[Asterisk-Users] Anyone using SipP to produce RTP load?

[Zoa]

That would probably be me.

You could use a lot of different things to do the testing,
one would be the tcl script in your asterisk/contrib/scripts directory,
some more can be found in the beginning of this presentation:
http://astertest.com/astricon_performance.ppt

We started some callgenerator for asterisk a very long time ago. ( I
have to admit its far from ready and contains many bugs).
A howto for this tool can be found at :
http://www.asteriskguru.com/tutorials/astertest.html

If you want to use sipp, be sure to use playback on your asterisk server
and not app_milliwatt, meetme or echo. (those applications will not send
any rtp if nothing was received). SIPP only does echoing

Zoa

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Matthew Boehm wrote:

>Hey gang,
> I've been able to use sipp to produce some call volume on our asterisk
>server. The server has no problems handling 50 simul calls. But then again,
>no RTP is being done. I tried to use the rtp echo ability of sipp but that
>doesn't seem to work right.
> I also setup a fake number in asterisk that when called by sipp, would dial
>another number via PRI, hoping that some 729 conversion would occur.
>Nothing. I was able to pump 10 simul calls that went this path:
>
>  sipp -> asterisk -> pri -> telco ->pri ->asterisk
>
>..and still no 729 usage or any other discernable load on the server.
>
>Can anyone offer suggestion on how to really simulate calls (using sipp or
>other tester) to asterisk to verify its ability to process X calls?
>
>I know someone out there has done this, but forget who it was.
>
>Thanks,
>Matthew
>
>
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