Re: [SR-Users] Sending ReINVITE from Kamailio

[Olle E. Johansson]

> On 21 Oct 2015, at 08:31, Grant Bagdasarian  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

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[SR-Users] Sending ReINVITE from Kamailio

[Grant Bagdasarian]

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
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Re: [SR-Users] Sending ReINVITE from Kamailio

[ycaner]

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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Re: [SR-Users] Sending ReINVITE from Kamailio

[Olle E. Johansson]

> On 21 Oct 2015, at 09:27, ycaner  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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Re: [SR-Users] Sending ReINVITE from Kamailio

[Grant Bagdasarian]

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-boun...@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.ca...@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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