PSA: WebRTC M72 Release Notes

[Niklas Blum]

WebRTC M72 Release Notes

WebRTC M72 branch (cut at r25798)

Summary

WebRTC M72, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains the three milestones features: Unified Plan SDP, support for getDisplayMedia and improved privacy protection using mDNS (still behind flag), in addition to the numerous enhancements of the getUserMedia and RTCPeerConnection API’s. As with previous releases, 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!

The Chrome release schedule can be found here. Native libraries for Android and iOS are built on a weekly basis and are available on JCenter and CocoaPods; the Changelog is available here.

Features

Unified Plan enabled by default (Stable Rollout)

Unified Plan is the spec-complaint SDP format, it is being rolled out together with the RTCRtpTransceiver APIs that are heavily tied to the standard format. This is a big milestone for cross-browser compatibility, but also a breaking change for applications with “Plan B” assumptions - see PSA and the “Unified Plan” Transition Guide (JavaScript). Experiments making Unified Plan the default in Canary, Dev and Beta channels are ongoing and the plan is to roll this out to 100% in M72 Stable.

getDisplayMedia() enabled by default

Screen Capture API is now available for WebRTC in Chrome. It is possible to request access for acquiring user’s display as a video track without going through extensions based APIs. Current implementation produces the same UI and permissions flow as the extensions for an easy migration. Users are asked for permission in each request as their answer does not persist. Note that extensions-based screen capture APIs are still available and getDisplayMedia() does not have audio-capture support yet.

resizeMode constrainable property

The resizeMode constrainable property allows control of the resolution-adjustment mechanism for video tracks. If the property is false, getUserMedia() will use only native resolutions without adjustment for the video tracks it returns. If the property is true, cropping and downscaling may be used to satisfy other resolution constraints such as width, height and aspectRatio.

Private IP Concealment via mDNS

In line with this IETF spec IP host addresses are replaced with generic multicast DNS names for local ICE candidates. This improvement limits the amount of unique information exposed by browsers, while maintaining the ability to establish P2P connection within local networks. This feature is only available behind a flag and can be tested by setting chrome://flags/#enable-webrtc-hide-local-ips-with-mdns.

Deprecations

Deprecating webrtc::HighPassFilter and aecm_generate_comfort_noise option

The mobile aec (AECM) will not produce comfort noise any more, and the high pass filter in the audio processing module has changed interface for toggling on/off. See PSA for details.

Platform	Issue	Description	Component
Chrome	8572	Remove the old AudioProcessing statistics interface	Audio
Chrome	8995	Delete RTPPayloadRegistry	Network>RTP
Chrome	7882	Remove audio call duration UMA metric WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds	Audio, Stats

Features and Bugfixes

Type	Issue	Description	Component
Feature	9898	Potential high render delays with bursty video.	Video
Feature	10014	AEC3: Clockdrift detection	Audio
Feature	6458	Use codec rate when generating RTCP for audio	Audio
Feature	9866	Expose parameters for configuring packet loss rate set in Opus encoder	Audio
Feature	9904	fake_network_pipe counts only buffer overflows as loss.	BWE
Feature	9796	Add unacknowledged but allocated traffic to bandwidth estimate.	BWE
Feature	10068	Adaptive loss-backoff thresholds	BWE
Feature	9681	Integrate Per Frame Encryption Interface Into WebRTC	Network
Feature	9923	It should be possible to enable the CongestionWindowPushbackController without a field trial	Network
Feature	9168	Support VPN adapter type in WebRTC	Network>ICE
Feature	8789	Make RTCP interval configurable.	Network>RTP
Feature	9989	Add RTCRtpReceiver::getParameters() RTCP and simulcast support	PeerConnection
Feature	9795	Enable Video Encryption Using the FrameEncryptorInterface	Video
Feature	9969	Enable target bitrate RTCP XR message from field trial	Video
Feature	9376	Add VP9 Profile 2 to supported codecs	Video
Feature	9950	Don't buffer encoded frames if full superframe drop mode is enabled	Video
Feature	9768	Set frame duration per spatial layer	Video
Feature	9953	Send Nack with a delay after packets out of order is detected on packet receive	Video
Feature	8762	Make WavReader tolerant towards optional Wave chunks	Audio
Feature	7990	Add support for RTP two-byte header extensions	Network>RTP
Feature	10053	Add PeerConnection option to configure minimum audio jitter buffer delay	Audio
Feature	10043	Expose metric for delayed packet outage in getStats	Audio
Feature	854980	Add support for the resizeMode MediaStreamTrack constraint	Blink>GetUserMedia
Bug	9967	AEC3: Comfort noise is too low	Audio
Bug	9906	Usage of the dominant nearend functionality during the initial phase may cause echo	Audio
Bug	9879	The adaptive filter in AEC3 looses all information at echo path delay changes	Audio
Bug	9998	The custom suppressor behavior in AEC3 during stationary render noise causes echo leakage	Audio
Bug	895435	During call startup, AEC3 transparency is lower than later in the call	Blink>Media>Audio
Bug	895082	Timeout in neteq_signal_fuzzer	Blink>WebRTC>Audio
Bug	895431	The AEC3 echo suppressor is to conservative during saturated echo	Blink>WebRTC>Audio
Bug	896334	AEC3: Faster delay detection is needed	Blink>WebRTC>Audio
Bug	905291	The custom suppressor behavior in AEC3 during stationary render noise causes echo leakage	Blink>WebRTC>Audio
Bug	900951	Better metrics for Unified Plan breakages	Blink>WebRTC>PeerConnection
Bug	906029	Add metrics to find out how far PCs come in negotiation before failing	Blink>WebRTC>PeerConnection
Bug	906988	RtcStats doesn't report remote data properly with unified plan	Blink>WebRTC>Network
Bug	9807	Make ice credentials same for createOffer/createAnswer and pooled port allocator session.	Network>ICE
Bug	9964	Race condition between mDNS name registration and cricket::Port::SignalPortComplete	Network>ICE
Bug	9861	Hide IP addresses obfuscated by .local names in stats	Network>ICE
Bug	9868	Specify single packet max size explicitly when splitting frame into packets	Network>RTP
Bug	9880	Support "a=ssrc:... msid:..." line with no stream ID	PeerConnection
Bug	9907	Fix FrameEncryptor being deallocated too early.	Audio
Bug	9865	AEC3: Avoid fading when render signal is stationary at call init.	Audio
Bug	9961	AEC3: Filter performance mismatch between direct path and reverberant paths.	Audio
Bug	9726	AEC3: Delay estimator delays the capture signal	Audio
Bug	9897	AEC3: Dominant nearend detection can be further utilized for transparency	Audio
Bug	9912	The parsing and output of the AEC3 parameter for using dominant nearend is missing	Audio
Bug	9921	Parsing of json file specifying the APM behavior fails silently	Audio
Bug	9894	Some elements of the AEC3 config struct are not parsed by the json parser	Audio
Bug	9843	The AEC3 echo reverb modelling is tailored to unreasonable large rooms	Audio
Bug	9835	During call startup, AEC3 transparency is lower than later in the call	Audio
Bug	9929	Clear locks from AEC2 and AECM audio processing submodules	Audio
Bug	10010	Data Race in channel_send.cc	Audio
Bug	9885	RTP header extensions not logged in RTC event log.	Audio, Tools
Bug	901711	virtual/webrtc-wpt-unified-plan/external/wpt/webrtc/simplecall-no-ssrcs.https.html failing on chromium.mac/Mac10.13 Tests (dbg)	Blink>WebRTC
Bug	894439	AEC3: Avoid ducking when render signal is stationary at call startup	Blink>WebRTC>Audio
Bug	9942	sqrt of negative number in aimd_rate_control.cc	BWE
Bug	7510	VideoSendStreamImpl::OnBitrateUpdated seems wrong when sent a bitrate of 0	BWE, Video
Bug	9917	Utility to create hkdf	Network
Bug	9926	RtpSender And RtpReceiver should not set a FrameEncryptor/FrameDecryptor in a stopped state.	Network
Bug	9891	Move CryptoOptions to RTCConfiguration	Network
Bug	9970	GetRtpReceiver/SenderCapabilities() return multiple RTX codec entries	PeerConnection
Bug	9935	peerconnection_client|server doesn't support field trials.	SampleApps
Bug	9123	Can not see "Local Preview" window in video_loopback. Pressing Enter does nothing on Windows.	SampleApps, Video
Bug	9597	Add support for RtpEncodingParameters max_framerate.	Video
Bug	9873	AEC3: ERLE might not be updated during reverberation	Audio
Bug	9932	Prevent SetChannel On Stopped Receiver	PeerConnection
Bug	10067	FlexFEC causes retransmit bitrate increase.	Video
Bug	866390	Echo cancellation cannot be disabled on Android	Blink>GetUserMedia>Mic, Blink>WebRTC>Audio

[Philipp Hancke]

I find it rather interesting that despite the suggested restrictions or tabsharing  in the specification that is now included by default and with virtually no changes to the picker (other than the default being "abort")

Can you explain the security rationale?

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[Tommi]

Can you clarify what exact bug/feature you're referring to? We can look for the specific conversations related to that and give an update (assuming the information is missing from the bug or not viewable for some reason).

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[Philipp Hancke]

tab sharing, the third option in the window picker. The spec is pretty clear on doing a bit more effort for those:

  https://w3c.github.io/mediacapture-screen-share/#elevated-permissions

(I don't think getDisplayMedia current supports the nice feature of chooseDesktopMedia(["tab], ...) to only show tabs either)

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