Re: Testing an ARI application using SIPp

Ben Klang <bklang@xxxxxxxxxxxxx> · Wed, 2 Mar 2016 12:28:30 -0500

On Mar 2, 2016, at 11:30 AM, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:

Hello,
I have an ARI application that I would like to load test using SIPp (http://sipp.sourceforge.net/).

Shameless plug: you may find SippyCup will help make creating load test profiles easier: https://mojolingo.github.com/sippy_cup

I understand how to REGISTER and send an INVITE out to a callee, but it is not clear to me:

(a) how to choose a RANDOM port for each client I launch using say http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=pl 
(REGISTER_INVITE_client.xml). Should I generate this beforehand and pass it as field3, say? Or is there a method already available in SIPp to do this?

I don’t know about making the ports random, but you can allocate a unique port-per-connect by passing the “-t un” flag to sipp. Did you really need them random, or just unique? And out of curiosity, why?  Most of the time, the default of “-t u1”, which is a single UDP port shared for all dialogs, works fine.

(b)  How do I receive a call into a SIPp client? In other words, how do I simulate a SIPp client to REGISTER and then wait for a call (from a SIPp client connected to the same PBX)?

This gets tricky because SIPp isn’t a real UA.  However, you can write a SIPp scenario that sends a REGISTER then waits for an INVITE.  The Sippy Cup documentation has an example of this.

(c) Can I play an audio file (or an mp3) after the call starts from both the caller and callee?

You can replay audio in the form of a pcap file. If I need real audio, the easiest path is usually to place a real call with tcpdump running.
http://sipp.sourceforge.net/doc/reference.html#PCAP+Play

/BAK/

-- 
Ben Klang
Principal/Technology Strategist, Mojo Lingo
bklang@xxxxxxxxxxxxx
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo

Your help is received with thanks!

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