Streaming is choking WebRTC

[Samson_Radu]

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.

[Jason_Tuchler]

Hello @Samson Radu

If you are using H.264 as your preferred video codec I would recommend setting this to VP9 to see if you get better results in your [install-dir]/conf/webrtc/Application.xml:

<Property>
	<Name>webrtcPreferredCodecsVideo</Name>
	<Value>vp9,vp8,h264</Value>
	<Type>String</Type>
</Property>

Please save and restart your webrtc application for changes to take effect.

If you leave the bitrate empty it will default to about 30 fps and will adjust according to the current network conditions.

Regards,

Jason

[Samson_Radu]

Was using vp8,h264. Will update and test.

[Jason_Tuchler]

@Samson Radu

Please let us know how that setting works for you.

Regards,

Jason

[Samson_Radu]

Works better when using VP9 instead of VP8! Thanks @Jason Tuchler