d9a51b05da
Also updating SocketOptionsMergedOnSetTransport test code to make the call to SetRtpTransport from the right context. Bug: webrtc:12636 Change-Id: I343851bcf8ac663d7559128d12447a9a742786f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213660 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33633}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs