On Sat, Oct 17, 2009 at 11:32 AM, Patrick Collins <normalbl...@gmail.com>wrote:
> Hello,
> Can anyone please give me some hints on how to use Sipp to do load testing
> on our IVR system. What I need to do is dial the IVR via a PSTN number wait
> for it to play it's voice prompt and then send a DTMF 9, wait some more and
> then hangup.
>
Maybe something like this would work:
<send retrans="500">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
...
]]>
</send>
<recv response="200" rrs="true">
</recv>
<send>
<![CDATA[
ACK [next_url] SIP/2.0
...
]]>
</send>
<pause milliseconds="(DurationOfThePrompt)"/>
<nop>
<action>
<exec play_pcap_audio="filecontainingDTMF.pcap"/>
</action>
</nop>
<pause milliseconds="(Post_DTMF_Pause)"/>
<send retrans="500">
<![CDATA[
BYE [next_url] SIP/2.0
...
]]>
</send>
(or you might send a SIP INFO message instead of playing a pcap file if the
PSTN-GW converts it to DTMF).
>
> I have successfully installed Sipp on FreeBSD 7.2 (a virtual provided by
> RootBSD) and have created an account with Flowroute. I want to use Sipp
> running on the FreeBSD machine to create calls to the PSTN via Flowroute. I
> need to ramp up the call rate from 0 to 100 calls per second over 50 seconds
> and then sustain it for 5 minutes and then ramp down back to 0.
>
If you want this automatized (meaning not having to increase/decrease call
rate by hand), you could use some scripting to control SIPp. This topic
hints you can send commands to SIPp via UDP socket so you just have to write
a small perl/python/ruby etc script:
http://sipp.sourceforge.net/doc/reference.html#Controlling+SIPp
(however, I never trying any of the above).
HTH
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