| commit | 7a43d253f99f3dce7123cdabb7c99a7985dbb021 | [log] [tgz] |
|---|---|---|
| author | skvlad <skvlad@webrtc.org> | Tue Mar 22 22:32:27 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Mar 22 22:32:31 2016 |
| tree | 22a0ff3302cba33ab5c8f0ffd25951fb0b3f3a52 | |
| parent | 01bcbd0df647dff61c4e209444e5cc5f5b6dddc1 [diff] |
Make the audio channel communicate network state changes to the call. This change enables voice-only calls to keep track of the network state. This is only a partial fix - the last modality to change state controls the state for the entire call, so a call with a failed video transport will also stop sending audio packets. Handling this condition correctly would require the call to keep track of network state for each media type separately, and take care of conditions such as a failed video channel getting removed, while a functioning audio channel remains. BUG=webrtc:5307 Review URL: https://codereview.webrtc.org/1757683002 Cr-Commit-Position: refs/heads/master@{#12093}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.