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.
First I connect two of our clients together in a video chat.
Then I start logging the netstreamInfo from my audioSubscriber, and my WebcamSubscriber every 10 seconds
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?