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

[Sean Brady]

You can also use pjsua if you're looking for something simpler. If you know
Python there are bindings for it as well. Freeswitch is incredibly suited
to what I believe is your use case.

HTH

On Mar 20, 2016, at 20:39, Tickling Contest <tickling.contest at gmail.com>
wrote:

Thanks George, I think I am very, very, very confused with sipp and how it
handles the coordination (I thought I knew this well, but the pause and ti.
There _HAS_ to be a simpler way. It is just way. Too. Complex. I just
surprised that there isn't a better tool for something that has a load  of
use. Maybe I should move to Asterisk based testing. Known beast...

I have already gotten it working for a single call; you will recall in my
OP I wasn't able to push it beyond about a 100 calls concurrently, and
that's when I decided to let sipp manage everything.

The sipp software, I think is also quite buggy.

For example, I know that -p flag is supposed to take the port over which
the peer registers. This port shows up when you do pjsip show endpoint
<peerExtension>. I know this (i.e., -p param)  works in my sipp because
that's how I controlled each peer in my earlier sipp load test scenario.

Well, now, the local_port does not work when I pass it as a CSV file and
modify the [local_port] to [field0] etc, and as a result, the calls are not
going through.

They also don't have the updated documentation for release 3.5.1 which is
what I am using. Sigh!

On Sun, Mar 20, 2016 at 9:12 PM, George Joseph <george.joseph at fairview5.com>
wrote:

>
> 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.
>> ​
>>
>>
> _______________________________________________
> asterisk-app-dev mailing list
> asterisk-app-dev at lists.digium.com
> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
>
>
_______________________________________________
asterisk-app-dev mailing list
asterisk-app-dev at lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-app-dev/attachments/20160320/8c55d559/attachment.html>