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}
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.