| commit | 9657172a81f6caf150b431c2b756a0a2a69f850c | [log] [tgz] |
|---|---|---|
| author | henrik.lundin <henrik.lundin@webrtc.org> | Wed Aug 30 07:41:30 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 30 07:41:30 2017 |
| tree | 15bc89642ca411b3d7c091c65e297229d7a81a45 | |
| parent | 85e6a4ba1372f21b8648ffaad2fd19a76a8bb316 [diff] |
neteq_rtpplay: Add one more RTP header extension and fix some stats The extension ID for transport sequence number is added to the list of known RTP header extensions. Also, the minimum and maximum waiting time for packets is now aggregated as minimum and maximum, respectively, not as averages. BUG=none Review-Url: https://codereview.webrtc.org/3004783003 Cr-Commit-Position: refs/heads/master@{#19593}
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.