WebRTC M119 Release Notes

[Harald Alvestrand]

WebRTC M119 is going to be released as part of Chrome M119, currently planned for release on October 31st 2023 (M118 was pretty short)

We had two PSAs for M119:

• PSA regarding CVE-2023-5217: Heap buffer overflow in vp8 encoding in libvpx

• PSA: libWebRTC won't send RTCP BYE when disabling a layer (M119)

Also we have an upcoming deprecation:

• PSA: WebRtcAllowLegacyTLSProtocols enterprise policy to be removed from Chrome M121

and please test the DTLS changes which we plan to experiment with in M120:

• PSA: DTLS client hello extension permutation will be enabled soon, do interop testing now

The following issues were marked as fixed or verified and had at least one commit in M119 (build, test and trivial code changes are not included):

Issue	Summary	Component
webrtc:11082	Setting RTCRtpSender encodingParameters.active to false causes RTCP-BYE (spec violation)	Network>RTP
webrtc:14392	Add feature to probe at NetworkStateEstimate if available	BWE
webrtc:15410	Remove deprecated versions of SendRtp and SendRtcp	Internals
webrtc:14782	BUNDLE does not check for header extension id collisions	PeerConnection
webrtc:15496	EncoderBitrateAdjuster does not read min bitrates from EncoderInfo	Video
webrtc:15471	CHECK crash when remote description recycles m-line with already used mid and different kind	PeerConnection
webrtc:15464	Improvements for Pipewire campera capture	Video
webrtc:14154	Field trial policy	Documentation
webrtc:15508	WebRTC hits absl hardening check when timestamp is negative	Video
webrtc:15415	VideoCaptureImpl::IncomingFrame only checks ConvertToI420 result for invalid param, not all errors	Video
webrtc:12197	webrtc can be tricked into sending rtp with a payload type reserved for RTCP	PeerConnection
webrtc:15530	VP8 rate controller sometimes dropping too many frames	Video
webrtc:15377	issue about congestion window when send data one-way	BWE
webrtc:15477	Rare RTC_CHECK firing in RtpPacketizerH264::NextAggregatePacket	Video
webrtc:15499	CreateOffer/CreateAnswer should return RTCError instead of nullptrs	PeerConnection
chromium:1478172	[ThumbnailCapturerMac] Stable ordering	Blink>GetDisplayMedia
webrtc:15178	SRTP GCM ciphers are not enabled by default in native code	PeerConnection
webrtc:14600	dcSCTP enters deferred stream reset processing mode and never exits	DataChannel
chromium:1487223	RTCEncodedVideoFrameMetadata::rtpTimestamp is not set the the RTCEncodedFrameSetMetadata OT	Blink>WebRTC >PeerConnection
chromium:1423413	Improve MediaRecorder standards compliance.	Blink >MediaRecording
chromium:416876	WebRTC-internals: Keep peerConnections in the HTML table after they are closed/garbage collected	Blink>WebRTC >Tools
chromium:1480383	Add "focus-capturing-application" CaptureStartFocusBehavior	Blink >GetDisplayMedia
chromium:1481448	Refactor the frame drop UMAs on top of the MediaStreamTrack Stats API	Blink>MediaStream
chromium:1477706	Implement and ship the DisplayMediaStreamOptions monitorTypeSurfaces	Blink >GetDisplayMedia
chromium:1375217	tracking bug for webrtc-internals ui/ux improvements	Blink>WebRTC >Tools
chromium:1411164	Enable WebRtcEncoderAsyncEncode	Blink>WebRTC >Video
chromium:1484453	Prioritize displaying last gUM/gDM requests in chrome://webrtc-internals	Blink>WebRTC >Tools
chromium:1481501	getUserMedia: requesting an unprocessed stream and then a processed yields two unprocessed streams on ChromeOS	Blink >GetUserMedia
chromium:1482684	[webrtc-internals] Fix bug where computed metrics are marked "(removed)"	Blink>WebRTC >Tools

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.