[asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

Oh jeez. Another GUI...

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of lenz
> Sent: Thursday, August 09, 2007 6:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> 
> I have used this freeware tool in the past:
> http://sineapps.com/sinestatiax.php
> maybe you can have a look at it as well....
> l.
> 
> 
> In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd <jtodd at loligo.com>
ha
> scritto:
> 
> > At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
> >>  > At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
> >>>  >
> >>>  >How can I objectively measure jitter in Asterisk on a SIP
channel?
> >>>  >
> >>>  >I don't just want to turn the new 1.4 jitter buffer on. I want
to
> >>>  >measure jitter.
> >>>  >
> >>>  >Thanks,
> >>>  >Doug.
> >>>
> >>>  You could look at the txjitter and rxjitter values (and other
values)
> >>>  stored in the CHANNEL() function, and those values you're looking
for
> >>>  were previously known as RTPAUDIOQOS.  Or is this not sufficient?
> >>
> >> Are txjitter and rxjitter working reliably? These calls are going
to be
> >> placed from AMI and bridged together. Do you think the variables
would
> >> be correctly set for each leg of the call?
> >>
> >> Doug.
> >
> > I think the best way to determine this would be to compare the
> > numbers provided by CHANNEL() versus the numbers provided by
> > something with a little more reliability, such as wireshark, in a
> > controlled set of circumstances.
> >
> > Please post your results here - it would be an interesting test.
> >
> > JT
> >
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