asterisk users - Jul 2008 - Best Practices: Empirical measure of call latency

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.

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