Hi Adam,

thanks a lot for your input - I'll work on setting up a data-channel based 
messaging ping back as a measure as it seems to be a solid way to get equally 
meaningful stats in both Firefox and Chrome - thanks for the tip.


I'll ping you separately to set up a quick call with you to get your thoughts 
on what wwe have so far ok? Looking forward to that!


Thanks again


Jamie


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From: Adam Roach <[email protected]>
Sent: 03 February 2014 18:06
To: Jamie McDonnell; [email protected]
Cc: David Patten
Subject: Re: peerConnection getStats

On 2/3/14 03:33, Jamie McDonnell wrote:

so I took a look into the stats available currently, and unfortunately 
Available Send Bandwidth and Available Receive Bandwidth don't seem to be 
exposed, which is what I am using to represent the quality of the call, along 
with the jitter value in Chrome.

We're targeting the evolving stats interface that is being discussed in the 
W3C, which does not include these metrics (at least, not at the moment), 
although we may introduce additional information on an experimental basis (just 
like Chrome has with the "bandwidth" stats you mention):

http://www.w3.org/2011/04/webrtc/wiki/Stats

Basically, if you don't see a stat listed on that page, I'd be very cautious 
about using it, as it may be removed or changed significantly, based on the 
conversations currently underway at the W3C. (And, even if something is listed, 
it might be changed based on those discussions, although it's less likely.)


What are your thoughts on representing call quality to the end user in a simple 
visual way?

The dominating factors in media quality are going to be jitter, packet loss, 
and RTT.


  *   For jitter, we already have an entry in the stats report
  *   I'll suggest to the working group that we should add packet loss to the 
evolving list of stats
  *   As far as I know, there's no good way to get RTT information out of RTP 
and RTCP, so you'll probably want to rig something up with a DataChannel that 
basically sends a small message with a sequence number to the other side, which 
responds immediately; the time between sending the message and getting the 
response is you RTT

>From there, I'd try to work out a quality metric by starting with something 
>like:

(min(6ms - jitter_in_ms, 0) * (100% - packet_loss_in_percent) * min(200ms - 
rtt_in_ms, 0))/600

...which should get you a number between 1 and 100. You might want to tweak 
things a bit, but that should get you a good first pass estimate.


I'm happy to demo to you in Chrome what we currently have (if that doesn't go 
against your morals? ;)

Sure, send me a URL to look at. :)


--
Adam Roach
Principal Platform Engineer
[email protected]<mailto:[email protected]>
+1 650 903 0800 x863
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