[Asterisk-Users] SIP calls-per-second performance test tool

[Juan J. Sierralta P.]

On Thu, 2004-05-13 at 03:58, John Todd wrote:

> Chris -
>    It does not appear that sipp is a User Agent that is authenticated,

	Yes.

>  
> which is probably something that needs to be included in the tests, 
> since that adds ~30% additional packet chatter on an INVITE, and 
> involves some computation which could significantly change the 
> results of what SIPP finds vs. "real-world" situations.
> 
>    More investigation would lead me back to sipsak 
> (http://sipsak.berlios.de/) to see if perhaps some grafting of the 
> two packages could be made, such that the extended features of sipsak 
> (including authorization) could be programmed to include the RTP echo 
> module and "end-to-end" mode that sipp appears to support.  I'm not 
> sure which program would be better to modify...

	I like the feature of SIPP to be able to modify the UA using .xml
scenarios. And SIPP do echo the received audio the problem is that it
doesn´t generate audio.

> JT
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-- 
Juanjo sin .sig