This change enables voice-only calls to keep track of the network state. This is only a partial fix - the last modality to change state controls the state for the entire call, so a call with a failed video transport will also stop sending audio packets. Handling this condition correctly would require the call to keep track of network state for each media type separately, and take care of conditions such as a failed video channel getting removed, while a functioning audio channel remains. BUG=webrtc:5307 Review URL: https://codereview.webrtc.org/1757683002 Cr-Commit-Position: refs/heads/master@{#12093}
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.