<div dir="ltr"><div dir="ltr">Hi Daniel<div><br></div><div>The Unknown Dialog comes because the server hang up the call 30 seconds earlier. We never gets these BYE messages, thus the door phone hangs times out and hangs up.</div><div><br></div><div>My question is still, which port is the BYE from the server supposed to be sent to?</div><div><br></div><div>The original 37148</div><div>The new 37150</div><div>or the advertised 5071</div><div><br></div><div>Regards Kjeld</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">Den fre. 6. nov. 2020 kl. 10.18 skrev Daniel-Constantin Mierla <<a href="mailto:miconda@gmail.com">miconda@gmail.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<p>Hello,</p>
<p>I think you hunt a mirage problem here by looking at the ports of
tcp connections, if you think that being different ports is the
cause of BYE failure. The ACK fpr 200ok is independent of the
INVITE transaction and can have a completely different path than
INVITE, thus is completely valid to use another connection. Of
course, if follows the same path as INVITE, if the connection is
still open, it should be reused. But is a matter of matching, it
can be that the INVITE uses different destination identifiers or
the connection gets cut from different reasons. But the point is
that even if there is a different connection, it should work.</p>
<p>So, I actually looked at the pcap capture you sent in one of your
previous emails and the BYE is sent out, but gets back:</p>
<p>SIP/2.0 481 Unknown Dialog.</p>
<p>Therefore it gets to the end point, which doesn't match it with
any of its active calls. Looking at the headers, the 200ok/INVITE
has:</p>
<p>From: "Front Door"
<a><sip:32221660@194.255.22.44:5071></a>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.<br>
To: <a><sip:004540294149@127.0.0.1:5071></a>;tag=12003375157297.<br>
Call-ID: ***FgXBdt966gypC5V1T5VGUtLILtzxsJJ57NRSL5UMUiq*.</p>
<p>And the BYE:</p>
<p>From: "Front Door"
<a><sip:u0@192.168.2.9></a>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.<br>
To:
<a>sip:195.249.145.198:5060;transport=udp;line=sr-z-yMngm27FwI73qx0CQo6gm2n3ao03LMn5UILt2NziWIO3ooTDc*;tag=12003375157297</a>.<br>
Call-ID: ***FgXBdt966gypC5V1T5VGUtLILtzxsJJ57NRSL5UMUiq*.</p>
<p>While the dialog should be matched on call-id, from/to-tags, the
From/To URI should be the same to be strict conformant with
RFC3261 (that mandates unchanged From/To for backward
compatibility with RFC2543). Likely you do some From/To header
changes that are not done correctly to update/restore the values
for traffic within dialog.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div>On 06.11.20 09:31, Kjeld Flarup wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Thanks Juha
<div><br>
</div>
<div>That makes it somehow easier to understand my capture. My
Kamailio must then have detected a broken TCP connection,
though I cannot see why in the capture, neither in the log,
but I only run on debug level 2.</div>
<div>It receives a 200 OK on port 37148, and then establishes
37150 to reply with an ACK. </div>
<div><br>
</div>
<div>However two seconds before receiving the 200 OK, there are
some spurious retransmissions and out of order on 37148.
Perhaps this has caused Kamailio to deem the connection bad,
but it still receives data on it.</div>
<div>Now I assume that the providers server (Which also is
flying Kamailio) should detect the new port, and continue
using that. I got a trace from the provider, where there is no
disturbance. Thus the server thinks that the connection is
OK. </div>
<div><br>
</div>
<div>Now my next question is, what makes a Kamailio detect this
change?</div>
<div>Is it a problem that I only rewrite To and From in the
INVITE, thus the ACK contains some other values. <br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>It is also a bit strange that I get this error exactly, the
same place in the conversation every time I make a call.
Somehow I suspect some NAT timeout in the router. (it is not
carrier grade NAT).</div>
<div>Can I do anything to prevent a NAT timeout from the client
side?</div>
<div><br>
</div>
<div><br>
</div>
<div>Another thing. I assume that sending my internal port in
the From field, or any kind of advertising, should be ignored
by the server.</div>
<div><br>
</div>
<div>Regards Kjeld</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">Den fre. 6. nov. 2020 kl.
07.45 skrev Juha Heinanen <<a href="mailto:jh@tutpro.com" target="_blank">jh@tutpro.com</a>>:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Kjeld
Flarup writes:<br>
<br>
> How is TCP SIP actually supposed to handle a BYE, when
the client is <br>
> behind NAT.<br>
<br>
Client behind NAT is supposed to keep its TCP connection to
SIP Proxy<br>
alive and use it for all requests of the call. If the
connection breaks<br>
for some reason, the client sets up a new one for the
remaining<br>
requests.<br>
<br>
-- Juha<br>
<br>
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<pre cols="72"> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
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<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">www.asipto.com</a>
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</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature"><div dir="ltr"><p>--------------------- Med Liberalistiske Hilsner ----------------------</p><pre cols="72"> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
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