[asterisk-app-dev] Testing an ARI application using SIPp

[Tickling Contest]

Thanks for your response, Ben.

To clarify, I'd like to just wait for an INVITE and then ANSWER. And then
I'd like to do it a random number of times (so I can ANSWER a random number
of calls). I don't need to hear audio, but I do want to simulate two people
being on a call (no need to call a real endpoint etc.).


BTW, when I try to install sippy_cup, I get an error:

~/sipp-3.5.0$ sudo gem install sippy_cup
Fetching: network_interface-0.0.1.gem (100%)
Building native extensions.  This could take a while...
ERROR:  Error installing sippy_cup:
ERROR: Failed to build gem native extension.

        /usr/bin/ruby1.9.1 extconf.rb
/usr/lib/ruby/1.9.1/rubygems/custom_require.rb:36:in `require': cannot load
such file -- mkmf (LoadError)
from /usr/lib/ruby/1.9.1/rubygems/custom_require.rb:36:in `require'
from extconf.rb:1:in `<main>'


Gem files will remain installed in
/var/lib/gems/1.9.1/gems/network_interface-0.0.1 for inspection.
Results logged to
/var/lib/gems/1.9.1/gems/network_interface-0.0.1/ext/network_interface_ext/gem_make.out


On Wed, Mar 2, 2016 at 12:28 PM, Ben Klang <bklang at mojolingo.com> wrote:

>
> On Mar 2, 2016, at 11:30 AM, Tickling Contest <tickling.contest at gmail.com>
> 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 at mojolingo.com
> +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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