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

Tickling Contest tickling.contest at gmail.com
Sun Mar 20 18:29:57 CDT 2016 

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.

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?

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!

On Sun, Mar 20, 2016 at 3:53 PM, George Joseph <george.joseph at fairview5.com>
wrote:

>
>
> On Sun, Mar 20, 2016 at 11:52 AM, Tickling Contest <
> tickling.contest at gmail.com> wrote:
>
>> Here you go, George. I appreciate your energy into this:
>> https://gist.github.com/ticklingcontest/4762a57457b73db1a170
>>
>
> ​Hmmm.
>
> So are you running 1 instance of the script for each call?   I think
> that's the issue.
>
> Why not let sipp do the work?
> For the callee, combine registration.xml and callee.xml into the same file
> and set -l to the max number of simultaneous calls to process.  Then set
> the transport to tn instead of t1 and just let it sit and listen for
> connections.
>
> ​Same thing for caller, except for the registration.
>
> ​Now you only have 2 instances of sipp.​
>
>
>
>> On Sun, Mar 20, 2016 at 10:15 AM, George Joseph <
>> george.joseph at fairview5.com> wrote:
>>
>>>
>>>
>>> On Sat, Mar 19, 2016 at 8:29 PM, Tickling Contest <
>>> tickling.contest at gmail.com> wrote:
>>>
>>>> Thanks again, George,
>>>>
>>>> I am running PJSIP v. 2.4.5 that is prescribed for Asterisk 13.7.2. I
>>>> made the changes you proposed re: --enable-epoll, recompiled PJSIP/Asterisk
>>>> and redid the tests.
>>>>
>>>> I am attempting a total of 100 or so connections (50 SIP/TCP callers
>>>> REGISTERing, INVITEing; 50 SIP/TCP callees REGISTERing, accepting calls). I
>>>> am afraid the latest Asterisk is not something I can try right now.
>>>>
>>>>
>>> ​How many sipp instance are you running?
>>> Can you share the command lines and scenario xml files?
>>>
>>>
>>> I am left now with an issue that I have not been able to get help
>>>> elsewhere re: SIPp. I hope I can ask them here, but if I can't, I am sorry
>>>> in advance!
>>>>
>>>> I get these errors:
>>>>
>>>> Unable to bind audio RTP socket (IP=127.0.0.1, port=6100), errno = 98
>>>> (Address already in use). (also get the same error but with video RTP
>>>> socket, but I am NOT running any video tests).
>>>>
>>>> OR
>>>>
>>>> Unable to bind main socket, errno = 98 (Address already in use).
>>>>
>>>> OR
>>>>
>>>> Unable to bind remote control socket (tried UDP ports 8888-8947):
>>>> Address already in use. (no idea what this is, and why this occurs)
>>>>
>>>> I tried passing the -mp parameter (for sipp, i.e.,) a random port
>>>> number (e.g., see
>>>> http://stackoverflow.com/questions/2556190/random-number-from-a-range-in-a-bash-script)
>>>> but I still get these issues (though they dramatically decreased the number
>>>> of issues I see due to port number).
>>>>
>>>> Apart from ulimit/FD_SETSIZE related changes, what else can I do?
>>>>
>>>> Where can I get information about load testing Asterisk for more than
>>>> 100 concurrent calls (I use ARI, so I have to test my backend application
>>>> too) etc.?
>>>>
>>>> Thanks!
>>>>
>>>> On Sat, Mar 19, 2016 at 5:13 PM, George Joseph <
>>>> george.joseph at fairview5.com> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Sat, Mar 19, 2016 at 1:30 PM, Tickling Contest <
>>>>> tickling.contest at gmail.com> wrote:
>>>>>
>>>>>> Thank you, George!
>>>>>>
>>>>>> That did solve the trouble for the most part, but I still get a few
>>>>>> of these (not rarely, unfortunately):
>>>>>>
>>>>>> Mar 19 15:23:06] ERROR[1981]: pjsip:0 <?>: tcpc0x7fb0083a TCP
>>>>>> connect() error: Connection refused [code=120111]
>>>>>> [Mar 19 15:23:06] WARNING[1981]: pjsip:0 <?>: tsx0x7fb00008c Failed
>>>>>> to send Request msg BYE/cseq=8737 (tdta0x7fb010246eb0)! err=120111
>>>>>> (Connection refused)
>>>>>> [Mar 19 15:23:16] ERROR[1981]: pjsip:0 <?>: tcpc0x7fb0083a TCP
>>>>>> connect() error: Connection refused [code=120111]
>>>>>> [Mar 19 15:23:16] WARNING[1981]: pjsip:0 <?>: tsx0x7fb010169 Failed
>>>>>> to send Request msg BYE/cseq=5278 (tdta0x7fb010096950)! err=120111
>>>>>> (Connection refused)
>>>>>>
>>>>>> I set the value of PJ_IOQUEUE_MAX_HANDLES to FD_SETSIZE per the link
>>>>>> you mentioned and ulimit for the root account, which runs asterisk is 8192.
>>>>>>
>>>>>> The Asterisk VM has been upped to 8GB memory and 4 cores now. There
>>>>>> cannot be network related latencies as the entire test is in the local
>>>>>> network.
>>>>>>
>>>>>> Any insight is deeply appreciated.
>>>>>>
>>>>>
>>>>>
>>>>> ​T​
>>>>> hose errors you're seeing are outgoing connection attempts
>>>>> ​ so there a few things to check...​
>>>>>
>>>>>
>>>>> ​What version of pjproject are you running?  There's an issue in 2.4.5
>>>>> and earlier where TCP sockets aren't being reused​
>>>>> ​.  It's fixed in their trunk.  Unfortunately, you can't use their
>>>>> trunk with Asterisk 13.7.2 because of a new api they introduced.  You'll
>>>>> have to use Asterisk's current 13 branch from git.  If you 're going to do
>>>>> that, check out the bundled pjproject option which also has that patch.
>>>>>
>>>>>
>>>>> Check that sipp isn't terminating early and forcing Asterisk to open a
>>>>> new connection just to send the BYE.​
>>>>> ​
>>>>>
>>>>>
>>>>> ​How many connections are you attempting?
>>>>>
>>>>> If you really want to get the max connections, set the --enable-epoll
>>>>> ​option on pjproject's ./configure line (assuming you're on Linux) and set PJ_IOQUEUE_MAX_HANDLES
>>>>> to 5000 or something.  This won't fix the connection refused messages
>>>>> however.
>>>>>
>>>>>
>>>>>> On Sat, Mar 19, 2016 at 12:09 PM, George Joseph <
>>>>>> george.joseph at fairview5.com> wrote:
>>>>>>
>>>>>>> I'll bet your pjproject install still has the default value of 64
>>>>>>> for PJ_IOQUEUE_MAX_HANDLES.  That limits the number of simultaneous TCP
>>>>>>> sockets.
>>>>>>>
>>>>>>>
>>>>>>> https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Mar 19, 2016 at 9:20 AM, Tickling Contest <
>>>>>>> tickling.contest at gmail.com> wrote:
>>>>>>>
>>>>>>>> I am load-testing an Asterisk 13.7.2 installation with SIPp 3.5.1.
>>>>>>>>
>>>>>>>> I am running into an issue where Asterisk (core set debug 99)
>>>>>>>> complains about the following:
>>>>>>>>
>>>>>>>> [Mar 19 11:12:55] WARNING[13078]: pjsip:0 <?>: tsx0x7fc60840e
>>>>>>>> ...Failed to send Request msg INVITE/cseq=28144 (tdta0x7fc60840b590)!
>>>>>>>> err=70010 (Too many objects of the specified type (PJ_ETOOMANY))
>>>>>>>> [Mar 19 11:12:55] WARNING[13078]: pjsip:0 <?>: tsx0x7fc60841a
>>>>>>>> ...Failed to send Request msg INVITE/cseq=31094 (tdta0x7fc6140ebc10)!
>>>>>>>> err=70010 (Too many objects of the specified type (PJ_ETOOMANY))
>>>>>>>> [Mar 19 11:13:22] ERROR[13083]: pjsip:0 <?>: tcpc0x7fc6082b TCP
>>>>>>>> connect() error: Connection refused [code=120111]
>>>>>>>> [Mar 19 11:13:22] WARNING[13083]: pjsip:0 <?>: tsx0x7fc60822e
>>>>>>>> Failed to send Request msg BYE/cseq=29352 (tdta0x7fc6080e3600)! err=120111
>>>>>>>> (Connection refused)
>>>>>>>> [Mar 19 11:13:25] ERROR[13083]: pjsip:0 <?>: tcpc0x7fc6083a TCP
>>>>>>>> connect() error: Connection refused [code=120111]
>>>>>>>> [Mar 19 11:13:25] WARNING[13083]: pjsip:0 <?>: tsx0x7fc60822e
>>>>>>>> Failed to send Request msg BYE/cseq=13911 (tdta0x7fc6140bc200)! err=120111
>>>>>>>> (Connection refused)
>>>>>>>>
>>>>>>>> Any help figuring out this issue is deeply appreciated.
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>>
>>>>>>>> _______________________________________________
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>>>>>>>> asterisk-app-dev at lists.digium.com
>>>>>>>> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
>>>>>>>>
>>>>>>>>
>>>>>>>
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>>>>>>
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>>>>>
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