New subject: Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

23 Jun 2011

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" &lt;abalashov(a)evaristesys.com&gt; 
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing
List" &lt;sr-users(a)lists.sip-router.org&gt; 
Cc: sr-users(a)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(a)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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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint