[Asterisk-Dev] Asterisk stress test?

[Atif Rasheed]

sipp can help you, if I am not wrong, sipp echo's back rtp generated 
from other end. so if you generate rtp from the end asterisk, you can 
test the middle one with two-way rtp streams.

look into sipp documentation for details

Roy Sigurd Karlsbakk wrote:

> hi
>
> are there any known scipts etc to do a good stress test with  
> asterisk? I've been having some rather unpleasant issues with the so- 
> called stable version of asterisk for months, and people say that is  
> fixed in cvs head (http://bugs.digium.com/view.php?id=3986). i'll  
> need to mainly do SIP traffic generation, asterisk to asterisk, i  
> guess. I don't think sipp can help me out, as AFAIK that doesn't  
> handle RTP.
>
> all help appreciated
>
> roy
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