbos via GitHub
• Re: [webrtc-stats] Define object lifetimes and mention end-of-life emission Varun Singh via GitHub
• Re: [webrtc-stats] Add per layer stats for SVC Harald Alvestrand via GitHub
• Re: [webrtc-stats] Typo - RTCLocalAudioStreamTrackStats should inherit from RTCAudioSenderStats Varun Singh via GitHub
• Re: [webrtc-stats] Typo - RTCLocalAudioStreamTrackStats should inherit from RTCAudioSenderStats Harald Alvestrand via GitHub
• Re: [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries henbos via GitHub
• Re: [webrtc-stats] Cleanup: reformat the sections Harald Alvestrand via GitHub
• [webrtc-stats] Work through implications of simulcast on the receiver side Harald Alvestrand via GitHub
• Re: [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries Harald Alvestrand via GitHub
• Re: [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries Varun Singh via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Harald Alvestrand via GitHub
• Closed: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Deleting framesCorrupted Varun Singh via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Harald Alvestrand via GitHub
• Re: [webrtc-stats] fix camel case averageRtcpInterval and also put in RTCInboundRTPStreamStats Varun Singh via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Add fecPacketsSent/Received Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] What happens when a partial keyFrames is received? Varun Singh via GitHub
• Closed: [webrtc-stats] packetsDuplicated is not reported in an RTCP Report Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Definitions from MSE need re-targeting Varun Singh via GitHub
• Re: [webrtc-pc] Respec issue: Locked to too old version? Vivien Lacourba via GitHub
• Re: [webrtc-pc] Respec issue: Locked to too old version? Harald Alvestrand via GitHub
• [webrtc-stats] Typo - RTCLocalAudioStreamTrackStats should inherit from RTCAudioSenderStats karthikbr82 via GitHub
• [webrtc-stats] Cleanup: reformat the sections Varun Singh via GitHub
• Re: [webrtc-pc] Sort out unmute and BYE Harald Alvestrand via GitHub
• [webrtc-stats] Pull Request: Define object lifetimes and mention end-of-life emission Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename RTCRTPStreamStats to RTCRtpStreamStats? Varun Singh via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] Respec issue: Locked to too old version? Vivien Lacourba via GitHub
• Closed: [webrtc-pc] Respec issue: Locked to too old version? Vivien Lacourba via GitHub
• [webrtc-pc] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
• Re: [webrtc-pc] Upgrade to respec 1.9.0.2 Vivien Lacourba via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages stefan hakansson via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Anne van Kesteren via GitHub
• [webrtc-pc] Pull Request: Upgrade to respec 1.9.0.2 Vivien Lacourba via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages stefan hakansson via GitHub
• [webrtc-pc] Respec issue: Locked to too old version? Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: Add "statslifetimeended" event Harald Alvestrand via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Anne van Kesteren via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages stefan hakansson via GitHub
• [webrtc-pc] Sort out unmute and BYE stefan hakansson via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Adam Rice via GitHub

Tuesday, 30 January 2018

• [webrtc-pc] Need <section> around RTCIceTransport dictionaries Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Need <section> around dictionaries under RTCRtpSender Bernard Aboba via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries Varun Singh via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Varun Singh via GitHub
• [webrtc-stats] Pull Request: fix for timestamp for remote and local objects Varun Singh via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Harald Alvestrand via GitHub
• Re: [webrtc-pc] Need <section> around dictionaries under RTCRtpSender Harald Alvestrand via GitHub
• [webrtc-pc] Need <section> around dictionaries under RTCRtpSender Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: Gating 'unmute' events on track with transceiver direction. stefan hakansson via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Taylor Brandstetter via GitHub

Monday, 29 January 2018

• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? jan-ivar via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Taylor Brandstetter via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? jan-ivar via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? jan-ivar via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Taylor Brandstetter via GitHub
• [webrtc-pc] Direction is not readonly stefan hakansson via GitHub
• Re: [webrtc-pc] Integrate CSP access control into algorithms Harald Alvestrand via GitHub
• Re: [webrtc-pc] Codify refusing to generate an empty Offer Philipp Hancke via GitHub
• Re: [webrtc-pc] Codify refusing to generate an empty Offer Harald Alvestrand via GitHub
• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: Pending remote updated by setRemote. stefan hakansson via GitHub
• Re: [webrtc-pc] Codify refusing to generate an empty Offer Philipp Hancke via GitHub
• [webrtc-pc] Codify refusing to generate an empty Offer Harald Alvestrand via GitHub

Sunday, 28 January 2018

• Re: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Varun Singh via GitHub

Saturday, 27 January 2018

• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats alex gouaillard via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Harald Alvestrand via GitHub

Friday, 26 January 2018

• [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated? Taylor Brandstetter via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Anne van Kesteren via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub

Thursday, 25 January 2018

• Closed: [webrtc-pc] specify legacy onaddstream? jan-ivar via GitHub
• Closed: [webrtc-pc] When to fire events triggered by setRemoteDescription. jan-ivar via GitHub
• Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes Harald Alvestrand via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Harald Alvestrand via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Harald Alvestrand via GitHub
• Closed: [webrtc-pc] canTrickleIceCandidiates question Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Clarify that "disconnected" is transient. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Clarify that "disconnected" is transient. 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] WHATWG streams for data channel messages Lennart Grahl via GitHub
• [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries Varun Singh via GitHub
• [webrtc-stats] Pull Request: fix camel case averageRtcpInterval and also put in RTCInboundRTPStreamStats Varun Singh via GitHub
• [webrtc-stats] Pull Request: FEC sent and received metrics Varun Singh via GitHub
• [webrtc-stats] Pull Request: moved packetsDuplicated to the right dictionary Varun Singh via GitHub
• [webrtc-stats] Deleting framesCorrupted Varun Singh via GitHub

Wednesday, 24 January 2018

• Re: [webrtc-pc] WHATWG streams for data channel messages Bernard Aboba via GitHub
• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Cullen Jennings via GitHub
• Re: [webrtc-stats] Remove references to MSE. Fixes #161 Varun Singh via GitHub
• Re: [webrtc-stats] RTCCodecStats in every getStats() is redundant. karthikbr82 via GitHub
• Re: [webrtc-stats] Remove references to MSE. Fixes #161 Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Rename "objectDeleted" to something else. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename "objectDeleted" to something else. Harald Alvestrand via GitHub
• Re: [webrtc-stats] terminology: rename mediaType to kind Harald Alvestrand via GitHub
• Re: [webrtc-stats] Lifetime of RTPStreamStats Varun Singh via GitHub
• Re: [webrtc-stats] Lifetime of RTPStreamStats Varun Singh via GitHub
• Re: [webrtc-stats] CodecType "encode" / "decode" needs to default to "both" Harald Alvestrand via GitHub
• Closed: [webrtc-stats] CodecType "encode" / "decode" needs to default to "both" Varun Singh via GitHub
• Re: [webrtc-stats] CodecType "encode" / "decode" needs to default to "both" Varun Singh via GitHub
• Re: [webrtc-stats] CodecType "encode" / "decode" needs to default to "both" Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stats need to mark up which members are required Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? jan-ivar via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCCodecStats in every getStats() is redundant. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? jan-ivar via GitHub
• Re: [webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Fixes #161 Varun Singh via GitHub
• [webrtc-stats] Pull Request: Fixes #161 Varun Singh via GitHub
• Closed: [webrtc-stats] RTCCodecStats in every getStats() is redundant. Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCCodecStats in every getStats() is redundant. Harald Alvestrand via GitHub

Tuesday, 23 January 2018

• Re: [webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted? jan-ivar via GitHub
• Re: [webrtc-stats] RTCCodecStats in every getStats() is redundant. Philipp Hancke via GitHub
• [webrtc-stats] RTCCodecStats in getStats() is redundant. karthikbr82 via GitHub
• Re: [webrtc-pc] Multiple SRDs may leave streams and tracks in unexpected state. jan-ivar via GitHub
• [webrtc-stats] averageRTCPInterval should be averageRtcpInterval Harald Alvestrand via GitHub
• [webrtc-stats] Stats need to mark up which members are required Harald Alvestrand via GitHub
• [webrtc-stats] CodecType "encode" / "decude" needs to default to "both" Harald Alvestrand via GitHub
• Re: [webrtc-pc] Multiple SRDs may leave streams and tracks in unexpected state. stefan hakansson via GitHub
• [webrtc-stats] Lifetime of RTPStreamStats Harald Alvestrand via GitHub

Monday, 22 January 2018

• Closed: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers jan-ivar via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers jan-ivar via GitHub
• [webrtc-pc] Multiple SRDs may leave streams and tracks in unexpected state. jan-ivar via GitHub
• Re: [webrtc-pc] Defer SRD add/remove tracks until right before firing track events. jan-ivar via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Varun Singh via GitHub
• Closed: [webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? Varun Singh via GitHub
• Re: [webrtc-stats] moved bytes and packets received counters Varun Singh via GitHub
• [webrtc-stats] terminology: rename mediaType to kind Philipp Hancke via GitHub
• [webrtc-pc] Pull Request: Properly defined direction when creating transceiver. stefan hakansson via GitHub
• Re: [webrtc-pc] Terminology around "setting" attributes may be incorrect Adam Bergkvist via GitHub
• Re: [webrtc-pc] Terminology around "setting" attributes may be incorrect Adam Bergkvist via GitHub

Friday, 19 January 2018

• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Data channel closing procedure Lennart Grahl via GitHub
• Re: [webrtc-pc] Integrate CSP access control into algorithms Sergio Garcia Murillo via GitHub
• [webrtc-pc] Integrate CSP access control into algorithms Harald Alvestrand via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Taylor Brandstetter via GitHub

Thursday, 18 January 2018

• Re: [webrtc-pc] Data channel closing procedure Taylor Brandstetter via GitHub
• [webrtc-pc] Pull Request: Clarify that "disconnected" is transient (editorial) jan-ivar via GitHub
• [webrtc-pc] Clarify that "disconnected" is transient. jan-ivar via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? Harald Alvestrand via GitHub
• Re: [webrtc-pc] WebRTC bypass CSP connect-src policies Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection? Harald Alvestrand via GitHub
• Re: [webrtc-pc] Data channel closing procedure Lennart Grahl via GitHub
• Closed: [webrtc-pc] order of transceivers, senders/receivers? Bernard Aboba via GitHub
• Closed: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Bernard Aboba via GitHub
• Closed: [webrtc-pc] Processing Remote MediaStreamTracks should pass on receiver when constructing track event 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
• Closed: [webrtc-pc] Effect of mute/disable on on-the-wire framerate is not described Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Do not consider direction in "need negotiation" evaluation for replaceTrack Taylor Brandstetter via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] RTCDtlsTransportState enum descriptions are lacking Taylor Brandstetter via GitHub
• Re: [webrtc-pc] replaceTrack "negotiation needed" clarification Daniel Burnett via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Private key access? Harald Alvestrand via GitHub
• Re: [webrtc-pc] Add support for WebRTC Data Channel in Workers Benjamin Gruenbaum via GitHub
• [webrtc-pc] Private key access? Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] Using setConfiguration() to add application certificates to an RTCPeerConnection post-construction? stefan hakansson via GitHub
• [webrtc-pc] Pull Request: Passing receiver when constructing track event in 'Processing Remote … stefan hakansson via GitHub
• Re: [webrtc-pc] Processing Remote MediaStreamTracks should pass on receiver when constructing track event stefan hakansson via GitHub
• Re: [webrtc-pc] Define pc.getTransceivers() et al to be in insertion order. stefan hakansson via GitHub
• Re: [webrtc-pc] replaceTrack "negotiation needed" clarification stefan hakansson via GitHub
• Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes stefan hakansson via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? henbos via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId henbos via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers stefan hakansson via GitHub
• Re: [webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats jan-ivar via GitHub
• Re: [webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted? jan-ivar via GitHub

Wednesday, 17 January 2018

• Re: [webrtc-stats] Is keeping stats around a memory problem? jan-ivar via GitHub
• [webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted? jan-ivar via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId Taylor Brandstetter via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Domenic Denicola via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Bernard Aboba via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers jan-ivar via GitHub
• Re: [webrtc-pc] Add direction change as reason for "mute" event jan-ivar via GitHub
• Re: [webrtc-pc] Add direction change as reason for "mute" event Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• [webrtc-stats] Add fecPacketsSent/Received Varun Singh via GitHub
• [webrtc-pc] Add direction change as reason for "mute" event stefan hakansson via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename RTCRTPStreamStats to RTCRtpStreamStats? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats henbos via GitHub
• Re: [webrtc-stats] Rename "objectDeleted" to something else. Varun Singh via GitHub
• Re: [webrtc-stats] Rename "objectDeleted" to something else. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats Varun Singh via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? henbos via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Varun Singh via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub
• Re: [webrtc-stats] moved bytes and packets received counters Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings? Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats henbos via GitHub
• Re: [webrtc-stats] Rename sender/receiver/track stats Harald Alvestrand via GitHub
• [webrtc-stats] Rename sender/receiver/track stats henbos via GitHub
• Re: [webrtc-stats] Additional description of audioLevel Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId henbos via GitHub
• Re: [webrtc-stats] Remove reference to RTCRtpSynchronizationSource.audioLevel. Harald Alvestrand via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers stefan hakansson via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Adam Rice via GitHub
• [webrtc-pc] Pull Request: Convert audioLevel to double. jan-ivar via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Taylor Brandstetter via GitHub

Tuesday, 16 January 2018

• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers jan-ivar via GitHub
• [webrtc-pc] Editorial: PendingRemoteDescription is set by.... Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add per layer stats for SVC Sergey Silkin via GitHub

Monday, 15 January 2018

• Re: [webrtc-pc] Should the spec describe addStream/onaddstream as legacy API? youennf via GitHub
• Re: [webrtc-pc] Maximum message size slightly incorrect Philip Jägenstedt via GitHub
• Closed: [webrtc-pc] Maximum message size slightly incorrect Adam Bergkvist via GitHub
• Re: [webrtc-pc] Maximum message size slightly incorrect Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Maximum message size slightly incorrect Philip Jägenstedt via GitHub

Sunday, 14 January 2018

• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers stefan hakansson via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers Harald Alvestrand via GitHub

Saturday, 13 January 2018

• Re: [webrtc-pc] Should the spec describe addStream/onaddstream as legacy API? Philip Jägenstedt via GitHub
• Re: [webrtc-pc] WHATWG streams for data channel messages Lennart Grahl via GitHub

Friday, 12 January 2018

• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats Taylor Brandstetter via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Taylor Brandstetter via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats jan-ivar via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Iñaki Baz Castillo via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Martin Thomson via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Martin Thomson via GitHub

Thursday, 11 January 2018

• [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats na-g via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Taylor Brandstetter via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId Taylor Brandstetter via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers Taylor Brandstetter via GitHub
• [webrtc-pc] Pull Request: Make audioLevel and voiceActivityFlag optional and non-nullable. jan-ivar via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? jan-ivar via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Lennart Grahl via GitHub
• Re: [webrtc-pc] WHATWG streams for data channels Domenic Denicola via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Varun Singh via GitHub
• [webrtc-stats] Rename "objectDeleted" to something else. henbos via GitHub
• Re: [webrtc-pc] Effect of mute/disable on on-the-wire framerate is not described Harald Alvestrand via GitHub
• [webrtc-stats] Define timestamps in terms of performance.origin Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition Harald Alvestrand via GitHub
• Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? Harald Alvestrand via GitHub
• Re: [webrtc-pc] addTransceiver woes Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Lennart Grahl via GitHub
• Re: [webrtc-pc] Define pc.getTransceivers() et al to be in insertion order. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Maximum message size slightly incorrect Adam Bergkvist 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] replaceTrack: Clarify how the UA determines if negotiation is needed Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub
• [webrtc-pc] Pull Request: RTCRtpSender: Specify advise about framerate for disabled/muted tracks Adam Bergkvist via GitHub
• [webrtc-pc] Do not consider direction at "need negotiation" evaluation for replaceTrack stefan hakansson via GitHub
• Re: [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId henbos via GitHub
• [webrtc-stats] Rename RTCRTPStreamStats to RTCRtpStreamStats? henbos via GitHub
• [webrtc-stats] RTCRTPStreamStats should have senderId/receiverId henbos via GitHub

Wednesday, 10 January 2018

• Re: [webrtc-stats] Need DSCP information for outgoing RTP streams jan-ivar via GitHub
• Re: [webrtc-pc] editorial: media api introduction Philipp Hancke via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers docfaraday via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers Taylor Brandstetter via GitHub
• Re: [webrtc-pc] editorial: media api introduction Taylor Brandstetter via GitHub
• Re: [webrtc-stats] Enable continuous publication Vivien Lacourba via GitHub
• Closed: [webrtc-stats] Need DSCP information for incoming RTP streams Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Need DSCP information for outgoing RTP streams Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Enable continuous publication Vivien Lacourba via GitHub
• [webrtc-stats] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data Varun Singh via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question cdh4u via GitHub
• Re: [webrtc-stats] Enable continuous publication Harald Alvestrand via GitHub
• [webrtc-stats] Enable continuous publication Harald Alvestrand via GitHub
• Re: [webrtc-stats] WiFi Stats. Harald Alvestrand via GitHub
• Re: [webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type henbos via GitHub
• Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter Varun Singh via GitHub
• Re: [webrtc-stats] Adds per-DSCP packet counters to RTP streams Harald Alvestrand via GitHub
• Re: [webrtc-stats] Introduction to RTP stream statistics Harald Alvestrand via GitHub
• Re: [webrtc-stats] moved bytes and packets received counters Varun Singh via GitHub
• Re: [webrtc-stats] Additional description of audioLevel Harald Alvestrand via GitHub
• Closed: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one henbos via GitHub
• Closed: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one henbos via GitHub
• Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add "sender" and "receiver" stats. Harald Alvestrand via GitHub
• Re: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers docfaraday via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. Harald Alvestrand via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Philipp Hancke via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? youennf via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? youennf via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Philip Jägenstedt via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? henbos via GitHub
• Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Should the spec describe addStream/onaddstream as legacy API? youennf via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? youennf via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Possibly racy replaceTrack() stefan hakansson via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? henbos via GitHub
• Re: [webrtc-pc] Should the spec describe addStream/onaddstream as legacy API? Philip Jägenstedt via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Philipp Hancke via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Harald Alvestrand via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Philip Jägenstedt via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? Harald Alvestrand via GitHub
• Re: [webrtc-pc] editorial: media api introduction Philipp Hancke via GitHub
• Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries. jan-ivar via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. jan-ivar via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Philipp Hancke via GitHub
• Re: [webrtc-pc] editorial: media api introduction Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Possibly racy replaceTrack() Taylor Brandstetter via GitHub
• [webrtc-pc] "track" event will fire extra times if applying multiple remote offers Taylor Brandstetter via GitHub

Tuesday, 9 January 2018

• Re: [webrtc-pc] canTrickleIceCandidiates question Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Possibly racy replaceTrack() henbos via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition Boris Zbarsky via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition jan-ivar via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition jan-ivar via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition Boris Zbarsky via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition Boris Zbarsky via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition jan-ivar via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Philipp Hancke via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question cdh4u via GitHub
• Re: [webrtc-pc] Add support for WebRTC Data Channel in Workers Lennart Grahl via GitHub
• Re: [webrtc-pc] Possibly racy replaceTrack() henbos via GitHub
• [webrtc-pc] Possibly racy replaceTrack() henbos via GitHub
• Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Bernard Aboba via GitHub

Monday, 8 January 2018

• Re: [webrtc-pc] canTrickleIceCandidiates question Taylor Brandstetter via GitHub
• Re: [webrtc-pc] editorial: media api introduction Bernard Aboba via GitHub
• Re: [webrtc-pc] editorial: media api introduction Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: replaceTrack "negotiation needed" clarification Bernard Aboba via GitHub
• Re: [webrtc-pc] Data channel closing procedure Bernard Aboba via GitHub
• Re: [webrtc-pc] Whether KeyingMaterial internal slot can be stored and retrieved Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Whether KeyingMaterial internal slot can be stored and retrieved Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Underlying data transport explanation Bernard Aboba via GitHub
• [webrtc-stats] Pull Request: Introduction to RTP stream statistics Harald Alvestrand via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Harald Alvestrand via GitHub
• Re: [webrtc-pc] canTrickleIceCandidiates question Harald Alvestrand via GitHub
• [webrtc-pc] canTrickleIceCandidiates question Philipp Hancke via GitHub
• [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
• Re: [webrtc-stats] Adds per-DSCP packet counters to RTP streams Varun Singh via GitHub
• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Dominique Hazael-Massieux via GitHub
• [webrtc-stats] Pull Request: Adds per-DSCP packet counters to RTP streams Harald Alvestrand via GitHub

Sunday, 7 January 2018

• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Iñaki Baz Castillo via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub

Saturday, 6 January 2018

• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Miguel París via GitHub
• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Miguel París via GitHub
• Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type Miguel París via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs stefan hakansson via GitHub

Friday, 5 January 2018

• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Iñaki Baz Castillo via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs stefan hakansson via GitHub
• Re: [webrtc-stats] WiFi Stats. Varun Singh via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Iñaki Baz Castillo via GitHub
• [webrtc-pc] Pull Request: Defer SRD add/remove tracks until right before firing track events. jan-ivar via GitHub

Thursday, 4 January 2018

• Re: [webrtc-stats] WiFi Stats. Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Define pc.getTransceivers()) et al to be in insertion order. jan-ivar via GitHub
• Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] addTrack can reuse a sender whose transceiver is stopped 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] Maximum message size slightly incorrect Adam Bergkvist via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Philipp Hancke via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs stefan hakansson via GitHub
• Closed: [webrtc-pc] At risk text in wrong location stefan hakansson via GitHub
• Re: [webrtc-pc] At risk text in wrong location stefan hakansson via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Philipp Hancke via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data minyuel via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Philipp Hancke via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs stefan hakansson via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs Philipp Hancke via GitHub
• Re: [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? henbos via GitHub
• Closed: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? henbos via GitHub
• [webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs henbos via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? stefan hakansson via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? henbos via GitHub
• Re: [webrtc-pc] Make promise rejection/enqueing consistent stefan hakansson via GitHub
• Re: [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection? Harald Alvestrand via GitHub
• Re: [webrtc-pc] replaceTrack: Never negotiate when replacing an ended track? Adam Bergkvist via GitHub
• Re: [webrtc-stats] WiFi Stats. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Using setConfiguration() to add application certificates to an RTCPeerConnection post-construction? Bernard Aboba via GitHub
• Re: [webrtc-pc] At risk text in wrong location Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: addTrack cannot reuse a sender whose transceiver is stopped Bernard Aboba via GitHub
• Re: [webrtc-pc] example 14: render before verifying the remote fingerprint? Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Remove reference from RTPSender to RTCPeerConnection Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Move operations queue definition paragraph to right section (editorial) jan-ivar via GitHub
• Re: [webrtc-pc] enqueue an operation: is executing async? jan-ivar via GitHub
• Closed: [webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event jan-ivar via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub

Wednesday, 3 January 2018

• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Validation of reordered readonly parameters in setParameters Bernard Aboba via GitHub
• Closed: [webrtc-pc] replaceTrack and removeTrack: Synchronous? Bernard Aboba via GitHub
• Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous? Bernard Aboba via GitHub
• Re: [webrtc-pc] RTCCertificate Interface should (or should not) be backed up. Tim Panton via GitHub
• Re: [webrtc-pc] RTCCertificate Interface should (or should not) be backed up. Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Maximum message size slightly incorrect Bernard Aboba via GitHub
• Re: [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection? Bernard Aboba via GitHub
• Re: [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub
• Re: [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection? Philipp Hancke via GitHub
• [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection? Harald Alvestrand via GitHub
• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• Re: [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub
• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• Re: [webrtc-stats] WiFi Stats. Varun Singh via GitHub
• [webrtc-stats] WiFi Stats. karthikbr82 via GitHub
• Re: [webrtc-pc] specify legacy onaddstream? jan-ivar via GitHub
• Re: [webrtc-pc] When to fire events triggered by setRemoteDescription. jan-ivar via GitHub
• Re: [webrtc-stats] packetsDuplicated is not reported in an RTCP Report Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stats for Audio network adaptation Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Stats for Audio network adaptation Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data henbos via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data Varun Singh via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data Harald Alvestrand via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data Harald Alvestrand via GitHub
• Re: [webrtc-stats] Audio/Video sync follow-up Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Audio/Video sync follow-up Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. Harald Alvestrand via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? henbos via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? stefan hakansson via GitHub
• Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous? stefan hakansson via GitHub
• Closed: [webrtc-pc] Consider using generic algorithm stefan hakansson via GitHub
• Re: [webrtc-pc] Consider using generic algorithm stefan hakansson via GitHub
• Re: [webrtc-pc] example 14: render before verifying the remote fingerprint? stefan hakansson via GitHub
• Re: [webrtc-pc] Example 14 never sends any media Harald Alvestrand via GitHub
• [webrtc-pc] enqueue an operation: is executing async? Philipp Hancke via GitHub

Tuesday, 2 January 2018

• Re: [webrtc-pc] example 14: render before verifying the remote fingerprint? Nils Ohlmeier via GitHub
• Re: [webrtc-pc] addTransceiver woes Taylor Brandstetter via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• [webrtc-pc] Pull Request: replaceTrack: Never negotiate when replacing an ended track? Bernard Aboba via GitHub
• Re: [webrtc-pc] replaceTrack: Never negotiate when replacing an ended track? Bernard Aboba via GitHub
• Re: [webrtc-pc] addTransceiver woes henbos via GitHub
• Re: [webrtc-pc] Example 14 never sends any media stefan hakansson via GitHub
• Re: [webrtc-pc] Example 14 never sends any media Harald Alvestrand via GitHub
• Re: [webrtc-pc] replaceTrack: Never negotiate when replacing an ended track? stefan hakansson via GitHub
• Closed: [webrtc-pc] Section 4.3.3: optional methods stefan hakansson via GitHub
• Re: [webrtc-pc] Section 4.3.3: optional methods stefan hakansson via GitHub
• Closed: [webrtc-pc] `parameters` not defined in `setParameters` algorithm stefan hakansson via GitHub
• Re: [webrtc-pc] `parameters` not defined in `setParameters` algorithm stefan hakansson via GitHub
• Re: [webrtc-pc] Consider using generic algorithm stefan hakansson via GitHub
• Re: [webrtc-pc] Make promise rejection/enqueing consistent stefan hakansson via GitHub
• Re: [webrtc-pc] example 14: render before verifying the remote fingerprint? Harald Alvestrand via GitHub
• Re: [webrtc-pc] example 14: render before verifying the remote fingerprint? Harald Alvestrand via GitHub
• Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous? henbos via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? henbos via GitHub
• Re: [webrtc-pc] Processing Remote MediaStreamTracks should pass on receiver when constructing track event stefan hakansson via GitHub
• Re: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions? stefan hakansson via GitHub
• Re: [webrtc-pc] Should rollback fire addtrack/removetrack events? stefan hakansson via GitHub
• [webrtc-pc] example 14: render before verifying the remote fingerprint? Philipp Hancke via GitHub
• Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous? stefan hakansson via GitHub
• [webrtc-pc] Example 14 never sends any media stefan hakansson via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub

Last message date: Wednesday, 31 January 2018 20:50:32 UTC