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If RED is no longer used the send codec needs to be reconfigured. To test on https://webrtc.github.io/samples/src/content/peerconnection/audio/ run: await pc1.setLocalDescription(); await pc1.setRemoteDescription({type: 'answer', sdp: pc1.remoteDescription.sdp.replace('red/48000', 'blue/48000')}) As a result, RED will be turned off and the bitrate will drop. BUG=webrtc:11640 Change-Id: Icc7a83ae29e67d054399bf42010264e94c32127d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221360 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34719}
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