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

Carsten Bock carsten at ng-voice.com
Thu Jun 23 09:46:11 CEST 2011


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

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