asapersson fc5e81c979 Replace first_packet_sent_ms_ in Call.
Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet).

BUG=webrtc:6244

Review-Url: https://codereview.webrtc.org/2825333002
Cr-Commit-Position: refs/heads/master@{#17777}
2017-04-20 06:28:53 +00:00
..
2017-04-11 18:28:10 +00:00
2017-04-19 15:37:36 +00:00
2017-04-11 08:50:23 +00:00
2017-04-19 15:37:36 +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.