M124 Release Notes

[Harald Alvestrand]

WebRTC M124 is going to be released as part of Chrome M124, currently planned for release on April 16, 2024.

We had one PSAs for M124:

• PSA: setCodecPreferences requires codecs from the RTCRtpReceiver.getCapabilities (M124)

and an upcoming one for M125:

• PSA: Flexible mode in VP9 RTP payload descriptor

One of the highlights of this release is that the Chromium DevTools network emulation finally affects WebRTC connections!

If you notice any broken links, in particular due to the recent Chromium issue tracker migration please let us know.

The following issues were marked as fixed or verified and had at least one commit in M124 (build, test and trivial code changes are not included):

Issue	Summary	Component
webrtc:15852	VP9 simulcast default bitrates are too high	Video
webrtc:15703	Backfill codec-specific default parameters	PeerConnection
webrtc:15847	Clarify the different ways to match a codec	PeerConnection
webrtc:15828	Disable multithreading for QVGA in libvpx VP8 encoder on Android	Video
webrtc:14169	rtc_base/task_queue.* is problematic	Internals
webrtc:15866	Enable TCP Keep Alive socket options to rtc::Socket	Network
webrtc:15791	Pass Environment to VideoDecoder at construction	Video
webrtc:15852	VP9 simulcast default bitrates are too high	Video
webrtc:15530	VP8 rate controller sometimes dropping too many frames	Video
chromium:40276865	Screensharing does not stop when terminated from GNOME tray
chromium:41485050	WebRCT receive buffer size too small for high frame rate devices
chromium:41175925	DevTools: Device Mode Emulation network throttling doesn't throttle WebRTC
chromium:40072842	WebRTC DataChannel messages are not sent when and bufferedAmount is 0 and it is closed
chromium:41221056	Closing WebRTC data channels is broken and only works chrome to chrome
chromium:326493639	createAnswer fails with "Failed to create transport answer, transport is missing"
chromium:331346276	Chromoting broken by change 342520
chromium:40889961	MediaRecorder reports incorrect AVC codec level in ondataavailable callback
chromium:1354626	implement relayProtocol and url properties on RTCIceCandidate object
chromium:40790593,40277804,41490235	webrtc insertable streams: too large audio frames are silently dropped

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.

[Philipp Hancke]

Also note that there are some DTLS changes which caused a breakage, see

  https://github.com/sipsorcery-org/sipsorcery/issues/1104

for details.

--
This list falls under the WebRTC Code of Conduct - https://webrtc.org/support/code-of-conduct.
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