[SR-Users] Kamailio - Asterisk: Handling loss of SIP BYE and dangling channels

[Sergey Okhapkin]

RTP timeout in asterisk is the best place to handle the situation. Another 
option is SIP session timer, but it could give false negatives with NATed 
clients.

On Friday 08 January 2016 11:56:51 Benjamin Fitzgerald wrote:
> Hi,
> 
> I'm wondering what the best approach to handling a SIP dialog when one
> endpoint disappears/fails to send the BYE message.
> 
> I have Kamailio as a proxy for all mobile (iPhone/Android) SIP clients.
> Occasionally, the user hangs up the call but no BYE message is received.
> This means that Asterisk has an open channel even though there is no
> client. Kamailio also continues to receive successful registrations from
> the SIP client so the endpoint is not down completely.
> 
> Is Kamailio the appropriate place to handle this situation? What do you
> recommend? If not could you point me in the right direction? RTP timeout?
> Asterisk? The SIP client itself?
> 
> Thanks for your help.
> 
> Benjamin Fitzgerald
> LETS Corporation
> (925) 235-1154
> ben at letscorp.us