Re: [Sipp-users] How do I load test for a simple caller->callee tests through Asterisk PBX?

[Tickling C. <tic...@gm...>]

Sorry, I did not send it to the correct mailing list originally:

Tyler,

Thanks, I am happy there is somebody else who shares my frustration with
sipp (and I enjoyed the adjectives you used to describe the life of a
SIPper on the github page). SIPp is very useful software, for sure, but it
is godawfully badly documented (and uses (ugh!) XML) and there are many
confusing handles. They will save a world of questions and trouble if they
could just present an integrated scenario for a simple caller->callee
scenario with auth using known SIP controllers like Asterisk.

My hope is that if I ever get out of this process alive, I am going to
document the crap out of this, and contribute it back here. I have consumed
enough of everyone's time to warrant that sort of PSA.

Anyway, how would you write the following scenario with pysipp?

Callers (for a bunch of callers in CallerCSV.csv):
======================================
REGISTER
Send out INVITE

(when INVITE was accepted)
pause for a call duration selected from a uniform (10 s to 180 s)
Send BYE

Callees (for a bunch of callees in CalleeCSV.csv):
======================================
REGISTER
Wait for INVITE
Accept the call
Wait for BYE
send 200 OK

I am sold on pysipp if it can do this in the simplest possible fashion. The
current example on your github page probably does this, but I wanted to
check if there is something else I need to do.

What would my caller.xml and callee.xml (any other xml needed?) look like?
What are the referee and referer xmls in your page?

Thanks!

On Wed, Mar 23, 2016 at 12:04 PM, Tyler Goodlet <tgo...@gm...> wrote:

> Hey Tickling,
>
> Usually when you see a TCP connection error like that it means you're
> trying to contact a socket that isn't being listenend on.
> Just breifly looking over your shell scripts I can see that caller.sh
> isn't referencing the callee.sh $SIPPort anywhere.
> Are you routing to that port in the asterisk dialplan?
> So it might be simple issue that when asterisk bridges the two legs it's
> making the initial request to port 5060 on the b-leg and you don't have the
> callee SIPp agent listening there.
>
> On another note, (and I think I've recommended this to you before) you
> might want to try out pysipp https://github.com/SIPp/pysipp.
> It's going to let you avoid all these separate shell scripts and will make
> running the registration in sequence with the call a lot easier.
> If you want example code I'd be glad to modify your gist.
>
> Let me know!
>
>
> Tyler Goodlet
>
>
> On Mon, Mar 21, 2016 at 9:53 PM, Tickling Contest <
> tic...@gm...> wrote:
>
>> Hello,
>>
>> I am using an Asterisk PBX 13.7.2 which I would like to load test. For
>> this question, please use my gist at
>> https://gist.github.com/ticklingcontest/a67a4386cf73398efb82
>>
>> I  am trying to load the PBX using calls generated by sipp using the
>> shell scripts in the gist in the following manner:
>>
>> (a) callee.sh is run first with the calleeCSVFile (with the fields
>> flipped; I do not show this CSV file in the gist)
>> (b) caller.sh is run next
>>
>> The hope is that callers are able to call the callee.
>>
>> Unfortunately, I see that the INVITEs are being received at the Asterisk
>> PBX, but they are never sent to the callee endpoints because:
>>
>> [Mar 21 21:24:56] ERROR[1535]: pjsip:0 <?>: tcpc0x7efd6807 TCP connect()
>> error: Connection refused [code=120111]
>> [Mar 21 21:24:56] WARNING[1535]: pjsip:0 <?>: tsx0x7efd6808d Failed to
>> send Request msg INVITE/cseq=17210 (tdta0x7efd68085a80)! err=120111
>> (Connection refused)
>>
>> How do I fix this?
>>
>> Any help is appreciated.
>>
>> Also, can I please get some help with my *.xml files? All my
>> registrations end up getting the following error even though I use the -aa
>> parameter:
>> Aborted call with Call-ID 'de92a989-dea1-45a3-835f-07d57fcddd76'. (The
>> NOTIFY that is received has the same call-id).
>>
>> BTW, my caller script works when calling a real SIP endpoint (on my
>> phone). I just cannot get it to work on this script.
>>
>> Can someone tell me how I can make a simple call using endpoints that
>> connect via Asterisk AFTER two endpoints REGISTER at a PBX? Do I need 3 XML
>> files like I use (1 for registration and 1 each for caller and callee) or
>> is there a better way? If I do need separate XML files, how do I make sure
>> that a number of endpoints on a sipp instance are found by the call
>> receiving instance of sipp?
>>
>> Any help is appreciated.
>>
>> Thanks.
>>
>>
>> ------------------------------------------------------------------------------
>> Transform Data into Opportunity.
>> Accelerate data analysis in your applications with
>> Intel Data Analytics Acceleration Library.
>> Click to learn more.
>> http://pubads.g.doubleclick.net/gampad/clk?id=278785351&iu=/4140
>> _______________________________________________
>> Sipp-users mailing list
>> Sip...@li...
>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>
>>
>
>
> ------------------------------------------------------------------------------
> Transform Data into Opportunity.
> Accelerate data analysis in your applications with
> Intel Data Analytics Acceleration Library.
> Click to learn more.
> http://pubads.g.doubleclick.net/gampad/clk?id=278785351&iu=/4140
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>