Ahh, right, for tls the reliability of transmission is given by transport layer. So no much to try on this option.>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@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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