6c87a67b63
With this change, when max-bundle and rtcp-mux are both enabled, we no longer create and destroy a temporary transport channel when a media channel gets added. Instead, the media channel uses the correct bundled transport channel from the start. This fixes a bug where adding a media type would cause the ICE state to briefly become Disconnected and then immediately recover. The temporary channel was created in a non-writable state, which caused the TransportController to declare the ICE state to be Disconnected (as not all transport channels were writable). Right after creation, the temporary channel was then destroyed and the ICE state went back to the correct one. BUG=webrtc:5856 Review-Url: https://codereview.webrtc.org/1972493002 Cr-Commit-Position: refs/heads/master@{#12781}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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