WebRTC 116 Release Notes

[Harald Alvestrand]

WebRTC M116 is going to be released as part of Chrome M116, currently planned for release on August 15th 2023.

We had two PSAs for M116:

• RTP header id collisions are rejected during setRemoteDescription in M116+

• PSA: stricter checking of SSRCs in the SDP

Be careful with your ids, they are supposed to be unique identifiers.

The legacy callbased-based getStats is still on the way out (and going to be removed by default in M117) and throws an exception 50% of the time in the Beta and Canary channels:

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

The following issues were marked as fixed or verified and had at least one commit in M116 (build, test and trivial code changes are not included):

Issue	Summary	Component
webrtc:14205	Echo leaks when early reverberation occurs before the strongest echo	Audio
webrtc:15085	The video implementation of RTCInboundRtpStreamStats.jitterBufferDelay does not match the spec	Stats
webrtc:15241	Add missing definition for StatsReport::Value::id_val	Stats
chromium:1463451	Cloned or sender RTCEncodedVideoFrames fed into a decoder break playback delay buffering	Blink>WebRTC >PeerConnection
chromium:1291247	[Remoting host] Tracking bug to support remote desktop on linux/wayland	Services>Chromoting
webrtc:15203	Use a constant for invalid PipeWire file descriptor	DesktopCapture
webrtc:15182	Allow WebRTC to send padding packets with 255 bytes	BWE
webrtc:14985	TemporalLayers/PictureIdTest.IncreasingAfterRecreateStream* tests are flaking	Video,Network
chromium:1455428	WebRTC + MediaFoundation = RtpPayloadParams::Vp9ToGeneric produces wrong metadata resulting in 30 keyframes/s	Blink>WebRTC >Video
webrtc:15258	DataChannel hanging in "connecting" state if createOffer used before createAnswer	DataChannel
webrtc:14227	Need a process for release notes	Documentation
webrtc:14244	Implement RTCInboundRtpStreamStats.jitterBufferTargetDelay for video streams	Stats
chromium:1453226	Transformable Encoded Audio frames should always inherit from TransformableAudioFrameInterface regardless of direction	Blink>WebRTC >PeerConnection
webrtc:15162	Cleanup NonStandardGroupId, StatExposureCriteria and RTCNonStandardStatsMember	Stats
webrtc:15185	Missing RTP packet info after parsing RED packets.	Audio
webrtc:15245	Consider deleting RTCStats enum values	Stats
webrtc:15198	Delete "track_id" stats member temporarily added to assist a roll	Stats
webrtc:14884	Support VP9 simulcast	Video
webrtc:15202	PipeWire camera - split portal and PipeWire implementations	Video
webrtc:14175	Deprecate and remove RTCMediaStreamTrackStats	Stats
webrtc:15240	Allow setting a custom randomness source	Internals
webrtc:12194	the range of dynamic rtp payload types is exhausted	PeerConnection
webrtc:15257	use RGB565 camera on windows maybe crash
chromium:1455847	serialize RTCDataChannelInit	Blink>WebRTC>Tools
chromium:1447318	Add "Copy video frame" to video element's context menu	Internals>Media>UI
webrtc:14175	Deprecate and remove RTCMediaStreamTrackStats	Stats
chromium:1451036	MediaRecorder throws with mimeType video/webm and audio-only stream	Blink >MediaRecording
chromium:1443983	Cleanup WebRtcStatsReportIdl experiment	Blink>WebRTC >PeerConnection
chromium:1450496	Confusing DOMException message mentioning "setFocusBehavior()"	Blink >GetDisplayMedia
chromium:1429996	MediaStreamAudioTrackUnderlyingSource::OnData allocates memory on RT thread
chromium:1448816	WebRTC low bitrate scenario: HW encoder worse than SW in QVGA	Blink>WebRTC >Video
chromium:1453226	Transformable Encoded Audio frames should always inherit from TransformableAudioFrameInterface regardless of direction	Blink>WebRTC >PeerConnection
chromium:1451358	Encoded transforms for video have an unexpected dependency on SSRC	Blink>WebRTC >PeerConnection
chromium:1448046	VideoTrackAdapter drops too many frames	Blink>WebRTC >Video
chromium:1375217	tracking bug for webrtc-internals ui/ux improvements	Blink>WebRTC>Tools
chromium:1450844	structuredClone() on RTCEncodedVideoFrame and RTCEncodedAudioFrame only performs shallow copy	Blink>WebRTC >PeerConnection
chromium:1454398	[WebRTC] Add flag to control if forcing SW should include or exclude 360p	Blink>WebRTC >Video

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.