Check bandwidth during webRTC call

[Remya Das]

How can I check the bandwidth usage while a WebRTC call is in progress?

[Gustavo Garcia]

If you mean in the browser then you can use chrome://webrtc-internals page. 

If you mean programmatically then you can use the getStats API [1]

[1] https://w3c.github.io/webrtc-pc/#dom-peerconnection-getstats

On Tue, Apr 28, 2015 at 12:58 AM, Remya Das <remya...@gmail.com> wrote:

How can I check the bandwidth usage while a WebRTC call is in progress?

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[Remya Das]

Using chrome://webrtc-internals how to check?

I can see so many graphs there. Which one is for bandwidth?

I want to check audio bandwidth and video bandwidth.

[Peter Boström]

I think they're reported per stream, I usually check "Stats graphs for ssrc_3463008876_send" and similar. You can quite easily tell which ones are audio and which ones are video. (Audio doesn't have send width/height.)

[Remya Das]

The problem I'm facing is: I'm trying a iPhone (webRTC based App) to browser (Firefox) call. when the call is in progress if i'm doing some video streaming in youtube or if i'm making a voip call using the same network, the audio quality of webRTC drops and its very choppy.

How can fix this.

[Stefan Holmer]

You'd have to check webrtc-internals to see why the quality is low. Is it due to packet loss? Could be that your link gets saturated.

[Remya Das]

Which stats graph i have to check?

[Stefan Holmer]

I would look at audio jitter and packet loss on your receive streams.