[SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

[Brett Woollum]

Hi Alex, 

Thanks for this information. I've started researching the session-timer capabilities in Asterisk, and I think that's my solution. I've already implemented it on a test system and it works well, except that it's using reINVITES to update as opposed to UPDATE messages, resulting in chops in the audio every so often. I'll research this further though. 

Thanks again! 
Brett 

----- Original Message ----- 
From: "Alex Balashov" <abalashov at evaristesys.com> 
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users at lists.sip-router.org> 
Cc: sr-users at lists.sip-router.org 
Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint 


This is a complex topic. There is no way for a proxy like Kamailio to detect this scenario per se. Kamailio reacts to and forwards signaling events. If an endpoint disappears, it won't send any of those to indicate that it has gone away. How would Kamailio know? Media stream timeout? Kamailio doesn't relay media. 


Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module. But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful. 


Your endpoint solution is SIP Session Timers. 

-- 
Alex Balashov - Principal 
Evariste Systems LLC 
260 Peachtree Street NW 
Suite 2200 
Atlanta, GA 30303 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/ 

On Jun 23, 2011, at 1:10 AM, Brett Woollum < brett at woollum.com > wrote: 







Hello, 

We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb . Everything seems to be working well, until today. 

I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. 

If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. 

What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. 

Thanks! 
Brett 



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