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

Wed Mar 2 12:17:46 CST 2016

On Mar 2, 2016, at 12:57 PM, Tickling Contest <tickling.contest at gmail.com> wrote:
> 
> 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.).
> 

Are you talking about having SIPp A call SIPp B? If so, it’s possible, you just need two separate scenarios: One to send the INVITE  and wait for the ANSWER, and another to (optionally) REGISTER and wait for the INVITE, then send the ANSWER.  You can also use RTP Echo within SIPp to reflect any received media back to the far end, which is an easy way to pass audio without messing with recordings.

> 
> 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
> 
> 

You need the Ruby development package installed to get mkmf, something like “apt-get install ruby-dev".  If you have further questions on SippyCup I’ll be happy to help you off-list, as it’s not really relevant here. Feel free to open an issue at https://github.com/mojolingo/sippy_cup/issues/new <https://github.com/mojolingo/sippy_cup/issues/new>

/BAK/

--
Ben Klang
Principal/Technology Strategist, Mojo Lingo
bklang at mojolingo.com <mailto:bklang at mojolingo.com>
+1.404.475.4841

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

> On Wed, Mar 2, 2016 at 12:28 PM, Ben Klang <bklang at mojolingo.com <mailto:bklang at mojolingo.com>> wrote:
> 
>> On Mar 2, 2016, at 11:30 AM, Tickling Contest <tickling.contest at gmail.com <mailto: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/ <http://sipp.sourceforge.net/>).
>> 
> 
> Shameless plug: you may find SippyCup will help make creating load test profiles easier: https://mojolingo.github.com/sippy_cup <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 <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 <http://sipp.sourceforge.net/doc/reference.html#PCAP+Play>
> 
> 
> /BAK/
> 
> --
> Ben Klang
> Principal/Technology Strategist, Mojo Lingo
> bklang at mojolingo.com <mailto:bklang at mojolingo.com>
> +1.404.475.4841 <tel:%2B1.404.475.4841>
> 
> Mojo Lingo -- Voice applications that work like magic
> http://mojolingo.com <http://mojolingo.com/>
> Twitter: @MojoLingo
> 
> 
>> 
>> Your help is received with thanks!
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> 
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