8c16f745ab
1. Pass media transport factory to JSEP transport controller. 2. Pass media transport to voice media channel. 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 Bug: webrtc:9719 Reviewed-on: https://webrtc-review.googlesource.com/c/105542 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Commit-Queue: Anton Sukhanov <sukhanov@google.com> Cr-Commit-Position: refs/heads/master@{#25152}
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