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@evaristesys.com To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" sr-users@lists.sip-router.org Cc: sr-users@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.
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@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@evaristesys.com To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" sr-users@lists.sip-router.org Cc: sr-users@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@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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Brett Woollum -
Carsten Bock