[OpenSIPS-Users] help on failover routing

[Giovanni Maruzzelli]

Liviu,

while you are at it (feature deleting record if TLS connection is down), do
it for wss too :))




On Fri, Mar 20, 2020 at 2:06 PM johan <johan at democon.be> wrote:

> as for point 3, I will check.
>
>
> On 20/03/2020 11:43, Liviu Chircu wrote:
> > On 20.03.2020 12:37, johan wrote:
> >>
> >> Hence,
> >>
> >> - when the softphone is registered, a call comes on that DID in udp
> >> (we do lookup_location) and we send it to the user in tls (this works)
> >>
> >> - when the softphone is off for a long time, there is no record in
> >> location so then I route the call via the provider to his real mobile
> >> number (this works also)
> >>
> >> - the problem is when the mobile looses his dataconnection, then I do
> >> have a record in location, I try to send the call, which will fail.
> >> Upon failure, I drop the record in subscriber. And here the problem
> >> begins.
> >>
> >> The invite is adapted at this point already for tls => provider
> >> doesn't want it as he is udp.
> >>
> >>
> >> So how can I have the original request back for routing to the real
> >> mobile number ? Or how can I check if the user is still connected
> >> (aka how can I send options to see if he's alive) before calling
> >> t_relay.
> >
> > Hi, Johan!
> >
> > 1.  this solution of calling remove() after a routing failure is
> > nice.  Alexey Vasilyev put together a feature request [1] related to
> > this problem, where he asks for an automated mechanism of deleting a
> > contact whenever its TLS connection is found to be dead.
> >
> > 2.  Did you try to force the sending socket of the INVITE ($fs
> > variable) to your "udp:1.2.3.4:5060" listener?  I think this should
> > work inside a failure_route and should properly route to your provider
> > via UDP.  Also, I believe Bogdan fixed this recently [2] (but master
> > branch only?!), such that "$fs" is not set to the TLS listener inside
> > failure_route - might wanna check.
> >
> > 3.  As a long-term solution to this problem, we are working on adding
> > RFC 8599 Push Notification support via SIP in OpenSIPS 3.1.  The spec
> > is still rather new, but I'm curious if your app's SIP stack supports
> > it :)  Basically, this will allow you to wake up the phone so it
> > re-registers whenever you need to deliver an INVITE to it, in a
> > standards-approved way.
> >
> > Best regards,
> >
> > [1]: https://github.com/OpenSIPS/opensips/issues/1769
> >
> > [2]: https://github.com/OpenSIPS/opensips/commit/f73abff9
> >
> > [3]: https://tools.ietf.org/html/rfc8599
> >
>
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>


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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