21 Oct
2015
21 Oct
'15
9:31 a.m.
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
21 Oct
21 Oct
10:26 a.m.
On 21 Oct 2015, at 08:31, Grant Bagdasarian
<gb(a)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
10:27 a.m.
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
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10:30 a.m.
On 21 Oct 2015, at 09:27, ycaner
<yasin.caner(a)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
11:17 a.m.
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(a)lists.sip-router.org>
Subject: Re: [SR-Users] Sending ReINVITE from Kamailio
On 21 Oct 2015, at 09:27, ycaner
<yasin.caner(a)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
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4 comments
3 participants
participants (3)
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Grant Bagdasarian -
Olle E. Johansson -
ycaner