hbos a7a9be159d 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}
2017-03-01 09:02:45 +00:00
..
2017-02-28 23:43:10 +00:00
2017-02-28 23:43:10 +00:00
2017-02-27 10:18:46 +00:00
2015-11-16 19:02:02 +00:00
2017-01-30 08:54:19 +00:00

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.