| commit | a7a9be159df454db9d861b49b94b3d02a283eb1c | [log] [tgz] |
|---|---|---|
| author | hbos <hbos@webrtc.org> | Wed Mar 01 09:02:45 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Mar 01 09:02:45 2017 |
| tree | 397c1d4b1883fb6adf63e3f006ded8f95f24cefb | |
| parent | deaf6fb071e68a242db342d61a11220edd2bb546 [diff] |
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. The value is being moved: https://github.com/w3c/webrtc-stats/pull/167 Stop collecting this value. Our previous value was incorrect, our RTT value was a smoothed value based on STUN pings but the spec says it should be based on RTCP timestamps in RTCP Receiver Report (RR) on inbound streams with isRemote=true (not supported). Updated some bug references. BUG=webrtc:7065, webrtc:7066 Review-Url: https://codereview.webrtc.org/2722633005 Cr-Commit-Position: refs/heads/master@{#16931}
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.