| commit | c3d4b48e7e87a3d0ed0a6444cd7f15fa6ef622c7 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Tue May 23 06:07:11 2017 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Tue May 23 13:07:11 2017 +0000 |
| tree | baaf5180f02ad90cb7c2935a939f9f9be6a1dbe6 | |
| parent | 23ac8b49f4f8f5cbf709ddefcb2c721bab2505de [diff] |
Store/restore RTP state for audio streams with same SSRC within a call This functionality already exists for video streams, so not having it for audio is unexpected and has lead to problems. BUG=webrtc:7631 Review-Url: https://codereview.webrtc.org/2887733002 Cr-Commit-Position: refs/heads/master@{#18231}
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.