[OpenSIPS-Users] help on failover routing

[johan]

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
>