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" <[EMAIL PROTECTED]> said: > On Tue, 01 Jul 2008 17:57:31 -0500, [EMAIL PROTECTED] > 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:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Fife [EMAIL PROTECTED] _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
