| commit | e5c4a810e21e74acec09c987f20fb0e6788062fd | [log] [tgz] |
|---|---|---|
| author | sprang <sprang@webrtc.org> | Tue Jul 11 10:44:17 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Jul 11 10:44:17 2017 |
| tree | 71bdb883c3afa3c5614b79e99911c4a514acc2f6 | |
| parent | 2910357621dee4368bd3eaa0040cec82ac230dad [diff] |
Move RTP keep-alive config from VideoSendStream::Config to Call::Config This makes more sense since logically it's a transport level feature, not a media stream feature. Even if the implementation details forces it to be an rtp stream detail, for the moment. BUG=webrtc:7907 Review-Url: https://codereview.webrtc.org/2978503002 Cr-Commit-Position: refs/heads/master@{#18963}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.