WebRTC 117 release notes

[Harald Alvestrand]

WebRTC M117 is going to be released as part of Chrome M117, currently planned for release on September 12th 2023.

In terms of features this release adds the RTP header extension control API as well as two new RTP statistics.

We had a couple of PSAs:

• PSA: WebRTC now mixes all audio receive streams

• PSA: Removal of WebRTC getStats encoderImplementation "unknown" and dataChannelIdentifier "-1"

• PSA: Extra usages of video-content-type RTP extension are gone

• PSA: webrtc::Transport::SendRtp is changing its interface

The legacy callbased-based getStats is going away (finally):

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

   

The following issues were marked as fixed or verified and had at least one commit in M117 (build, test and trivial code changes are not included):

webrtc:10234	Always send abs-send-time when negotiated and do not filter it out.
webrtc:15071	Do not expose dataChannelIdentifier stat -1	Stats
webrtc:14906	stats: default decoderImplementation "unknown" is not useful	Stats
chromium:1456628	Remove RtpHeader from TransformableAudioFrameInterface	Blink>WebRTC> PeerConnection
chromium:1446655	Constant choppy audio can be observed when there are multiple WebRTC in-bound audio streams	Blink>WebRTC> Audio
webrtc:15212	TimeUTCMicros is limited to millisecond resolution on Windows	Stats
webrtc:15381	The RTCInboundRtpStreamStats.contentType field does not always accurately reflect screenshare status	Stats
webrtc:15250	stats: implement fecBytesReceived	Stats
webrtc:15258	DataChannel hanging in "connecting" state if createOffer used before createAnswer	DataChannel
webrtc:15383	Stop using video-content-type to carry experiments and simulcast info.	Video,Network
webrtc:15096	Implement retransmittedPacketsReceived and retransmittedBytesReceived for inbound-rtp	Stats
webrtc:15390	x-google-max-bitrate only cares about encoding maxBitrate if encodings.size() == 1	Video
webrtc:15363	Add an additional API to desktop capturer to request screen and window sharing simultaneously	DesktopCapture
chromium:1470261	Sending stereo audio with Opus Red crashes the tab	Blink>WebRTC
chromium:1426440	getStats producing misleading scalabilityMode values when encoderImplementation changes	Blink>WebRTC> Video
chromium:1455962	Multiple encodings but only first is active + scalabilityMode is specified = maxBitrate is ignored	Blink>WebRTC> PeerConnection
chromium:1463451	Cloned or sender RTCEncodedVideoFrames fed into a decoder break playback delay buffering	Blink>WebRTC> PeerConnection
chromium:1448816	WebRTC low bitrate scenario: HW encoder worse than SW in QVGA	Blink>WebRTC> Video
chromium:1467461	webrtc-internals: Metrics going stale is very misleading (e.g. invalid scalabilityMode)	Blink>WebRTC> Tools
chromium:1456628	Remove RtpHeader from TransformableAudioFrameInterface	Blink>WebRTC> PeerConnection
chromium:1469318	No H264 encoder on Android due to SW fallback at <360p resolutions	Blink>WebRTC> Video
chromium:1414363	getStats(): Define stats objects in WebIDL	Blink>WebRTC> PeerConnection
webrtc:14522	Unship non-standard video "track" metrics	Stats
webrtc:15162	Cleanup NonStandardGroupId, StatExposureCriteria and RTCNonStandardStatsMember	Stats
chromium:1375217	tracking bug for webrtc-internals ui/ux improvements	Blink>WebRTC> Tools
chromium:1462567	webrtc-internals JSON dump should include the timestamp	Blink>WebRTC> Tools
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.