WebRTC 115 Release Notes

[Harald Alvestrand]

WebRTC M115 is going to be released as part of Chrome M115 , currently planned for release on  May 30th 2023.

We had three PSAs for M115:

• PSA: new format for WebRTC event logs in Chrome

• PSA: libwebrtc now reports spec compliant and internally consistent video de-jitter buffer delays

• PSA: inbound-rtp headerBytesReceived, bytesReceived and packetsReceived now include retransmissions

and the legacy callbased-based getStats is still on the way out and throws an exception 50% of the time in the Beta and Canary channels:

• PSA: Callback-based getStats() is now deprecated

The most important bugfix is this one which fixes a nasty bug showing multiple cursors when capturing the screen on Windows.

The following issues were marked as fixed or verified and had at least one commit in M115 (build, test and trivial code changes are not included):

Issue	Summary	Component
chromium:1444615	[WPT] New failures introduced in external/wpt/webrtc by import https://crrev.com/c/4521919	Blink>WebRTC
webrtc:11013	video is decoded for inactive m-lines	Video,Network>RTP
webrtc:14892	Crash in RTCMTLVideoView when trying to draw an invalid sized frame	Mobile
webrtc:15109	VideoCaptureDS::StopCapture leaves captured webcam with camera light on	Video
chromium:1421656	ScreenShare WebRTC cursor flickering when hover on clickable element	Internals>Media>ScreenCapture
webrtc:15172	RTX padding packets sent before media have no timestamps	Video
chromium:1442604	Chrome 113+ WebRTC Calling - Duplicate data channels in offer if answer disabled it	Blink>WebRTC>DataChannel
webrtc:15094	H264 profile level in iPhone 14	Mobile
webrtc:9674	getStats: mediaType was renamed to kind	Stats
webrtc:14919	Echo leaks at the beginning of calls.	Audio
chromium:1291247	[Remoting host] Tracking bug to support remote desktop on linux/wayland	Services>Chromoting
webrtc:15085	The video implementation of RTCInboundRtpStreamStats.jitterBufferDelay does not match the spec	Stats
webrtc:15096	Implement retransmittedPacketsReceived and retransmittedBytesReceived for inbound-rtp	Stats
webrtc:15106	For AV1, disable error resilience on upper temporal layers.	Video
webrtc:15182	Allow WebRTC to send padding packets with 255 bytes	BWE
webrtc:15177	Implement RtpTransceiver.setCodecPreferences() for Android	PeerConnection
webrtc:15161	Delete support for WebRTC-H264Simulcast/Disabled/	Video
webrtc:14830	Add support for more formats in v4l2 video capture	Video
webrtc:15174	PipeWire screencasting is randomly broken	DesktopCapture
webrtc:14334	Improve IPv6 network resolution and candidate creation	Network>ICE
webrtc:14138	Remove special case CPU overshoot limit for MacOS on low core count
chromium:1439830	WebRTC video send: reduce IPCs.	Blink>WebRTC>Video, Blink>WebRTC>Network
chromium:1435005	Use the worker thread to deliver audio on BreakoutBox	Blink>MediaStream
chromium:1320617	webrtc-internals: request current state of peerconnection for existing connections	Blink>WebRTC>Tools
chromium:1448046	VideoTrackAdapter drops too many frames	Blink>WebRTC>Video
chromium:1448816	WebRTC low bitrate scenario: HW encoder worse than SW in QVGA	Blink>WebRTC>Video
webrtc:15162	Cleanup NonStandardGroupId, StatExposureCriteria and RTCNonStandardStatsMember	Stats
chromium:1382329	Cleanup kAllowCaptureControllerForGetUserMediaScreenCapture	Blink>GetDisplayMedia

For the full list of commits please refer to the git log between this branch and the previous branch. See here for a description of what the release notes contain.

We strongly recommend WebRTC developers to fully test their services in Chrome Beta to ensure stability for end-users.

 

The Chrome release schedule can be found here.

These release notes were prepared by Philipp Hancke.