[SR-Users] Routing for case of client's changing networks

[Dmytro Bogovych]

>You can try not sending the 100 trying, so the sender does retransmissions
Sender did not receive 100 answer; anyway it did not retransmit. It is
based on resiprocate.

>maybe the sender should send a new re-invite as it changes to a new
network again.
Is it legal in SIP? I ask because such attempt violates resiprocate's
invite session state machine rules; resiprocate will throw exception.



On Mon, Aug 11, 2014 at 3:17 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  The SIP reply is routed based on the address in the VIA header. The
> address in VIA is set by each node sending a request, if between the
> request and the response the sender is no longer available at the address
> it added as VIA, then it is little one can do for that.
>
> You can try not sending the 100 trying, so the sender does
> retransmissions. Also, maybe the sender should send a new re-invite as it
> changes to a new network again.
>
> Cheers,
> Daniel
>
>
> On 11/08/14 13:10, Dmytro Bogovych wrote:
>
> Greetings all.
> I have following use case:
>
>  0) Peer A and B registers on kamailio via TLS.
> 1) Peer A establishes call to peer B.
> 2) Peer A changes network and call has to be refreshed. Peer A
> reregisters, gathers ICE candidates, builds reINVITE and sends it to B.
> After peer A changes network again, reregisters and waits for answer from
> old reINVITE.
> 3) Peer B receives reINVITE and sends answer.
>
>  The problem is this answer never gets to peer B.  Kamailio cannot route
> answer as old TLS connection is closed and new one did not established
> before reregistering.
>
>  Is there way to solve or avoid this situation?
>
>  Thank you :)
>
>
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>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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