[asterisk-users] SIP per-call heartbeat?

[Paul Hayes]

On 24/05/11 12:08, Tony Mountifield wrote:
>
> OK, thanks. Sounds like there was some kind of issue at the ITSP then.
>

I have seen this happen with broken SIP-ALGs in routers too.  The ITSP 
sends the BYE but for some reason a broken SIP ALG will not deliver the 
packet to the right place.  The ITSP will resend the BYE several times 
if they don't receive the responding OK message (or some error such as 
481 etc....) but after a few attempts it's pointless them continuing.

>>> 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.
>

Setting absolute timeouts on all calls might help too:

http://www.the-asterisk-book.com/unstable/funktionen-timeout.html

Although it's a bit of a balancing act, it can be used to limit these 
things to a couple of hours rather than having "stuck" calls going on 
for days.


cheers,
Paul.

> Cheers
> Tony