PSA: WebRTC M89 Release Notes

[Huib Kleinhout]

WebRTC M89 Release Notes

WebRTC 4389 branch (cut at r7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5)

Summary

WebRTC M89, currently available in Chrome's beta channel, contains over 39 bug fixes, enhancements and stability/performance improvements. 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.

PSAs

WebRTC's Plan B SDP semantics will be deprecated and removed

WebRTC 1.0 is now a W3C Recommendation (press release) with the standard SDP format, Unified Plan, being supported by all major browsers. This year it is time to deprecate and remove the non-standard Chromium SDP format, Plan B (intent thread). Timeline:

• M89 (Stable in February, 2021): Deprecation warning is added in the developer console.

• M93 (Stable in August, 2021): Plan B is removed, with the option to extend this deadline by opting-in to a Reverse Origin Trial.

• M96 (Stable in January, 2022): The extended deadline ends and Plan B is removed for everybody.

In M88 and M89 performance improvements significantly reduce the CPU usage and invocation time of WebRTC negotiation methods which greatly benefits Unified Plan usage.

Usage of rtp payload types in the range [35-65] in webrtc.org/chrome

The range 96-127 of dynamic RTP payload types is exhausted. This requires special considerations for interoperability with older versions of Chrome. Read the PSA for details

a=extmap-allow-mixed will be offered by default

The SDP attribute extmap-allow-mixed has been supported since Chrome M71. However, the SDP negotiation in Chrome versions before M71 will fail if extmap-allow-mixed is offered. From Chrome M89

extmap-allow-mixed will be offered by default [3]. PSA

Deprecations

Platform	Issue	Description	Component
webrtc	6471	Delete the class RTPFragmentationHeader	Video

Features and Bugfixes

Chrome

Type	Issue	Description	Component
Feature	1146942	Upgrade pipewire from 0.2 to 0.3 for chromium/webrtc	Internals>Media>ScreenCapture
Bug	1152841	Browser hangs occasionally when closing share target picker	Internals>Media>ScreenCapture
Bug	1155459	Default STUN attribute length limit is too small	Blink>WebRTC>Network
Bug	943975	Surface max message size in RTCSctpTransport	Blink>WebRTC>PeerConnection
Feature	10439	Provide common interface for bitstream parsers	Video
Feature	10480	Improve RNN VAD efficiency and code quality	Audio
Bug	10675	support for logging raw rtp in text2pcap format	Network>RTP
Feature	10897	Add TURN_LOGGING_ID	Network>ICE,PeerConnection
Bug	11266	Outdated information about working with branches	Documentation
Bug	11767	[Stats] Reduce the number of blocking-invokes from 2 to 1.	Stats
Feature	12111	VoipVolumeControl interface for VoIP API	Audio
Bug	12148	AV1 active decode target mask is not set properly	Video
Bug	12167	AV1 packetizer sets mark bit on each spatial layer	Network>RTP
Bug	12181	transportId is missing from RTCCodecStats	Stats
Bug	12185	Incorrect range of GetLinearAecOutput output	Audio
Feature	12193	VoIP API result types and enforcement	Audio
Bug	12194	the range of dynamic rtp payload types is exhausted	PeerConnection
Bug	12204	fix broken video_replay threading	Tools
Bug	12215	SetLocalDescription/SetRemoteDescription calling CreateSessionDescription thrice.	PeerConnection
Bug	12216	Enable initial frame drop for one active simulcast stream	Video
Bug	12217	Robotic audio heard when the connection is using TLSv1.2	Audio,Network
Bug	12238	RTCPeerConnection Create function should return an error code	PeerConnection
Bug	12261	[Adaptation] Allow simulating being CPU limited for TestBed	Video
Bug	12265	AEC3: Linear filters can gradually diverge in long calls	Audio
Bug	12274	Libvpx VP9 codec wrapper is hard to test	Video
Bug	12297	VideoReceiveStream2: remove unneeded PostTask	Perf
Bug	12314	Organize ilbc #includes per style guide	Audio
Bug	12323	JsepSessionDescription::Clone() does not copy ICE candidates	PeerConnection
Bug	8133	OPUS stereo audio over RTP is muxed to mono	Audio
Bug	9424	SrtpTransport::OnWritableState incorrectly calculates writability	PeerConnection

[aaba...@gmail.com]

As to Plan B deprecation, it's very sad that Issue 10208 is still  not  fixed.  The Unified Plan Chrome implementation still doesn't allow using of several (two) bundles in a RTCPeerConnection, so audio and video streams couldn't be separated by distinctive transport streams, as it works in Plan B for years.

https://bugs.chromium.org/p/webrtc/issues/detail?id=10208

[Philipp Hancke]

that only works in plan-b because you're not bundling at all then (but do demultiplex based on ssrc)?

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[aaba...@gmail.com]

I'm bundling them using two ICE candidate groups. Without BUNDLE keyword that has no effect in Plan B.