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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.<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 11/08/14 13:10, Dmytro Bogovych
wrote:<br>
</div>
<blockquote
cite="mid:CAJrBzoX7prAH9iXAmzVbrAZL8H_7jUWW+Z00pgfWO7Znb7YAnw@mail.gmail.com"
type="cite">
<div dir="ltr">Greetings all.
<div>I have following use case:</div>
<div><br>
</div>
<div>0) Peer A and B registers on kamailio via TLS.</div>
<div>1) Peer A establishes call to peer B.</div>
<div>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.</div>
<div>3) Peer B receives reINVITE and sends answer.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>Is there way to solve or avoid this situation?</div>
<div><br>
</div>
<div>Thank you :)</div>
</div>
<br>
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<br>
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
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