Re: [SR-Users] sipp and stateful transaction problem
On Tue, May 25, 2010 at 11:10 AM, Daniel-Constantin Mierla mico...@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 mico...@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 outasteriskkamailioround robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial outkamailioround 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] sipp and stateful transaction problem
On Mon, May 24, 2010 at 2:33 PM, Klaus Darilion klaus.mailingli...@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 outasteriskkamailioround robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial outkamailioround 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] sipp and stateful transaction problem
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 outasteriskkamailioround robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial outkamailioround 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. Thanks. JR -- JR Richardson Engineering for the Masses ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
