WebRTC M123 is going to be released as part of Chrome M123, currently planned for release on March 19, 2024.
We had two PSAs for M123:
Please also prepare for this upcoming PSA for M124:
SDES, which we agreed upon as “MUST NOT implement” at the IETF in Berlin in 2013, is finally gone after being unavailable in Chrome for a while. Kudos to the Twilio team for catching a related regression (issue 326493639) before it reached Chrome Stable!
If you notice any broken links, in particular due to the recent Chromium issue tracker migration please let us know.
The following issues were marked as fixed or verified and had at least one commit in M123 (build, test and trivial code changes are not included):
Issue | Summary | Component |
webrtc:15057 | Deprecate special-casing of TWCCv2 negotiation | PeerConnection |
webrtc:15703 | Backfill codec-specific default parameters | PeerConnection |
chromium:1520859 | Cloned received RTCEncodedAudioFrames lose sequenceNumber | Blink>WebRTC
>PeerConnection |
webrtc:15791 | Pass Environment to VideoDecoder at construction | Video |
webrtc:15835 | NetEq::Config.min/max delay ignored | Audio |
webrtc:11066 | Remove SDES (key exchange in SDP) | Network>DTLS |
webrtc:15791 | Pass Environment to VideoDecoder at construction | Video |
webrtc:15211 | Unable to share same camera with two web pages in Firefox when using PipeWire camera backend | Video |
webrtc:15759 | REMB missing when at first time receiving video | PeerConnection |
webrtc:15845 | Duplicate msid lines don't cause SDP rejection | Network |
webrtc:15164 | Replace use of RTCStatsMember<T> with absl::optional<T> instead | Stats |
chromium:326493639 | createAnswer fails with "Failed to create transport answer, transport is missing" | Blink>WebRTC |
chromium:1318448 | Deprecate and remove mediaConstraints from RTCPeerConnection (except DtlsSrtpKeyAgreement on Fuchsia) | Blink>WebRTC
>PeerConnection |
chromium:1508163 | Add temporal layer decoding support to AV1Decoder | Internals>Media>Hardware |
chromium:1503097 | AudioContext needs to pass the info about AudioDestination::FramesPerBuffer() during MediaStreamSource creation | Blink>WebAudio |
chromium:1520618 | blink::StatsCollector wraps a weak persistent on the wrong thread | Blink>WebRTC
,Blink>GarbageCollection |
chromium:1517949 | mp4: Inconsistent return for isTypeSupported and canPlay. | Blink>MediaRecording |
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.
