[asterisk-users] PJSIP keepalive only while calls are present
kingsley at dns99.co.uk
Tue Dec 21 08:27:45 CST 2021
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.
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
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?
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