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ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

1 0

Issue using pua_bla module
by Watkins, Bradley 31 May '11

31 May '11

I'm trying to get the pua_bla module to do its thing, and I'm running into an issue. I'm getting these errors when the first NOTIFY comes from the phone: ERROR: presence [publish.c:515]: Missing or unsupported event header field value ERROR: presence [publish.c:518]: event=[dialog;sla] I see the phone register and the subsequent SUBSCRIBE (both successful) from Kamailio, so this bit seems to be correct. I'm loading what I think are all the necessary modules: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "pua.so" loadmodule "pua_bla.so" And I'm handling the NOTIFY as per the documentation (xlogs to see that in fact the script executes): if(is_method("NOTIFY") && to_uri=~"bla_aor(a)domain.com") { xlog("Matched bla_aor to_uri\n"); if(bla_handle_notify()) { xlog("Handled NOTIFY\n"); t_reply("200", "OK"); } } I see the log entries from the above xlogs, so it certainly is executing this part of the script. Any ideas why I'm getting the errors? All ideas and/or assistance are greatly appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 6

Siremis: changes to Kamailio DB
by Klaus Darilion 31 May '11

31 May '11

Hi! I just read the Siremis installation howto. One of the task of the installer is: "Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules" I wonder what siremis exactly does to the DB. Is greped over the siremis source file but couldn't find the changes. Are they somewhere documented? (e.g. .sql file with the statemens). Anyway, I wonder changes are needed at all. IMO, if possible, the Kamailio DB should not be changed by an external too - otherwise this may cause problems with updates to the Kamailio DB structure. Thanks Klaus

3 3

kamailio debian package bug report : 3.1.3+lenny2
by Philippe HENSEL 31 May '11

31 May '11

Hi developpers and packagers, The file "/usr/lib64/kamailio/libsrdb2.so.1.0" seems to be included in both "kamailio_3.1.3+lenny2_amd64.deb" and "kamailio-mysql-modules_3.1.3 +lenny2_amd64.deb". So it is not possible to install "kamailio-mysql-modules". Thanks for help

5 8

Multidomains - quick question
by Lucas Alvarez 31 May '11

31 May '11

Hi, if I want to register clients from outside of my local network, do I have to enable the domain module and add the external IP as a new domain? Thanks in advance, Lucas Alvarez

2 2

acc Locks
by Borin 31 May '11

31 May '11

Hi all, Recently I met the following problem with kamailio acc table. We are using kamailio for calls dispatching to a couple of rtp gateways and for call statistics acc is used (with mysql). I tried to delete some old entries from acc table and for this time mysql "show processes" showed a lot of entries like " kamailio | Query | 0 | System lock | insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,start_time,src_addr,dst_addr,src_ |". I think that meant that kamailio could not write to the acc table for the time the data were being deleted. And it also did not forward INVITE requests further. Does that mean that if kamailio does not manage to write acc record for any reason it does not forward that INVITE/200ok? Best Regards, Kate

2 4

Green VoIP - energy efficiency and performaces of v3.0
by Daniel-Constantin Mierla 31 May '11

31 May '11

Hello, Jan Janak conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University. The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0. As part of the research work, Jan could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file: etc/sip-router-oob.cfg (involving authentication and NAT traversal as well). You can read the paper about energy efficiency at: - Green VoIP Article: http://asipto.com/u/2j The draft notes about capacity and performances of v3.0 are available at: - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k Some interesting results: - one instance of SIP server with 500 000 online users (mixed users – behind and not NAT routers) – consumed energy 210W - one instance of SIP server with 1 000 000 online users (no NAT involved) – consumed energy 190W - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 203W - one SIP server instance with 80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%. I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

5 9

Late parallel branch (without canceling previous pending branches)
by Iñaki Baz Castillo 31 May '11

31 May '11

Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X: 1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Dispatcher variables
by Lee Archer 31 May '11

31 May '11

Hi all, I am using dispatcher and on a failure I would like to log the server that has failed. I am using $dd but this only ever returns the address of the first server that was tried and not any subsequent servers that were tried and also failed. What should I be using to display the IP address of the server dispatcher is about to use? Thanks Lee

2 2

Fw: Load capacity of SER??
by Piyush.Bansal＠relianceada.com 31 May '11

31 May '11

Dear All, I'm using SER as a SIP proxy server for quite a long time. I'm not been able to find its load bearing capacity. If you can help me get a figure in terms of BHCA, that'll be quite helpful. Thanks and Regards, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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