Kingsley Tart
2021-Dec-21 14:27 UTC
[asterisk-users] PJSIP keepalive only while calls are present
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.
Joshua C. Colp
2021-Dec-21 14:30 UTC
[asterisk-users] PJSIP keepalive only while calls are present
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20211221/995e304a/attachment.html>