Issue  | Summary  | Component  | 
webrtc:11547  | High DataChannel.send() timing  | DataChannel  | 
webrtc:14859  | Ensure ReceiveSideCongestionController use modern types  | Cleanup,BWE  | 
chromium:1420245  | Remove TransformableVideoFrame::GetMetadata in favour of TransformableVideoFrame::Metadata()  | Blink>WebRTC  | 
webrtc:15097  | AV1 default maximum bitrate limit changed  | Video  | 
webrtc:11547  | High DataChannel.send() timing  | DataChannel  | 
chromium:1430806  | Add UMA to estimate how often WebRTC audio mixer mixers more than 3 streams  | Blink>WebRTC>Audio  | 
webrtc:14830  | Add support for more formats in v4l2 video capture  | Video  | 
webrtc:15059  | The target bitrate was mistakenly set to be the maximal bitrate when initializing the libaom encoder  | Video  | 
webrtc:14334  | Improve IPv6 network resolution and candidate creation  | Network>ICE  | 
webrtc:15087  | PipeWire based video capture doesn't set unique ID for the current device  | Video  | 
webrtc:14029  | Make sure audio level 127 in the audio level extension header, represents digital silence  | 
  | 
webrtc:4711  | Improved WebRTC self- and TCP-fairness.  | Video  | 
chromium:1385735  | Fix validation of TURN and STUN URLs in RTCPeerConnection  | Blink>WebRTC> PeerConnection  | 
webrtc:13427  | WebRTC Android's hardware encoder should handle the stride and slice-height of the input buffer.  | Video,Mobile  | 
webrtc:15033  | Exercise inactive streams paths  | Video  | 
webrtc:15015  | Get frame average QP from MediaCodec encoder  | Video  | 
webrtc:15030  | VideoAdapter has an integer overflow scaling very large frames  | Video  | 
webrtc:15098  | DegradedCall deadlocks if network thread = worker thread  | PeerConnection  | 
webrtc:15106  | For AV1, disable error resilience on upper temporal layers.  | Video  | 
chromium:1428098  | VP9 {L3T3_KEY,inactive,inactive} not sending correctly in M113  | Blink>WebRTC>Video  | 
webrtc:4299  | Remove a=ice-options:google-ice from generated offers  | PeerConnection  | 
webrtc:10739  | Add support for the abs-capture-time header extension.  | Network>RTP  | 
webrtc:11547  | High DataChannel.send() timing  | DataChannel  | 
webrtc:14801  | Add the dependency descriptor to the RTC event log.  | Video  | 
chromium:1428006  | Encoded Insertable Streams calls UnregisterEncoded{Audio,Video}StreamCallback on incorrect thread for transform errors in a worker  | Blink>WebRTC> PeerConnection  | 
chromium:1242842  | Camera capture incorrectly reports frame rate in MediaTrackSettings  | Blink>GetUserMedia  | 
chromium:1375217  | tracking bug for webrtc-internals ui/ux improvements  | Blink>WebRTC>Tools  | 
chromium:1427193  | Unstable fps when using MediaStream for video (Starting from Chrome v111)  | Blink>MediaStream  | 
chromium:1375217  | tracking bug for webrtc-internals ui/ux improvements  | Blink>WebRTC>Tools  | 
chromium:1382005  | Deprecate the RegionCaptureExperimentalSubtypes feature flag  | Blink>GetDisplayMedia> RegionCapture  | 
chromium:864871  | Implement standalone RTCIceTransport API  | Blink>WebRTC  | 
chromium:1423413  | Improve MediaRecorder standards compliance.  | Blink>MediaRecording  | 
chromium:1248479  | webrtc insertable streams: too large audio frames are silently dropped  | Blink>WebRTC> PeerConnection  | 
chromium:1429272  | When using insertable streams, getDisplayMedia video track `applyConstraints({framerate: { min: 1 }})` not working as expected  | Blink>GetUserMedia, Blink>GetDisplayMedia  | 
chromium:1414363  | getStats(): Define stats objects in WebIDL  | Blink>WebRTC> PeerConnection  |