24d251f796
Some of the tests are currently flaky because FEC is disabled if the RTT is <200 ms, and the simulated network is configured to use 100 ms for the send transport, but nothing is configured for the receive transport. This CL configures the receive transport to 100 ms delay. Bug: webrtc:10920 Change-Id: I79995693ba73683406fa9ced92a7918e6c05473f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154571 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29394}
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