| commit | 3d7db263b902f319b5b0c828ccff59cd8664b937 | [log] [tgz] |
|---|---|---|
| author | mflodman <mflodman@webrtc.org> | Fri Apr 29 07:57:13 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Apr 29 07:57:21 2016 |
| tree | 4a150dd46b505c4609f46afb7a5b968d169077b2 | |
| parent | 8f8c96d1924c96f3a5c149ca31883d296ddceca7 [diff] |
Switch voice transport to use Call and Stream instead of VoENetwork. VoENetwork is kept for now, but is not really used anylonger. webrtcvoiceengine is changed to have the same behavior for unsignaled ssrc as video has, which is reflected by disabling one test case and this will be discussed and followed up. BUG=webrtc:5079 TBR=tommi Review-Url: https://codereview.webrtc.org/1909333002 Cr-Commit-Position: refs/heads/master@{#12555}
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.