| commit | af4ced986bc62c263fbdb6eab68aef5c0d4e7c78 | [log] [tgz] |
|---|---|---|
| author | sprang <sprang@webrtc.org> | Tue Oct 06 13:02:53 2015 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 06 13:02:57 2015 |
| tree | 2101bff986c8f05362741f9d0d13d7cdc627a452 | |
| parent | 86fa298c3ead4fb207954325f33181a7ef9bf55e [diff] |
Transport sequence number should be set also for retransmissions. When fetching a packet from the rtp packet history, cuased by a retransmission, the transport seq extension header is enabled but the sequence number is set to 0. A new transport seq should be assigned in this case. BUG= Review URL: https://codereview.webrtc.org/1385563005 Cr-Commit-Position: refs/heads/master@{#10183}
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.