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

Thu Jun 23 09:49:00 CEST 2011

Hi Carsten, 

Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. 

I will look into that patch. 

Thanks! 

Brett 

----- Original Message ----- 
From: "Carsten Bock" <carsten at ng-voice.com> 
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users at lists.sip-router.org> 
Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint 

Hi, 

another solution might be, to either configure an RTP-Timeout on the 
Asterisk (if you send your calls through the asterisk anyway). 
You might also consider using the RTPProxy with the patch in the 
sip-router-repository. With the patch, the RTPProxy will trigger a 
teardown of calls (via XML-RPC) if the RTP-Session has a timeout. 

Carsten 

2011/6/23 Brett Woollum <brett at woollum.com>: 
> 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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-- 
Carsten Bock 
http://www.ng-voice.com 
mailto:carsten at ng-voice.com 

Schomburgstr. 80 
22767 Hamburg 
Germany 

Mobile +49 179 2021244 
Office +49 40 34927219 
Fax +49 40 34927220 

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