Re: [asterisk-users] Best Practices: Empirical measure of call latency
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 mgravesatmstvp.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
Re: [asterisk-users] Best Practices: Empirical measure of call latency
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 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. On Wed, 2 Jul 2008, Karl Fife wrote: 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. Several years ago, I replaced an aging Dialogic based adult chat system with Asterisk. One day I made the mistake of letting the client listen to the system with a handset on each ear. The delay was noticeable and the client was a stickler. (Note to self - NEVER let a client do that :)) I used 2 RadioShack 43-228A telephone recording controls (left and right) feeding into a Y and then into my laptop. Using Audacity, I would tap the left handset and you could measure the delay until the tap registered on the right. Only by demonstrating that the measured delay was below what several studies showed the threshold for interfering with conversation saved the project. That and demonstrating the delay in a cell to cell based call with a handset on each ear. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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
[asterisk-users] 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. ___ -- 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
Re: [asterisk-users] Best Practices: Empirical measure of call latency
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 mgravesatmstvp.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
