| commit | b56671e0517d4de773faeaa645af40840e863e79 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Sat May 27 01:40:05 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Sat May 27 01:40:05 2017 |
| tree | 1b7b3ce701bf7d821f4d0e1ea387c2543d3da884 | |
| parent | 1f3fa0843bdb69be57afefa08b59e7ea2a74b42b [diff] |
Fix issue with send-side bandwidth estimation over TURN TCP connections. AsyncStunTCPSocket wasn't firing SignalSentPacket, which the bandwidth estimator requires for every packet in order to look up send times when feedback arrives. If the signal isn't fired, it always assumes feedback is arriving extremely late, and decreases the bandwidth by a factor of 2 until it reaches the minimum of 10kbps. BUG=webrtc:7717 TBR=pthatcher@webrtc.org Review-Url: https://codereview.webrtc.org/2912523003 Cr-Commit-Position: refs/heads/master@{#18279}
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.