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@xxxxxxxxxxxxx> wrote:
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_cupI 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./BAK/--Ben KlangPrincipal/Technology Strategist, Mojo LingoMojo Lingo -- Voice applications that work like magicTwitter: @MojoLingo_______________________________________________Your help is received with thanks!
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