M120 Release Notes

[Harald Alvestrand]

WebRTC M120 is going to be released as part of Chrome M120, currently planned for release on December 5th 2023.

We had three PSAs for M120:

• PSA: Async DNS resolver API in WebRTC is changing

• PSA: libWebRTC will have SRTP GCM ciphers enabled by default (M120)

• PSA: DTLS client hello extension permutation will be enabled soon, do interop testing now (feedback looks good so expect some testing in the wild soon)

and still have an upcoming deprecation for M121:

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

Chrome 120 also adds support for entering picture-in-picture mode automatically when the user switches the tab focus:

https://developer.chrome.com/blog/automatic-picture-in-picture/

The following issues were marked as fixed or verified and had at least one commit in M120 (build, test and trivial code changes are not included):

Issue	Summary	Component
webrtc:15580	Add support for min value in SampleCounter	Stats
webrtc:12497	TURN ports should be restricted to not allow unbridled port numbers	Network
webrtc:14600	dcSCTP enters deferred stream reset processing mode and never exits	DataChannel
chromium:1430806	Add UMA to estimate how often WebRTC audio mixer mixers more than 3 streams	Blink>WebRTC>Audio
webrtc:15532	webrtc::PacedPacketInfo::send_bitrate_bps	Network
webrtc:15471	CHECK crash when remote description recycles m-line with already used mid and different kind	PeerConnection
webrtc:15579	Add mimeType to insertable streams metadata	PeerConnection
chromium:1354491	Make the TaskQueuePacedSender bursty.	Blink>WebRTC >PeerConnection
webrtc:15178	SRTP GCM ciphers are not enabled by default in native code	PeerConnection
webrtc:7101	Implement "quick mode" for webrtc_perf_tests	Video
webrtc:15544	Redundant permission (portal) dialogs while screen sharing on Wayland	DesktopCapture
webrtc:14154	Field trial policy	Documentation
chromium:1485906	h265_bitstream_parser_fuzzer: Integer-overflow in webrtc::H265SpsParser::ParseSpsInternal	Blink>WebRTC
chromium:1477192	Encoded frames fed to an audio RTCRtpSender without a track never get sent	Blink>WebRTC >PeerConnection
webrtc:15556	webrtc does not build on musl: int64_t is undeclared	Build
webrtc:11341	Implement injectable video encoder selctor.	Video
webrtc:15529	outbound-rtp rtxSsrc is the same for all three simulcast streams	Stats
webrtc:15530	VP8 rate controller sometimes dropping too many frames	Video
chromium:1480178	codec_descriptions support in the H264Encoder, VEAEncoder, AudioTrackMojoEncoder	Blink>Media>WebCodecs
chromium:1373032	Split device and display captures into separate queues in UserMediaClient	Blink>GetDisplayMedia
webrtc:14175	Deprecate and remove RTCMediaStreamTrackStats	Stats
chromium:1481448	Refactor the frame drop UMAs on top of the MediaStreamTrack Stats API	Blink>MediaStream
chromium:1440075	Delete in-capturer scaling code paths	Internals>Media >CameraCapture
chromium:1472978	Implement & ship MediaStreamTrack video frame stats API	Blink>MediaStream

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.

[Arun Raghavan]

On Thu, 23 Nov 2023, at 5:02 AM, 'Harald Alvestrand' via discuss-webrtc wrote:
> WebRTC M120
> <https://webrtc.googlesource.com/src.git/+log/refs/branch-heads/6099>

Is there a way to access this branch from outside of Google? The branch does not seem to be exposed on the gitiles server in any way that I could find.

Thanks,
Arun

[Arun Raghavan]

Answering this based on what I found, as there was also a post about this 8 months ago without a response.

There are two ways to do this:

1. Use depot_tools to fetch the webrtc repo, it automatically populates a bunch of the branch heads

2. Look up the git commit id from https://chromiumdash.appspot.com/branches for the branch you care about, and then do a `git fetch <commit-id>` from the webrtc git repo

-- Arun

[Harald Alvestrand]

For just getting the info: Click the link.

I don't know what the gitites server does or does not do, but I was able to open https://webrtc.googlesource.com/src.git/+log/refs/branch-heads/6099 in an incognito window, so I assume it's open to all.

For getting the branch info: Fetch the repo, and sync - if you're using native git commands, you may have to specify --with-branch-heads.

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