[SR-Users] sipp and stateful transaction problem

Daniel-Constantin Mierla miconda at gmail.com
Tue May 25 18:10:27 CEST 2010


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

--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/





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