WebRTC 110 Release notes

[Philipp Hancke]

WebRTC M110 is going to be released as part of Chrome M110 on February 1st 2023.

Note that the deprecated track and stream statistics are going to be removed soon (but not in M110).

See the intent to deprecate thread on blink-dev which should be shared on this list soon as well.

We also had a couple of PSAs which may require action from developers:

• PSA: Extra validation for TURN and STUN URLs in Chromium M110

• PSA: number of TURN servers is limited to 32 as of Chrome M110

• PSA: mids longer than 16 characters will be rejected as of M110 (originally M98)

• PSA: iSAC support will be removed in M110

This list of PSAs may still be incomplete, this will improve in future releases.

On Linux, the pipewire-based screen capturer is now enabled by default which means less completely black screen shares.

The following issues were marked as fixed or verified and had at least one commit in M110 (build, test and trivial code changes are not included):

Issue	Summary	Component
chromium:1381982	Deadlock in WebRTC metronome	Blink>WebRTC
chromium:1369050	Only expose decoderImplementation if "HW exposure is allowed"	Blink>WebRTC> PeerConnection
webrtc:10635	Implement stats for packet send-side delay for audio streams	Stats,Audio
webrtc:14522	Unship non-standard video "track" metrics	Stats
webrtc:13960	scalabilityMode support with VP9	PeerConnection,Video
chromium:1374310	Tab crashes after ICE restart when using TCP TURN	Blink>WebRTC
webrtc:14628	"outbound-rtp.active: false" observed while actively sending VP9 SVC	Stats
webrtc:14593	totalPacketSendDelay is incorrectly implemented	Stats
webrtc:14521	Move non-standard pause and freeze from track to inbound-rtp and make them standard	Stats
webrtc:5773	Bringing AppRTCDemo to the background and back to foreground can cause BWE to get stuck at low levels.	Video
webrtc:14608	Delete disable_ipv6 flag	PeerConnection
chromium:1374436	Improve Capturer Selection Logic on Wayland	Internals>Media> ScreenCapture
webrtc:14632	StreamInterface::Read/Write should use rtc::ArrayView	Internals
webrtc:14547	Peer connection's `getStats` by RtpSender/Receiver is not available in Android SDK	Mobile
webrtc:14627	Simplify PeerConnectionE2EQualityTestFixture::AddPeer signature	PeerConnection
webrtc:14587	Add googTimingFrameInfo to modern stats API	Stats
webrtc:14596	RTCTransportStats.dtlsCipher not available when building with OpenSSL	Stats
chromium:1379802	TrackControls::requested is redundant with TrackControls::stream_type	Blink> GetDisplayMedia
chromium:1375235	RTCVideoEncoder error handling inconsistency	Blink>WebRTC>Video
chromium:1215480	Conditional Focus - Tracking Bug	Blink> GetDisplayMedia
chromium:1379243	getDisplayMedia({video: {frameRate: {max: 0}}}) should fail with OverconstrainedError.	Blink> GetDisplayMedia
chromium:1103280	Audio issues with insertable streams enabled under load	Blink>WebRTC
chromium:1378667	Implement suppressLocalAudioPlayback	Blink> GetDisplayMedia
chromium:1381982	Deadlock in WebRTC metronome	Blink>WebRTC
chromium:1381265	Correct default value for selfBrowserSurface	Blink> GetDisplayMedia

For the full list of commits please refer to the git log between this branch and the previous branch.

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