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


 
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asterisk-users at kfif...
Guest
PostPosted: Tue Jul 01, 2008 5:57 pm    Post subject: [asterisk-users] Best Practices: Empirical measure of call l Reply with quote
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.
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mgraves at mstvp.com
Guest
PostPosted: Tue Jul 01, 2008 10:40 pm    Post subject: [asterisk-users] Best Practices: Empirical measure of call l Reply with quote
On Tue, 01 Jul 2008 17:57:31 -0500, asterisk-users at kfife.mailworks.org
wrote:

Quote:
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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asterisk-users at kfif...
Guest
PostPosted: Wed Jul 02, 2008 12:30 pm    Post subject: [asterisk-users] Best Practices: Empirical measure of call l Reply with quote
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:
Quote:
On Tue, 01 Jul 2008 17:57:31 -0500, asterisk-users at kfife.mailworks.org
wrote:

Quote:
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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asterisk.org at sedwar...
Guest
PostPosted: Wed Jul 02, 2008 3:14 pm    Post subject: [asterisk-users] Best Practices: Empirical measure of call l Reply with quote
Quote:
On Tue, 01 Jul 2008 22:40:20 -0500, "Michael Graves" <mgraves at mstvp.com>
said:
Quote:
On Tue, 01 Jul 2008 17:57:31 -0500, asterisk-users at kfife.mailworks.org
wrote:

Quote:
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:

Quote:
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 Smile)

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 sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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