[asterisk-users] SIP per-call heartbeat?

[Tony Mountifield]

In article <098FAFC6-CC79-49D4-BDF9-95BEEB4DE1F3 at edvina.net>,
Olle E. Johansson <oej at edvina.net> wrote:
> 
> 24 maj 2011 kl. 12.19 skrev Tony Mountifield:
> 
> > One of our customers has an Asterisk conference bridge connected to a
> > SIP trunk from an ITSP. Yesterday, they had two inbound calls that
> > didn't get hung up properly. From the tcpdump SIP trace that we have
> > running continuously, I can see that no BYE was received by the bridge,
> > and when some hours later the hangup was forced from the bridge end, the
> > bridge sent a BYE to which it received a 481 Call Leg/Transaction Does
> > Not Exist.
> If the remote end send a BYE and doesn't receive a response, that bye will have to be
> retransmitted multiple times before it gives up. The SIP protocol includes retransmission
> over UDP, to cover up for packet loss. If it did not retransmit, you have other issues.

OK, thanks. Sounds like there was some kind of issue at the ITSP then.

> > Since SIP is UDP, this situation must occur from time to time, and I
> > wondered if it is possible to configure any kind of per-call SIP
> > heartbeat so that a dead call could automatically be identified with a
> > 481 response much sooner.
> 
> SIP session timers is what you need for that. Implemented in Asterisk 1.8.

That's useful to know. Planning on moving from 1.2 to 1.8 over the next
few months.

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org