cf6e24a12d
In current implementation, the DtlsSrtpTransport listens to the SignalNetworkRouteChanged but doesn't forward it to the BaseChannel which makes it impossible for the media engine to update the network route and the transport overhead. The BaseChannel unit tests failed to catch this issue because it used a plain unencrypted RTP transport for testing. This CL fix that issue and update the BaseChannel tests. Bug: webrtc:7013, b/73645191 Change-Id: I417b58ff9af4e3c4fac442ff10b5a85bc2093530 Reviewed-on: https://webrtc-review.googlesource.com/55940 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22140}
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