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

Joshua C. Colp jcolp at sangoma.com
Tue Dec 21 08:30:36 CST 2021

On Tue, Dec 21, 2021 at 10:28 AM Kingsley Tart <kingsley at dns99.co.uk> wrote:

> 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?

Allow traffic from specific IP addresses? Others may have better input or
guidance on such a situation.

Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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