I have set up the WebRTC demo code on a server in a different country. Then I tried to publish and play on 2 different machines.
Using the default settings that are provided with the example the stream chokes for a second or so every ~15-20 seconds.
Default params are:
var videoBitrate = 360;
var audioBitrate = 64;
var videoFrameRate = "29.97";
Lowering the frame rate does help, as choking happens less often so I assume it’s a bandwidth thing?
Where can I find docs about tuning these params.
Does leaving Bitrate empty mean adaptive to the network conditions? I tried it and it didn’t improve things.