a22c2a0c58
as this breaks RTCP assumptions about SSRCs being no longer active as defined in https://www.rfc-editor.org/rfc/rfc3550#section-6.6 This should not be sent in reaction to temporarily disabling a stream via RTCRtpParameters.active as this does not mean that the participant is leaving the session as defined in https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7 and does not indicate end of participation as defined in https://www.rfc-editor.org/rfc/rfc3550#section-6.1 which stipulates BYE should be the last packet sent from this SSRC. BUG=webrtc:11082 Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#38059}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation