97ba853295
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated. I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored. Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841 Bug: webrtc:14802 Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38980}
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