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

[Michael Graves]

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
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sip:mjgraves at pixelpower.onsip.com
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