[asterisk-app-dev] Does anybody know to how to load test simple scenario in Asterisk 13.x w SIPp?

[Tickling Contest]

Also, it would be amazing if someone could tell me how to stop a SIPp test
after one such call described in my OP.

On Mon, Mar 14, 2016 at 8:37 PM, Tickling Contest <
tickling.contest at gmail.com> wrote:

> Hello,
>
> I am testing an ARI application written over Asterisk 13.6.0. In my ARI
> application, I wrote a bunch of features that manipulate call state, and I
> would now like to stress test it so that the application is production
> ready.
>
> So, I am using SIPp, but I am having a LOT of trouble with docs.
>
> Here's what I want to do, assuming each SIPp "peer" in the following
> description means a single SIPp process on, say, a linux host:
> (a) REGISTER SIPp peer A and SIPp peer B to my Asterisk 13.6.0 using TCP
> transport using username and password.
> (b) Use SIPp peer A to call SIPp peer B, both of whom are registered to my
> Asterisk 13.6.0/TCP PBX using INVITE.
> (c) Wait for the call to be answered  at peer B, and then pause 2 seconds
> and then kill the call by sending a BYE from SIPp peer B.
>
> Here is what I am able to do so far: I am able to REGISTER the peers, but
> for the life of me, not able to get the peers to call each other.
>
> I have tried sippy_cup, but I have had a similar issue where the
> documentation is spotty and I don't think it actually works for TCP. So
> that's not an option for me.
>
> I would like to know how to create XML scenario files so that I can
> generate my own scenario files for my own set up from actual pjsip set
> logger on output.
>
> Any help is deeply appreciated!
>
> Thanks!
>
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