[This CL is work in progress.] Wire up the rtp keep-alive in webrtc::Call::Config using new SetRtpTransportParameters() method on RtpTransportInterface. BUG=webrtc:7907 Review-Url: https://codereview.webrtc.org/2981513002 Cr-Commit-Position: refs/heads/master@{#19287}
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.