Hello,
Is it possible to have Kamailio send a ReINVITE every X minutes to determine if a session is still active? I know it's a proxy and doesn't have B2BUA capabilities, but there was a module which allowed Kamailio to generate SIP messages, but I can't find it anymore. If Kamailio is not the place to do this, which component in the voip network should be responsible for this? Are there perhaps other ways to poll for session state in Kamailio?
Regards,
Grant
On 21 Oct 2015, at 08:31, Grant Bagdasarian gb@cm.nl wrote:
Hello,
Is it possible to have Kamailio send a ReINVITE every X minutes to determine if a session is still active? I know it’s a proxy and doesn’t have B2BUA capabilities, but there was a module which allowed Kamailio to generate SIP messages, but I can’t find it anymore. If Kamailio is not the place to do this, which component in the voip network should be responsible for this? Are there perhaps other ways to poll for session state in Kamailio?
The dialog module has in-dialog keepalives. Not Re-INVITE, but at least a message.
The best way is to use SIP Session Timers in both or at least one user agent.
/O
Hello; I think Dialog module can do it with ka_timer. take a look please. in addition , if you want to know call is still up , check the RTP session. if there isn't Rtp session , so call is hung up. Asterisk can listen rtp packet and then in silence it can close session.
have a look "rtptimeout" parameter
-- View this message in context: http://sip-router.1086192.n5.nabble.com/Sending-ReINVITE-from-Kamailio-tp142... Sent from the Users mailing list archive at Nabble.com.
On 21 Oct 2015, at 09:27, ycaner yasin.caner@netgsm.com.tr wrote:
Hello; I think Dialog module can do it with ka_timer. take a look please. in addition , if you want to know call is still up , check the RTP session. if there isn't Rtp session , so call is hung up. Asterisk can listen rtp packet and then in silence it can close session.
have a look "rtptimeout" parameter
This doesn’t always apply either - if the call is on hold there’s no RTP but it should not be hung up. Asterisk handles this, but for other proxys it’s hard to know the state of the media session.
/O
Thanks for the input.
I'll try the Session Timers and the ka_timer param from the dialog module.
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Olle E. Johansson Sent: Wednesday, October 21, 2015 9:30 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Subject: Re: [SR-Users] Sending ReINVITE from Kamailio
On 21 Oct 2015, at 09:27, ycaner yasin.caner@netgsm.com.tr wrote:
Hello; I think Dialog module can do it with ka_timer. take a look please. in addition , if you want to know call is still up , check the RTP session. if there isn't Rtp session , so call is hung up. Asterisk can listen rtp packet and then in silence it can close session.
have a look "rtptimeout" parameter
This doesn’t always apply either - if the call is on hold there’s no RTP but it should not be hung up. Asterisk handles this, but for other proxys it’s hard to know the state of the media session.
/O _______________________________________________ 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
-
Grant Bagdasarian -
Olle E. Johansson -
ycaner