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can I use SSRT as a QOS metric?


Level 4

I'm wondering if I can use SSRT as a good metric for how good the P2P link is between two video chat clients.  I did the following experiment.

  1. First I connect two of our clients together in a video chat.
  2. Then I start logging the netstreamInfo from my audioSubscriber, and my WebcamSubscriber every 10 seconds
  3. Then I slowly restrict the bandwidth on my mac from full open all the way down to about 8KBytes/second

What I see as I do this is SSRT climbs in proportion to how restricted my bandwidth becomes.   So at full open it's around 0-6, by the time I'm at 8KB/s it's up in 2000-3000 range.   I should also mention that I'm restricting my bandwidth symetrically so 8KB/s up 8KB/s down.  I also tried adjusting my encodeQuality, framesPerPacket at the 8KB/s and I think it gives me a better audio experience.

So I'd like to do some work to do this automatically but first I wanted to ask if you guys thought using SSRT for this was a good idea?

Thanks as usual,


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