[Asterisk-Users] quality and delay test

[Prezydent Kaczynski]

It there avalible quality and delay test for sip connections for
asterisk. Something like to clients making a call with different codecs and
measuring delay , jitter ? I know there is a Astertest but in that you
need 2 asterisk mashines (which is usually hard to have).
I was looking for perl/bash scripts running sip clients in a finite
loop + etheral to measure packet properties  , gathering logs.

Is there anything like that?

ps.
I have hudge delays (about 1 sec ! )when calling sip-sip and sip-isdn
on lan and i dont know what is the problem.

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Z uszanowaniem
 Prezydent                          mailto:prezydent_kaczynski at o2.pl