914351de5c
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1) Original cl description: Always offer transport sequence number header extension for audio If the extension is negotiated, it will only be used if the field trial WebRTC-Audio-SendSideBwe is enabled. This allows simpler experimentation if it should be used or not. Patchset 3 contain the only change: Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc TBR: srte@webrtc.org,ossu@webrtc.org Bug: webrtc:10309 webrtc:10286 Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f Reviewed-on: https://webrtc-review.googlesource.com/c/123183 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26706}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct