[asterisk-users] Best Practices: Empirical measure of call latency

[Karl Fife]

I like your idea Michael.  Is the increment of delay of the echo service
known?  I suppose you'd have to back that out of the measurement. 

I was thinking of something similar (using audio editing software to
measure time between 'clicks') but more kludgy than your idea -- my idea
was to test the services in the form of LOOPS so I could HEAR the delay
myself.  Then the idea was to mesure the time between the first click
and the return click.

I imagine that someone out ther must have created a more automated way
to do this. 
Maybe the best reasons to have it automated would be to test for
variance over time. 

I recall several occasions using VoicePulse to terminate calls to
Switzerland: Call latencies of one full second or greater--A callback
would often 'fix' the problem.  

Thanks for your input!
-Karl


On Tue, 01 Jul 2008 22:40:20 -0500, "Michael Graves" <mgraves at mstvp.com>
said:
> On Tue, 01 Jul 2008 17:57:31 -0500, asterisk-users at kfife.mailworks.org
> wrote:
> 
> >I would like to hear your favored method to obtain an empirical measure
> >of latency in the media path.  
> >I'm doing several things that bring the media path through asterisk, and
> >this would allow me to make informed decisions about
> >
> >(a)PSTN termination providers
> >(b)DIDs in local and remote locations (and variance between ITSP's)
> >(c)time to/from various cellular networks  (and variance between ITSP's)
> >
> >Thanks!  Your opinion would be greatly appreciated
> >-Karl Fife
> >
> >p.s.
> >Speaking of latency, I've noticed that some sip endpoints (i.e. Aastra
> >57i Wireless) add significant latency.  It would be interesting to do an
> >apples-to-apples comparison between with various fxo/dect, sip/dect,
> >wi/sip, fxo/Spread-spectrum digital , and fxo/analog 47/900/2400mhz.
> 
> I had a project not long ago where I thought I was going to have to
> make  a comparison between the latency presented by two different call
> paths. In the end it wasn't necessary, but it did get me thinking about
> what I could do, lacking for any special equipment.
> 
> I had thought that I'd locate an echo test on a remote server. Free
> World Dialup still runs one that's accessible by both SIP and IAX2. My
> hosted PBX provider has one accessible via PSTN or SIP.
> 
> Then I'd use a mechanical click generator (impulse) at the handset
> while recording the call. Then take the recording into a waveform
> editor software and measure the timing differences between the various
> paths.
> 
> I can't say that this would be any kind of recommended practice, but I
> do think that I could get a sense of the comparative path
> lengths/timings.
> 
> Michael
> --
> Michael Graves
> mgraves<at>mstvp.com
> http://blog.mgraves.org
> o713-861-4005
> c713-201-1262
> sip:mjgraves at pixelpower.onsip.com
> skype mjgraves
> 54245 at fwd.pulver.com
> 
> 
> 
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-- 
  Karl Fife
  kfife at fifefamily.net