Andreas Pehrson via GitHub
Balázs Kreith via GitHub
- [webrtc-stats] Add field `fecPacketsSent` to RtcOutboundRtpStreamStats (#692)
- [webrtc-stats] Summary table does not list `media-playout` stats (#691)
- [webrtc-stats] Duplicated field `kind` in RTCInboundRtpStreamStats (#690)
- [webrtc-stats] Pull Request: Fix descriptions according to the upgraded draft
Ben Wagner via GitHub
Bernard Aboba
- Re: [EXTERNAL] RTCSender still transmits audio packet even when a=inactive is received in SDP
- TPAC 2022 Schedule
Bernard Aboba via GitHub
- [webrtc-stats] Adding SVC-related stats fields (#673)
- [webrtc-nv-use-cases] Pull Request: Add requirements for low latency streaming use case
Byoungchan Lee via GitHub
Chris Needham (W3C Calendar)
- Event Updated: Web Real-Time Communications Working Group and Media Working Group and Media and Entertainment Interest Group - Joint meeting
- Event Updated: Web Real-Time Communications Working Group and Media Working Group and Media and Entertainment Interest Group - Joint meeting
- Event Updated: Web Real-Time Communications Working Group and Media Working Group and Media and Entertainment Interest Group - Joint meeting
- Event Updated: Web Real-Time Communications Working Group and Media Working Group and Media and Entertainment Interest Group - Joint meeting
Cullen Jennings via GitHub
docfaraday via GitHub
Dominique Hazael-Massieux
- Re: Fwd: VOTE by 2022-09-21/22: Proposed Charter for the Web Real-Time Communications Working Group
- [minutes] TPAC meetings
- Re: Fwd: VOTE by 2022-09-21/22: Proposed Charter for the Web Real-Time Communications Working Group
Dominique Hazael-Massieux via GitHub
- [webrtc-stats] Pull Request: Add media-playout to summary table
- [webrtc-pc] Pull Request: Update to ReSpec version 32.2.4
- [mediacapture-main] Pull Request: Update to latest ReSpec version 32.2.4
- [webrtc-pc] Pull Request: Update link to WebSockets interface
- [mediacapture-record] Broken references in MediaStream Recording (#215)
Elad Alon via GitHub
- [mediacapture-screen-share] Pull Request: s/controller = null;/controller;
- [mediacapture-screen-share] Pull Request: Add CaptureController to the spec
- [mediacapture-screen-share] Inconsistent formatting in doc (#234)
- [mediacapture-handle] Should the handle be an object? (#68)
- [mediacapture-main] Device enumeration spec-ed to hang (#903)
- [mediacapture-region] Behavior when cropTo races with the track ending (#72)
- [mediacapture-region] Rejecting cropTo() on ended tracks (#71)
- [mediacapture-screen-share] Spec appears to require gDM leak info on availability of audio (#233)
Florent Castelli via GitHub
- [webrtc-extensions] RTCDataChannel transfer and maxMessageSize (#114)
- [webrtc-pc] Pull Request: Add RTCPeerConnection.getDataChannels()
- [webrtc-svc] S modes and a single active simulcast layer (#73)
- [webrtc-pc] Pull Request: Remove references to RTCIceCredentialType
Francois Daoust
François Beaufort via GitHub
henbos via GitHub
- [webrtc-stats] Privacy concern: Leaking communication / plain text using patterns in packet size, frequency, etc. (#699)
- [webrtc-stats] Exposing HW for Cloud Gaming use cases (HW encoder/decoder revisited) (#698)
- [webrtc-stats] Pull Request: r/"Privacy considerations"/"Procedures for mitigating privacy concerns"
- [webrtc-stats] Rename "Privacy considerations" to "Procedures for mitigating privacy concerns" (#696)
- [webrtc-stats] Pull Request: Add droppedSamplesEvents and synthesizedSamplesEvents
- [webrtc-stats] Pull Request: Fix recently introduced typos.
- [webrtc-stats] Pull Request: Add totalCaptureDelay and totalSamplesCaptured to RTCAudioSourceStats.
- [webrtc-stats] Pull Request: Add droppedSamplesDuration to RTCAudioSourceStats.
- [webrtc-stats] Pull Request: Add totalPlayoutDelay and totalSamplesCount to RTCAudioPlayoutStats.
- [webrtc-stats] Pull Request: Add RTCAudioPlayoutStats, synthesizedSamplesDuration and totalSamplesDuration
- [webrtc-stats] Metrics for capture delay (#681)
- [webrtc-stats] Metrics for playout delay (#680)
- [webrtc-stats] How many times did capture glitches occur? (Follow-up to #678) (#679)
- [webrtc-stats] Need metrics for capture glitches (#678)
- [webrtc-stats] How many times did glitches occur? (Follow-up to #676) (#677)
- [webrtc-stats] Need metrics for playout glitches (#676)
- [webrtc-stats] The stats API allow hardware fingerprinting (encoder, powerEfficient) (#675)
- [webrtc-stats] Codec stats reveal hardware information which could be used for fingerprinting (#674)
- [webrtc-stats] Pull Request: Delete RTP stream stats when ssrc or codec changes
- [webrtc-stats] Pull Request: Make RTP creation dependent on SSRC being known rather than packet being sent
- [webrtc-stats] Pull Request: Adds powerEfficientDecoder/powerEfficientEncoder.
- [webrtc-stats] Pull Request: Only expose RTCCodecStats objects currently in use
- [webrtc-stats] When are RTP streams destroyed? (#668)
- [webrtc-stats] When are RTP stream stats created? (#667)
- [webrtc-stats] powerEfficientEncoder/powerEfficientDecoder (#666)
- [webrtc-pc] Is "same PT, different FMTP lines" allowed in BUNDLE? (#2766)
- [webrtc-stats] Is "same PT, different FMTP lines" allowed in BUNDLE? (#664)
Jan-Ivar Bruaroey via GitHub
- [mediacapture-screen-share] Pull Request: No need to test for presence of member with default value.
- [webrtc-pc] Pull Request: TypeError on duplicate rids
- [webrtc-pc] Pull Request: TypeError unless all or none of encodings have rids
- [webrtc-extensions] Need to specify behavior of detached RTCDataChannel objects. (#115)
- [webrtc-pc] Pull Request: Scaleaudio
- [mediacapture-screen-share] Spec should mention clickjacking concerns (#231)
Jeremy Soong via GitHub
Michele BRYANT
Ms2ger via GitHub
- [mediacapture-output] Spec uses settings object's responsible document which was removed (#132)
- [mediacapture-screen-share] "current settings object's relevant global object" is not defined (#232)
Philipp Hancke
Philipp Hancke via GitHub
- [webrtc-encoded-transform] Pull Request: describe (some) fields
- [webrtc-encoded-transform] Pull Request: audio: add rtp sequence number on incoming frames
- [webrtc-encoded-transform] Pull Request: idl: synchronizationsource is a *unsigned* long
- [webrtc-extensions] Pull Request: header extension API: remove setParameters support
- [webrtc-stats] Pull Request: editorial: remove trailing whitespace
- [webrtc-stats] Pull Request: inbound-rtp: add frame assembly time
- [webrtc-pc] Pull Request: move url from RTCIceEvent to the RTCIceCandidate
- [webrtc-encoded-transform] Pull Request: intro: remove funny hats
- [webrtc-encoded-transform] expose rid (and mid) as metadata on outgoing frames (#147)
Rajarshee Dhar (rajadhar)
Sergio Garcia Murillo via GitHub
sysbot+ipp@w3.org
Varun Singh via GitHub
W3C Webmaster via GitHub API
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)
webrtc@bytedance via GitHub
- [webrtc-extensions] enable opus bite rate control by js api instead of SDP munging (#117)
- [webrtc-stats] add video totalFreezesDuration &totalPausesDuration etc to standard getStats (#695)
- [webrtc-extensions] add AES-256 media stream encryption control to peerconnection (#113)
youenn fablet
youennf via GitHub
- [webrtc-encoded-transform] Pull Request: Structure clone frame with passing frame.data as transferables and send frame's clone to packetizer or decoder.
- [webrtc-encoded-transform] Pull Request: Remove unused RTCInsertableStreams dictionary
- [webrtc-encoded-transform] Enqueuing a RTCEncodedVideoFrame/RTCEncodedAudioFrame should transfer the data (#150)
- [webrtc-encoded-transform] Pull Request: Make use of SFrame WG spec link
- [webrtc-encoded-transform] Pull Request: Make RTCRtpSender.generateKeyFrame take a single optional rid parameter.