On Thu, 7 May 2015, Jim Gettys wrote:
Ideally, we need to get someone involved in WebRTC to help with this, to
present statistics that may be useful to end users to predict the
behavior of their service.
If nothing else, I would really like to be able to expose the realtime
application and its network experience, to the user.
For the kind of classic PSTNoverIP system I mentioned before, it was
usually possible to collect statistics such as:
Packet loss (packet was lost completely)
Packet re-ordering (packets arrived out of order)
Packet PDV buffer miss (packet arrived too late to be played on time)
I guess it's possible to get PDV buffer underrun or overrun (depending on
how one sees it), if I get a bunch of PDV buffer misses and then I halt
play-out to wait for the PDV buffer to fill up, and then I get 200ms worth
of packets at once and I don't have 200ms worth of buffer, then I throw
away sound due to that...
So it's all depending on the whole machinery and how it acts, you need
different statistics. How to present this in a useful manner to the user
is a very interesting problem, but it would be nice if most VoIP
applications at least had a "status window" where these values could be
seen in a graph or something similar to "task manager" in windows.
--
Mikael Abrahamsson email: swm...@swm.pp.se
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