[asterisk-app-dev] Fwd: Problem when load testing Asterisk 13.7.2

[George Joseph]

Oh, BTW...

If sipp doesn't do it for you, there's another great tool you can use for
load testing.  It's called Asterisk. :)

For more complex scenarios, what I've done in the past is set up 3 Asterisk
instances, 1 as the call generator, 1 as the system under test, and 1 as
the call receiver.

On the generator instance, I have a script that keeps enough call files in
/var/spool/asterisk/outgoing to simulate the number of calls I want.  On
the call receiver, I can set up the dialplan to do anything I want with the
calls.  Transfer, play something, echo, park.  Whatever.


On Sun, Mar 20, 2016 at 6:37 PM, George Joseph <george.joseph at fairview5.com>
wrote:

>
>
> On Sun, Mar 20, 2016 at 5:29 PM, Tickling Contest <
> tickling.contest at gmail.com> wrote:
>
>> OK. I did that, but now, all I do is get into an infinite loop with the
>> registrations at the callee. Here's the gist:
>> https://gist.github.com/ticklingcontest/a0754549a88dc748f52d
>>
>> Ideally, here's what I need:
>>
>> callee registers, and accepts an infinite number of calls.
>> caller registers, and then sends INVITES an infinite number of times so
>> as to keep the total number of calls per the (-l parameter).
>>
>> It is not clear to me how I would loop at the callee scenario or caller
>> scenario.
>>
>
> ​You don't loop anything.  sipp runs the scenarios itself repeatedly
> until​ -m calls have been processed.
>
> I'd start without your script or the csv files and just get a simple 1
> call scenario to work.
>
> If you want some good examples, look at the Asterisk testsuite
> tests/channels/pjsip/basic_calls scenarios.
>
> Here's a caller file I used often...
> https://gist.github.com/gtjoseph/ce5a719b11f307c7ec5e
>
> register
> pause 1 sec
> invite
> pause 1 sec
> bye
> pause 1 sec
> unregister
>
> To simulate a call from a phone with extension/endpoint name 1100, run it
> like so...
> # sipp -sf reg_and_call.xml -s 1100 -au 1100 -ap <password> -m 1
> <server:ip>
>
> If you want it to resister/call/unregister 100 times with 10 parallel
> calls over TCP, run
>
> # sipp -sf reg_and_call.xml -t tn -s 1100 -au 1100 -ap <password> -m 100
> -l 10  <server_ip:port>
>
> Once you get that working by itself to an existing extension,  set up your
> callee the same way, then call it from a normal working extension and make
> sure it responds correctly.
>
> Then have your caller call the callee, first as a single call, then try
> multiple calls.
>
> Only when you have that working should you  introduce your injection files.
>
>
>> What is really confusing in the caller script apart from the real
>> confusion I have with -m and -l parameters, is how the caller's INVITE goes
>> out from the same port as the registered port especially when they are
>> called as two separate processes. Does sipp write a  dot file somewhere
>> where it gets its information from?
>>
>
> ​Nope.​
>
>
>> BTW, In this model, I pass the CSV file that is pre-generated for the
>> calls using a python script that looks like this:
>>
>> SEQUENTIAL
>> callerID1;AsteriskIPAddress;[authentication username=silly
>> password=sillier];calleeID1;callDuration1;
>> callerID2;AsteriskIPAddress;[authentication username=silly
>> password=sillier];calleeID2;callDuration2;
>> ...
>> callerIDn;AsteriskIPAddress;[authentication username=silly
>> password=sillier];calleeIDn;callDurationn;
>> etc.
>>
>> Again, any help is appreciated. I can see how this is turning into a sipp
>> tutorial, so I understand if you have issues dealing with this here, but I
>> can tell that SIPp help is very sparing online.
>>
>> Thanks!
>>
>
> ​I'll say again...  If you want some good examples, look at the Asterisk
> testsuite tests/channels/pjsip/basic_calls scenarios.  There are both
> inbound and outbound scenarios, authed and unauthed.
> ​
>
>
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