1061686107
by not encoding redundancy. The timestamp gap of 400ms means a rtp timestamp difference of 19200 which would overflow the 14 bit RED timestamp difference field. To test in Chrome, go to https://webrtc.github.io/samples/src/content/peerconnection/audio/ set `useDtx = true` in the console and be very quiet. BUG=webrtc:13182,webrtc:11640 Change-Id: I1cedc7d846ac7ae821bb7cb5c0f37a17511ac727 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231940 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35005}
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