Re: Fwd: Problem when load testing Asterisk 13.7.2

Sean Brady <sbrady@xxxxxxxxxxxxxxxxxxxx> · Sun, 20 Mar 2016 22:24:09 -0600

Well yes, however you can use the compiled pjsua binary for script based testing as well. Also, looking at the pjsip-apps dir, there is a bin called siprtp that might do just what you’re looking for. Check out ./pjsip-apps/bin/samples/ for siprtp and others. 

HTH

 -- 
Sean Brady
Sent with Airmail

On March 20, 2016 at 9:10:12 PM, Tickling Contest (tickling.contest@xxxxxxxxx) wrote:

Sean,

I think I understood your comment incorrectly, let me try this
again. I did try a PJSIP based testing framework a while ago. I
could not get it to work, and I have forgotten what it was.

Anyway, what I think you are saying is that if I have come
this far, why not write the entire testing framework using PJSIP
(and Python, maybe) as it can get quite simple if I use the high
level PJSUA binding.

I have not tried that. And at this time, it is worth a look,
especially because I find SIPp very cumbersome.

So, thanks!

On Sun, Mar 20, 2016 at 10:47 PM, Sean
Brady <sbrady@xxxxxxxxxxxxxxxxxxxx> wrote:

Works fine for me, and it compiles with PJSIP. 

On Mar 20, 2016, at 20:46, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:

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@xxxxxxxxxxxxxxxxxxxx> 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@xxxxxxxxx> 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@xxxxxxxxxxxxx> 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@xxxxxxxxxxxxx> wrote:

On Sun, Mar 20, 2016 at 5:29 PM,
Tickling Contest <tickling.contest@xxxxxxxxx>
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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