That sounds logical. Thank you!

I'll try the setup.

-----Original Message-----
From: Andrew Miller [mailto:andrew.mil...@crocodile-rcs.com] 
Sent: maandag 10 september 2012 14:44
To: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio

Grant,

You could insert an extra Kamailio instance between SIPp and your test system. 
This would record-route and stay in dialog, allowing your "real" 
Kamailio to drop out of the dialog.

Effectively your load generator consists of SIPp + a Kamailio (or any
other) proxy

Andy Miller
Crocodile RCS Ltd


On 08/09/2012 23:04, Michael Hirschbichler wrote:
> Hi,
>
> this won't work AFAIK - sipp sends requests only to one remote socket 
> during a call.
>
> br
> Michael
>
> Am 07.09.2012 08:06, schrieb Grant Bagdasarian:
>> Hello,
>>
>> I'm trying to perform a loadtest on our Asterisk machines using Sipp, 
>> but there is a SIP Proxy(Kamailio) in between. Kamailio acts as a 
>> loadbalancer for our Asterisk machines, so it only remains in the 
>> dialog during the initial INVITE, TRYING and 200 OK. The ACK should 
>> be sent by SIPp directly to the Asterisk machines. This is where I'm 
>> having problems. I can't get SIPp to send the ACK directly to the 
>> Asterisk machines, without enabling record-route. When I enable 
>> record-route Kamailio stays in between, but this is not a 
>> representative loadtest for our live platform, since record-route is 
>> disabled on live. The contact header is set properly, but SIPp refuses to 
>> send the ACK directly.
>>
>> Is this a limitation of SIPp when used in this kind of setup, or am I 
>> doing something wrong? I have already loadtested Asterisk without 
>> Kamailio in between, that went fine.
>>
>> Regards,
>>
>> Grant
>>
>>
>>
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--
Andrew Miller
Crocodile RCS Ltd


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