[SR-Users] sipp and stateful transaction problem

[JR Richardson]

On Tue, May 25, 2010 at 11:10 AM, Daniel-Constantin Mierla
<miconda at gmail.com> wrote:
> Hello,
>
> On 5/22/10 2:22 AM, JR Richardson wrote:
>>
>> On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla
>> <miconda at gmail.com>  wrote:
>>
>>>
>>> Hello,
>>>
>>> On 5/21/10 10:47 PM, JR Richardson wrote:
>>>
>>>>
>>>> Hi All,
>>>>
>>>> I'm doing some testing with kamailio 1.5:
>>>>
>>>> kamailio1:/etc/kamailio# kamailio -V
>>>> version: kamailio 1.5.4-notls (i386/linux)
>>>> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
>>>> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> svnrevision: 2:6005M
>>>> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
>>>> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2
>>>>
>>>> Using dispatcher module trying to load balance SIP calls across some
>>>> Asterisk servers.  I have it working fine when I test in this
>>>> scenario:
>>>>
>>>> sip phone dial out><asterisk><kamailio><round robin to several asterisk
>>>> servers
>>>>
>>>> This works stateful and stateless, handles everything gracefully.
>>>>
>>>> This scenario is giving me fits:
>>>>
>>>> sipp dial out><kamailio><round robin to several asterisk servers
>>>>
>>>> I get retransmits on every call back to sipp with errors like
>>>> "Discarding message which can't be mapped to a known SIPp call" and
>>>> "SIP/2.0 481 Call leg/transaction does not exist"
>>>>
>>>> This happens with kamailio setup stateful or stateless.  I'm wondering
>>>> if sipp is the problem or just doesn't play well with kamailio?
>>>>
>>>> I've kept the config as simple as possible for testing, it is listed
>>>> here http://pastebin.com/BZ8hJvJv
>>>>
>>>> Here is my sipp usage:
>>>>
>>>> sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1
>>>> -trace_err
>>>>
>>>> Any insight would be appriciated.
>>>>
>>>>
>>>>
>>>
>>> the problem is in your sipp scenario. The uac calls do not map to uas.
>>> kamailio does not reply 481,  check the uas scenario, that is the one
>>> that
>>> sends back the 481.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> Kamailio (OpenSER) Advanced Training
>>> Miami, Fl, USA - June 21-23, 2010
>>> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>
>>>
>>>
>>
>> Thanks Daniel, I reveiwed the sipp docs, '-sn uas' just sits there as
>> a responder, it will not initiate calls to kamailio.  I don't
>> understand what you are getting at?  How would I use this type of
>> scenario to test?
>>
>
> keeping the mailing list cc-ed is recommended, since others can respond
> faster and new people can benefit of the discussion.
>
> What I wanted to say is that kamailio does not reply 481. So the problem is
> in the responder of requests sent by UAC and forwarded by Kamailio. Somehow,
> the dialog is destroyed before the BYE (or other in-dialog request) is sent
> by UAS.
>
> If you can grab the SIP trace of such call (e.g., using ngrep on kamailio
> server), I can give more hits (try to select the sip flow for one such call
> only, sending full sip trace will be too big).
>
> Cheers,
> Daniel
Thanks Daniel.  I've moved on from the Dispatcher and now working with
the carrierroute.  The first thing I notice is the transactions, being
stateful has resolved the issue I was seeing with the dispatcher and
sipp sending invites.  So and I don't have the dispatcher setup to
collect sip traces, best I can tell is either my configuration or the
lack of stateful tracking in the dispatcher module was the cause of my
original post.

I'm going to move the discussion along to a new topic for now, thanks.

JR
-- 
JR Richardson
Engineering for the Masses