[SR-Users] sipp and stateful transaction problem

[JR Richardson]

On Mon, May 24, 2010 at 2:33 PM, Klaus Darilion
<klaus.mailinglists at pernau.at> wrote:
>
>
> On 21.05.2010 23:46, Daniel-Constantin Mierla 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
>
> what means "call back"?

sipp send invite to kamailio which forwards to asterisk in dispatcher
list, asterisk responds
back to kamailio which forwards that response back to sipp and I get the error:
SIP/2.0 481 Call leg/transaction does not exist on sipp.

So I think this is not supposed to work like I want it to.  The
dispatcher module is for
stateless processing only, so even though I have  RR and TM functions
in my routing script
it does not act properly.  I don't think I can use dispatcher for what
I want, which is a
stateful load balancer.

I am looking at 3.0 and carrierroute or lcr module.

Thanks.

JR


>
> You are operating sipp in uac mode - thus it is not capable of receiving
> requests.
>
> Maybe Asterisk is send reINVITEs which are not handled correctly by sipp.
> set canreinvite=no in sip.conf (Asterisk)
>
> regards
> klaus
>
>>> "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
>>
>



-- 
JR Richardson
Engineering for the Masses