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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.