[webrtc-pc] "a=msid" line should contain sender/receiver IDs, not track IDs
- Taylor Brandstetter via GitHub (Friday, 19 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Iñaki Baz Castillo via GitHub (Friday, 12 January)
- henbos via GitHub (Friday, 12 January)
- Taylor Brandstetter via GitHub (Thursday, 11 January)
- Iñaki Baz Castillo via GitHub (Sunday, 7 January)
- henbos via GitHub (Sunday, 7 January)
- henbos via GitHub (Sunday, 7 January)
- stefan hakansson via GitHub (Saturday, 6 January)
- Iñaki Baz Castillo via GitHub (Friday, 5 January)
- stefan hakansson via GitHub (Friday, 5 January)
- Iñaki Baz Castillo via GitHub (Friday, 5 January)
- Philipp Hancke via GitHub (Thursday, 4 January)
- stefan hakansson via GitHub (Thursday, 4 January)
- henbos via GitHub (Thursday, 4 January)
- Philipp Hancke via GitHub (Thursday, 4 January)
- henbos via GitHub (Thursday, 4 January)
- henbos via GitHub (Thursday, 4 January)
- Philipp Hancke via GitHub (Thursday, 4 January)
- stefan hakansson via GitHub (Thursday, 4 January)
- Philipp Hancke via GitHub (Thursday, 4 January)
- henbos via GitHub (Thursday, 4 January)
- henbos via GitHub (Thursday, 4 January)
[webrtc-pc] "track" event will fire extra times if applying multiple remote offers
- jan-ivar via GitHub (Monday, 22 January)
- stefan hakansson via GitHub (Thursday, 18 January)
- jan-ivar via GitHub (Wednesday, 17 January)
- stefan hakansson via GitHub (Wednesday, 17 January)
- jan-ivar via GitHub (Tuesday, 16 January)
- stefan hakansson via GitHub (Sunday, 14 January)
- Harald Alvestrand via GitHub (Sunday, 14 January)
- Taylor Brandstetter via GitHub (Thursday, 11 January)
- docfaraday via GitHub (Wednesday, 10 January)
- Taylor Brandstetter via GitHub (Wednesday, 10 January)
- docfaraday via GitHub (Wednesday, 10 January)
- Taylor Brandstetter via GitHub (Wednesday, 10 January)
[webrtc-pc] `parameters` not defined in `setParameters` algorithm
[webrtc-pc] Add direction change as reason for "mute" event
[webrtc-pc] Add support for WebRTC Data Channel in Workers
[webrtc-pc] addTransceiver woes
- Harald Alvestrand via GitHub (Thursday, 11 January)
- stefan hakansson via GitHub (Thursday, 4 January)
- stefan hakansson via GitHub (Thursday, 4 January)
- jan-ivar via GitHub (Wednesday, 3 January)
- Philipp Hancke via GitHub (Wednesday, 3 January)
- jan-ivar via GitHub (Wednesday, 3 January)
- jan-ivar via GitHub (Wednesday, 3 January)
- Philipp Hancke via GitHub (Wednesday, 3 January)
- jan-ivar via GitHub (Wednesday, 3 January)
- Taylor Brandstetter via GitHub (Tuesday, 2 January)
- stefan hakansson via GitHub (Tuesday, 2 January)
- henbos via GitHub (Tuesday, 2 January)
- stefan hakansson via GitHub (Tuesday, 2 January)
- stefan hakansson via GitHub (Tuesday, 2 January)
- stefan hakansson via GitHub (Tuesday, 2 January)
[webrtc-pc] At risk text in wrong location
[webrtc-pc] canTrickleIceCandidiates question
- Harald Alvestrand via GitHub (Thursday, 25 January)
- Taylor Brandstetter via GitHub (Friday, 12 January)
- cdh4u via GitHub (Wednesday, 10 January)
- Philipp Hancke via GitHub (Wednesday, 10 January)
- Taylor Brandstetter via GitHub (Tuesday, 9 January)
- Philipp Hancke via GitHub (Tuesday, 9 January)
- cdh4u via GitHub (Tuesday, 9 January)
- Taylor Brandstetter via GitHub (Monday, 8 January)
- Harald Alvestrand via GitHub (Monday, 8 January)
- Harald Alvestrand via GitHub (Monday, 8 January)
- Philipp Hancke via GitHub (Monday, 8 January)
[webrtc-pc] Clarify that "disconnected" is transient.
[webrtc-pc] Codify refusing to generate an empty Offer
[webrtc-pc] Consider using generic algorithm
[webrtc-pc] Data channel closing procedure
[webrtc-pc] Defer SRD add/remove tracks until right before firing track events.
[webrtc-pc] Define pc.getTransceivers() et al to be in insertion order.
[webrtc-pc] Direction is not readonly
[webrtc-pc] Do not consider direction at "need negotiation" evaluation for replaceTrack
[webrtc-pc] Do not consider direction in "need negotiation" evaluation for replaceTrack
[webrtc-pc] editorial: media api introduction
[webrtc-pc] Editorial: PendingRemoteDescription is set by....
[webrtc-pc] Effect of mute/disable on on-the-wire framerate is not described
[webrtc-pc] enqueue an operation: is executing async?
[webrtc-pc] Example 14 never sends any media
[webrtc-pc] example 14: render before verifying the remote fingerprint?
[webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions?
[webrtc-pc] Integrate CSP access control into algorithms
[webrtc-pc] Make promise rejection/enqueing consistent
[webrtc-pc] Maximum message size slightly incorrect
[webrtc-pc] Multiple SRDs may leave streams and tracks in unexpected state.
[webrtc-pc] Need <section> around dictionaries under RTCRtpSender
[webrtc-pc] Need <section> around RTCIceTransport dictionaries
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by alvestrand
- Harald Alvestrand via GitHub (Thursday, 25 January)
- Harald Alvestrand via GitHub (Thursday, 25 January)
- Harald Alvestrand via GitHub (Thursday, 25 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 18 January)
- Harald Alvestrand via GitHub (Thursday, 11 January)
- Harald Alvestrand via GitHub (Thursday, 11 January)
- Harald Alvestrand via GitHub (Thursday, 11 January)
- Harald Alvestrand via GitHub (Thursday, 4 January)
- Harald Alvestrand via GitHub (Thursday, 4 January)
- Harald Alvestrand via GitHub (Thursday, 4 January)
- Harald Alvestrand via GitHub (Thursday, 4 January)
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] new commits pushed by vivienlacourba
[webrtc-pc] onmute then onunmute can fire before negotiation completes
[webrtc-pc] Possibly racy replaceTrack()
[webrtc-pc] Private key access?
[webrtc-pc] Processing Remote MediaStreamTracks should pass on receiver when constructing track event
[webrtc-pc] Pull Request: Add "statslifetimeended" event
[webrtc-pc] Pull Request: addTrack cannot reuse a sender whose transceiver is stopped
[webrtc-pc] Pull Request: Clarify that "disconnected" is transient (editorial)
[webrtc-pc] Pull Request: Convert audioLevel to double.
[webrtc-pc] Pull Request: Defer SRD add/remove tracks until right before firing track events.
[webrtc-pc] Pull Request: Define pc.getTransceivers()) et al to be in insertion order.
[webrtc-pc] Pull Request: Gating 'unmute' events on track with transceiver direction.
[webrtc-pc] Pull Request: Make audioLevel and voiceActivityFlag optional and non-nullable.
[webrtc-pc] Pull Request: Maximum message size slightly incorrect
[webrtc-pc] Pull Request: Move operations queue definition paragraph to right section (editorial)
[webrtc-pc] Pull Request: Need <section> around dictionaries under RTCRtpSender
[webrtc-pc] Pull Request: Passing receiver when constructing track event in 'Processing Remote …
[webrtc-pc] Pull Request: Pending remote updated by setRemote.
[webrtc-pc] Pull Request: Properly defined direction when creating transceiver.
[webrtc-pc] Pull Request: Remove reference from RTPSender to RTCPeerConnection
[webrtc-pc] Pull Request: replaceTrack "negotiation needed" clarification
[webrtc-pc] Pull Request: replaceTrack: Never negotiate when replacing an ended track?
[webrtc-pc] Pull Request: RTCRtpSender: Specify advise about framerate for disabled/muted tracks
[webrtc-pc] Pull Request: Underlying data transport explanation
[webrtc-pc] Pull Request: Upgrade to respec 1.9.0.2
[webrtc-pc] Pull Request: Validation of reordered readonly parameters in setParameters
[webrtc-pc] Pull Request: Whether KeyingMaterial internal slot can be stored and retrieved
[webrtc-pc] replaceTrack "negotiation needed" clarification
[webrtc-pc] replaceTrack and removeTrack: Synchronous?
[webrtc-pc] replaceTrack: Clarify how the UA determines if negotiation is needed
[webrtc-pc] replaceTrack: Never negotiate when replacing an ended track?
[webrtc-pc] Respec issue: Locked to too old version?
[webrtc-pc] RTCCertificate Interface should (or should not) be backed up.
[webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want.
[webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats
- Philip Jägenstedt via GitHub (Saturday, 27 January)
- alex gouaillard via GitHub (Saturday, 27 January)
- Philip Jägenstedt via GitHub (Saturday, 27 January)
- Harald Alvestrand via GitHub (Saturday, 27 January)
- Philip Jägenstedt via GitHub (Friday, 26 January)
- Harald Alvestrand via GitHub (Friday, 26 January)
- Philip Jägenstedt via GitHub (Friday, 26 January)
- Philip Jägenstedt via GitHub (Friday, 26 January)
- Harald Alvestrand via GitHub (Friday, 19 January)
- Philip Jägenstedt via GitHub (Friday, 19 January)
- Taylor Brandstetter via GitHub (Wednesday, 17 January)
- Dominique Hazael-Massieux via GitHub (Monday, 15 January)
- Taylor Brandstetter via GitHub (Friday, 12 January)
- jan-ivar via GitHub (Friday, 12 January)
- na-g via GitHub (Thursday, 11 January)
[webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
[webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type
[webrtc-pc] Section 4.3.3: optional methods
[webrtc-pc] Should rollback fire addtrack/removetrack events?
[webrtc-pc] Should the spec describe addStream/onaddstream as legacy API?
[webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection?
[webrtc-pc] Sort out unmute and BYE
[webrtc-pc] specify legacy onaddstream?
- Philipp Hancke via GitHub (Wednesday, 10 January)
- youennf via GitHub (Wednesday, 10 January)
- youennf via GitHub (Wednesday, 10 January)
- Philip Jägenstedt via GitHub (Wednesday, 10 January)
- henbos via GitHub (Wednesday, 10 January)
- youennf via GitHub (Wednesday, 10 January)
- Philip Jägenstedt via GitHub (Wednesday, 10 January)
- henbos via GitHub (Wednesday, 10 January)
- Philipp Hancke via GitHub (Wednesday, 10 January)
- Harald Alvestrand via GitHub (Wednesday, 10 January)
- Philip Jägenstedt via GitHub (Wednesday, 10 January)
- Harald Alvestrand via GitHub (Wednesday, 10 January)
- jan-ivar via GitHub (Wednesday, 3 January)
[webrtc-pc] Terminology around "setting" attributes may be incorrect
[webrtc-pc] Upgrade to respec 1.9.0.2
[webrtc-pc] Using setConfiguration() to add application certificates to an RTCPeerConnection post-construction?
[webrtc-pc] WebRTC bypass CSP connect-src policies
[webrtc-pc] WHATWG streams for data channel messages
- Lennart Grahl via GitHub (Wednesday, 31 January)
- Lennart Grahl via GitHub (Wednesday, 31 January)
- stefan hakansson via GitHub (Wednesday, 31 January)
- Anne van Kesteren via GitHub (Wednesday, 31 January)
- Lennart Grahl via GitHub (Wednesday, 31 January)
- stefan hakansson via GitHub (Wednesday, 31 January)
- Lennart Grahl via GitHub (Wednesday, 31 January)
- Anne van Kesteren via GitHub (Wednesday, 31 January)
- stefan hakansson via GitHub (Wednesday, 31 January)
- Adam Rice via GitHub (Wednesday, 31 January)
- Lennart Grahl via GitHub (Tuesday, 30 January)
- Harald Alvestrand via GitHub (Tuesday, 30 January)
- Lennart Grahl via GitHub (Friday, 26 January)
- Anne van Kesteren via GitHub (Friday, 26 January)
- Harald Alvestrand via GitHub (Thursday, 25 January)
- Lennart Grahl via GitHub (Thursday, 25 January)
- Bernard Aboba via GitHub (Wednesday, 24 January)
- Lennart Grahl via GitHub (Wednesday, 17 January)
- Domenic Denicola via GitHub (Wednesday, 17 January)
- Lennart Grahl via GitHub (Wednesday, 17 January)
- Bernard Aboba via GitHub (Wednesday, 17 January)
- Adam Rice via GitHub (Wednesday, 17 January)
- Lennart Grahl via GitHub (Saturday, 13 January)
[webrtc-pc] WHATWG streams for data channels
[webrtc-pc] When to fire events triggered by setRemoteDescription.
[webrtc-pc] Whether KeyingMaterial internal slot can be stored and retrieved
[webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
[webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type
[webrtc-stats] Add "sender" and "receiver" stats.
[webrtc-stats] Add fecPacketsSent/Received
[webrtc-stats] Add per layer stats for SVC
[webrtc-stats] Add stat for inputAudioLevel, before the audio filter
[webrtc-stats] Add stat to reflect the redundancy of FEC/RED data
[webrtc-stats] Additional description of audioLevel
[webrtc-stats] Adds per-DSCP packet counters to RTP streams
[webrtc-stats] Audio/Video sync follow-up
[webrtc-stats] averageRTCPInterval should be averageRtcpInterval
[webrtc-stats] Cleanup: reformat the sections
[webrtc-stats] CodecType "encode" / "decode" needs to default to "both"
[webrtc-stats] CodecType "encode" / "decude" needs to default to "both"
[webrtc-stats] Define object lifetimes and mention end-of-life emission
[webrtc-stats] Define timestamps in terms of performance.origin
[webrtc-stats] Deleting framesCorrupted
[webrtc-stats] Enable continuous publication
[webrtc-stats] fix camel case averageRtcpInterval and also put in RTCInboundRTPStreamStats
[webrtc-stats] Fixes #161
[webrtc-stats] Introduction to RTP stream statistics
[webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats?
[webrtc-stats] Is keeping stats around a memory problem?
- jan-ivar via GitHub (Wednesday, 24 January)
- Harald Alvestrand via GitHub (Wednesday, 24 January)
- Harald Alvestrand via GitHub (Wednesday, 24 January)
- jan-ivar via GitHub (Wednesday, 24 January)
- Harald Alvestrand via GitHub (Wednesday, 24 January)
- henbos via GitHub (Thursday, 18 January)
- jan-ivar via GitHub (Wednesday, 17 January)
- henbos via GitHub (Wednesday, 17 January)
- Varun Singh via GitHub (Wednesday, 17 January)
- Harald Alvestrand via GitHub (Wednesday, 17 January)
- jan-ivar via GitHub (Thursday, 11 January)
- Varun Singh via GitHub (Thursday, 11 January)
- Lennart Grahl via GitHub (Thursday, 11 January)
- Harald Alvestrand via GitHub (Thursday, 11 January)
[webrtc-stats] Lifetime of RTPStreamStats
[webrtc-stats] Missing definition of "deleted". When exactly do stats get deleted?
[webrtc-stats] moved bytes and packets received counters
[webrtc-stats] Need DSCP information for outgoing RTP streams
[webrtc-stats] new commits pushed by alvestrand
[webrtc-stats] new commits pushed by vivienlacourba
[webrtc-stats] new commits pushed by vr000m
[webrtc-stats] packetsDuplicated is not reported in an RTCP Report
[webrtc-stats] Pivot from "track" to "sender" and "receiver" stats.
[webrtc-stats] Pull Request: Adds per-DSCP packet counters to RTP streams
[webrtc-stats] Pull Request: Define object lifetimes and mention end-of-life emission
[webrtc-stats] Pull Request: FEC sent and received metrics
[webrtc-stats] Pull Request: fix camel case averageRtcpInterval and also put in RTCInboundRTPStreamStats
[webrtc-stats] Pull Request: fix for timestamp for remote and local objects
[webrtc-stats] Pull Request: Fixes #161
[webrtc-stats] Pull Request: Introduction to RTP stream statistics
[webrtc-stats] Pull Request: moved packetsDuplicated to the right dictionary
[webrtc-stats] Pull Request: tidy checks, missed in latest PRs.
[webrtc-stats] Remove reference to RTCRtpSynchronizationSource.audioLevel.
[webrtc-stats] Remove references to MSE. Fixes #161
[webrtc-stats] Rename "objectDeleted" to something else.
[webrtc-stats] Rename RTCRTPStreamStats to RTCRtpStreamStats?
[webrtc-stats] Rename sender/receiver/track stats
[webrtc-stats] RTCCodecStats in every getStats() is redundant.
[webrtc-stats] RTCCodecStats in getStats() is redundant.
[webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one
[webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification
[webrtc-stats] RTCRTPStreamStats should have senderId/receiverId
[webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.
[webrtc-stats] Stats for Audio network adaptation
[webrtc-stats] Stats need to mark up which members are required
[webrtc-stats] terminology: rename mediaType to kind
[webrtc-stats] Typo - RTCLocalAudioStreamTrackStats should inherit from RTCAudioSenderStats
[webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries
[webrtc-stats] We need "sender" and "receiver" stats, not "track" stats
[webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated?
- Taylor Brandstetter via GitHub (Wednesday, 31 January)
- Varun Singh via GitHub (Wednesday, 31 January)
- Varun Singh via GitHub (Tuesday, 30 January)
- Taylor Brandstetter via GitHub (Tuesday, 30 January)
- jan-ivar via GitHub (Monday, 29 January)
- Taylor Brandstetter via GitHub (Monday, 29 January)
- jan-ivar via GitHub (Monday, 29 January)
- jan-ivar via GitHub (Monday, 29 January)
- Taylor Brandstetter via GitHub (Monday, 29 January)
- Harald Alvestrand via GitHub (Monday, 29 January)
- Varun Singh via GitHub (Sunday, 28 January)
- Taylor Brandstetter via GitHub (Friday, 26 January)
[webrtc-stats] WiFi Stats.
[webrtc-stats] Work through implications of simulcast on the receiver side
Closed: [webrtc-pc] "track" event will fire extra times if applying multiple remote offers
Closed: [webrtc-pc] `parameters` not defined in `setParameters` algorithm
Closed: [webrtc-pc] addTrack can reuse a sender whose transceiver is stopped
Closed: [webrtc-pc] At risk text in wrong location
Closed: [webrtc-pc] canTrickleIceCandidiates question
Closed: [webrtc-pc] Clarify that "disconnected" is transient.
Closed: [webrtc-pc] Consider using generic algorithm
Closed: [webrtc-pc] Effect of mute/disable on on-the-wire framerate is not described
Closed: [webrtc-pc] How to handle removing and re-adding remote streams/tracks - possible ID collisions?
Closed: [webrtc-pc] Maximum message size slightly incorrect
Closed: [webrtc-pc] order of transceivers, senders/receivers?
Closed: [webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event
Closed: [webrtc-pc] Processing Remote MediaStreamTracks should pass on receiver when constructing track event
Closed: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
Closed: [webrtc-pc] Respec issue: Locked to too old version?
Closed: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing
Closed: [webrtc-pc] RTCDtlsTransportState enum descriptions are lacking
Closed: [webrtc-pc] RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats
Closed: [webrtc-pc] Section 4.3.3: optional methods
Closed: [webrtc-pc] Should we require a reference from RTPSender to RTCPeerConnection?
Closed: [webrtc-pc] specify legacy onaddstream?
Closed: [webrtc-pc] When to fire events triggered by setRemoteDescription.
Closed: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
Closed: [webrtc-stats] Add fecPacketsSent/Received
Closed: [webrtc-stats] Audio/Video sync follow-up
Closed: [webrtc-stats] averageRTCPInterval should be averageRtcpInterval
Closed: [webrtc-stats] CodecType "encode" / "decode" needs to default to "both"
Closed: [webrtc-stats] Definitions from MSE need re-targeting
Closed: [webrtc-stats] Deleting framesCorrupted
Closed: [webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings?
Closed: [webrtc-stats] Enable continuous publication
Closed: [webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats?
Closed: [webrtc-stats] Need DSCP information for incoming RTP streams
Closed: [webrtc-stats] Need DSCP information for outgoing RTP streams
Closed: [webrtc-stats] packetsDuplicated is not reported in an RTCP Report
Closed: [webrtc-stats] Rename "objectDeleted" to something else.
Closed: [webrtc-stats] RTCCodecStats in every getStats() is redundant.
Closed: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one
Closed: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification
Closed: [webrtc-stats] Stats for Audio network adaptation
Closed: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats
Closed: [webrtc-stats] What happens when a partial keyFrames is received?
Closed: [webrtc-stats] When are "fractionLost", "packetsLost, " "jitter", and other RFC3550-based stats updated?
Last message date: Wednesday, 31 January 2018 20:50:32 UTC