[asterisk-users] PJSIP keepalive only while calls are present

[Kingsley Tart]

On Tue, 2021-12-21 at 09:45 -0400, Joshua C. Colp wrote:
> No. Session timers on the endpoint is the closest thing to making
> sure a call is active and keeping things open but does not use
> OPTIONS. Note that if you're sending calls to them, then without
> OPTIONS outside of calls any NAT mapping would go away unless they
> re-register frequently. If they did re-register frequently then you
> likely wouldn't need either.

Hi,

the example I'm testing with is with sending a call to Twilio.

SIP timers look perfect for this, except that after the first refresh,
Twilio turns them off :(

What I'm seeing is after a minute we send a re-invite with these
headers:

Session-Expires: 120;refresher=uac.
Min-SE: 90.

and the 200 OK coming back from Twilio omits them. Asterisk then
doesn't send any more.

This is not very helpful of them. If the callee hangs up after a while,
our system doesn't notice because our firewall blocks the BYE. We can't
leave these servers open to the world so need somehow to find a way of
keeping the firewall open for any calls we send out.

Any idea how we might solve that?

-- 
Cheers,
Kingsley.