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

Sun Mar 20 21:46:37 CDT 2016

Sorry, but I started with PJSUA and was told that the package is broken. I
couldn't get it work. Thanks, though.

On Sun, Mar 20, 2016 at 10:43 PM, Sean Brady <sbrady at haikuengineering.com>
wrote:

> 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.
>>> 
>>>
>>>
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