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

[Tickling Contest]

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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