PSA: WebRTC M98 Release Notes

[Huib Kleinhout]

WebRTC M98 Release Notes

Branch: WebRTC M98 branch

Summary

WebRTC M98, currently available in Chrome's stable channel, contains 3 new features and over 15 bug fixes, enhancements and stability/performance improvements. We encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found. Please take a look at this page, for some pointers on how to file a good bug report. The help we have received has been invaluable!

Note that the WebRTC release notes only cover WebRTC specific changes. Build, test and trivial code changes are not included. Follow the Chromium and Chrome releases blog for further updates on important changes in Chrome releases.

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

Highlights

Add a “Share this tab instead” button to the tab sharing ribbon for getDisplayMedia

When sharing a tab using the chrome.desktopCapture extension API, a button to “Share this tab instead” is displayed in the tab sharing ribbon. This change will enable this button also when sharing a tab using the getDisplayMedia web API.

Region Capture - API for Cropping Self-Capture Video Tracks

Region Capture enables the possibility to crop MediaStreamTracks produced by a call to getDisplayMedia. It allows applications to efficiently display shared content and non-shared content in one page. More information can be found here https://github.com/w3c/mediacapture-region

Features and Bugfixes

Type	Issue	Description	Component
Bug	http://crbug.com/1271469	Reintroduce SDES aka DtlsSrtpKeyAgreement=False for limited usage	PeerConnection
Bug	http://crbug.com/1272566	Received MediaStreamTracks not attached to audio elements are still played out	Blink>WebRTC
Bug	http://bugs.webrtc.org/11489	Frequent locking in ReceiveStatisticsProxy	Internals
Bug	http://bugs.webrtc.org/12713	NACK: erase an unreceived packet because of uncorrect packet ssrc	Network>RTP
Bug	http://bugs.webrtc.org/12905	RTCVideoSourceStats.framesPerSecond stat has wrong type	Stats
Bug	http://bugs.webrtc.org/12973	msan - SIGSEGV StatsEndToEndTest.VerifyNackStats	Video
Bug	http://bugs.webrtc.org/13287	setCodecPreferences changes where rtx codecs appear in the SDP	PeerConnection
Bug	http://bugs.webrtc.org/13346	FrameBuffer::ReturnReason::kStopped is unused	Video
Bug	http://bugs.webrtc.org/13354	RenegotiateManyVideoTransceiversAndWatchAudioDelay failed on Linux	PeerConnection
Bug	http://bugs.webrtc.org/13374	SDP integration fuzzer is broken	PeerConnection
Feature	http://bugs.webrtc.org/13377	Implement RTCOutboundRtpStreamStats.targetBitrate for audio	Audio,Stats
Bug	http://bugs.webrtc.org/13392	expose relayProtocol on peer-reflexive ice candidate stats	Network>ICE
Bug	http://bugs.webrtc.org/13444	NetEq::Create still in header file, but removed from source	Audio
Bug	http://bugs.webrtc.org/13469	ConfigureSvcNormalVideo does not consider frame orientation when choosing number of spatial layers	Video
Bug	http://bugs.webrtc.org/7063	Complete and spec-compliant RTCIceCandidateStats	Stats

Deprecations

Platform	Type	Issue	Description	Component
webrtc	Bug	http://bugs.webrtc.org/12875	Delete VideoEncoder::EncoderInfo::has_internal_source	Video

[helloj...@gmail.com]

Is there test support for SDES?

At present, I use the following code, 

const constraints = {
mandatory: {},
optional: [{ DtlsSrtpKeyAgreement: false }]
};
const pc = new RTCPeerConnection(configuration, constraints);

the crypto line is not found in the sdp generated by the pc, but the fingerprint line can be found,

My test sdes failed, my chrome version is "Version 100.0.4896.127 (Official Build) (x86_64)"

[Philipp Hancke]

There is not apart from generating an offer and checking for the presence of crypto lines.

This is an API you were not supposed to use for 5+ years...

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[Harald Alvestrand]

There is no support for SDES in Chrome.