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<p>Happy New Year everyone.</p>
<p><br>
</p>
<p>I haven't solved this problem yet. Although is it not critical,
it is a bit annoying.</p>
<p>I have tried to simplify things, and have a reference setup that
works. <br>
My linphone sends a TCP call and my Asterisk answers, plays a
speak and hangs up. <br>
</p>
<p><br>
</p>
<p>If I instead sends the call to my PBX, which handles the
authentication via UAC, it fails with this error, which the
customer site also generated. </p>
<blockquote>
<p>Status-Line: SIP/2.0 477 Unfortunately error on sending to next
hop occurred (477/SL)<br>
</p>
</blockquote>
<p>Unfortunately the error is not generated by my Kamailio, but by a
Kamailio at the provider, when it gets a Bye forwarded via their
SBC. <br>
</p>
<p><br>
</p>
<p>I have attached a capture which the provider send me. This is the
setup</p>
<blockquote>
<p>linphone -> My Kamailio PBX (194.255.22.44:36089) ->
provider Kamailio(194.247.61.26) -> SBC(194.247.61.32) ->
provider Kamailio(194.247.61.26) -> my Asterisk (194.255.22.44:45075)</p>
</blockquote>
<p>A note on the providers Kamailio. It listens on both port 5060
and 5070, and both UDP/TCP. <br>
It is also used as access point for both my PBX and my Asterisk,
thus the trace may be a little confusing to read.<br>
</p>
<p>As far as I can see, the provider Kamailio gets the correct
To/From and CallID in the bye. Thus it should be able to match the
TCP connection. <br>
The flow is also clean, there is no change of ports etc. <br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>I have this reference setup which works</p>
<blockquote>
<p>linphone -> provider Kamailio -> SBC -> provider
Kamailio -> my Asterisk</p>
</blockquote>
The only differences towards the reference I can see these. I do not
have a capture from the provider on this. <br>
<ul>
<li>There is an extra Via step.</li>
<li>Contact points to the Linphone IP, not the Kamailio IP</li>
</ul>
<p>Any hint will be appreciated.</p>
<p><br>
</p>
<p><br>
</p>
<pre class="moz-signature" cols="72">-------------------- Med Liberalistiske Hilsner ----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - <a class="moz-txt-link-abbreviated" href="http://www.liberalismen.dk">www.liberalismen.dk</a></pre>
<div class="moz-cite-prefix">On 11/9/20 12:06 PM, Daniel-Constantin
Mierla wrote:<br>
</div>
<blockquote type="cite"
cite="mid:a4fe56e7-2d76-2eae-1d0a-fc624ba1d528@gmail.com">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<p>Hello,</p>
<p>there is no association between a SIP call and a TCP
connection. SIP is not designed on TCP streams, the forwarding
is based on the headers. It doesn't matter if there are messages
belonging to same call or not, they can share same connection,
or can open a new one...<br>
</p>
<p>The BYE from caller gets to 194.247.61.32:5040, which cannot
deliver it further based on Route header. The system at
194.247.61.26:5070 must be able to accept connections on
advertised port of the Route address. Again, connection
interruption can happen from various cases, it cannot rely on
ephemeral ports, but on what the SIP system advertises as
"listen" address.</p>
<p>One can play with tcp port aliases, look at Kamailio core
cookbook, in case 194.247.61.32:5040 can do that. But that is
not the proper way for server to server communication, there
will be cases when the connection will be cut for various
reasons (can be also the IP routes in the path that get
congested).</p>
<p>Otherwise, you can follow the code of tcp_send() function in
the tcp_main.c from core to see how tcp connection is matched,
there are various cases there, also a matter of the config
parameters.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div class="moz-cite-prefix">On 09.11.20 10:13, Kjeld Flarup
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CACRoYp-Eknw2=8X-N+Wrs+aapDqLwZYj-WGCHiZVP-mcQXCmfg@mail.gmail.com">
<meta http-equiv="content-type" content="text/html;
charset=UTF-8">
<div dir="ltr">Hello<br>
<br>
I have attached a pcap received from the provider.<br>
<br>
It is quite informative as it shows bits of how they forward
the call.<br>
<br>
We send to 194.247.61.26 which is a Kamailio proxy, that
forwards the call to a SBC 194.247.61.32<br>
<br>
My guess is that the 194.247.61.26 kamailio gets confused,
and does not match the BYE with the ongoing TCP session.<br>
Instead it sees it as a new session, and forwards it according
to the route information.<br>
<br>
Can anybody help explaining what fields Kamailio uses to match
an ongoing TCP session.<br>
<br>
Regards Kjeld<br>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">Den fre. 6. nov. 2020 kl.
10.50 skrev Daniel-Constantin Mierla <<a
href="mailto:miconda@gmail.com" moz-do-not-send="true">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>
<div>Hello,</div>
<div><br>
</div>
<div>from SIP specs point of view, can be any port -- ACK
and BYE do not have to follow same path as INVITE, so
they can even come from a different IP.<br>
</div>
<div><br>
</div>
<div>Then, the call can be closed after 30secs because
also the ACK has the same problems with the header as
the BYE. Your pcap didn't include all the traffic, you
have to capture both directions on your kamailio server
to see what happens on each side.</div>
<div><br>
</div>
<div>Cheers,<br>
Daniel<br>
</div>
<div> </div>
<div><br>
</div>
<div><br>
</div>
<div>On 06.11.20 10:35, Kjeld Flarup wrote:<br>
</div>
<blockquote type="cite">
<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" target="_blank"
moz-do-not-send="true">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 moz-do-not-send="true"><sip:32221660@194.255.22.44:5071></a>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.<br>
To: <a moz-do-not-send="true"><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 moz-do-not-send="true"><sip:u0@192.168.2.9></a>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.<br>
To: <a moz-do-not-send="true">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" moz-do-not-send="true">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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-- <br>
<div dir="ltr">
<div dir="ltr">
<p>--------------------- Med
Liberalistiske Hilsner
----------------------</p>
<pre cols="72"> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - <a href="http://www.liberalismen.dk" target="_blank" moz-do-not-send="true">www.liberalismen.dk</a>
</pre>
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<br>
<fieldset></fieldset>
<pre>_______________________________________________
Kamailio (SER) - Users Mailing List
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</pre>
</blockquote>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank" moz-do-not-send="true">www.asipto.com</a>
<a href="http://www.twitter.com/miconda" target="_blank" moz-do-not-send="true">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" target="_blank" moz-do-not-send="true">www.linkedin.com/in/miconda</a>
Funding: <a href="https://www.paypal.me/dcmierla" target="_blank" moz-do-not-send="true">https://www.paypal.me/dcmierla</a></pre>
</div>
</blockquote>
</div>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr">
<div dir="ltr">
<p>--------------------- Med Liberalistiske
Hilsner ----------------------</p>
<pre cols="72"> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - <a href="http://www.liberalismen.dk" target="_blank" moz-do-not-send="true">www.liberalismen.dk</a>
</pre>
</div>
</div>
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<br>
<fieldset></fieldset>
<pre>_______________________________________________
Kamailio (SER) - Users Mailing List
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</pre>
</blockquote>
<p><br>
</p>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank" moz-do-not-send="true">www.asipto.com</a>
<a href="http://www.twitter.com/miconda" target="_blank" moz-do-not-send="true">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" target="_blank" moz-do-not-send="true">www.linkedin.com/in/miconda</a>
Funding: <a href="https://www.paypal.me/dcmierla" target="_blank" moz-do-not-send="true">https://www.paypal.me/dcmierla</a></pre>
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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
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - <a href="http://www.liberalismen.dk" target="_blank" moz-do-not-send="true">www.liberalismen.dk</a>
</pre>
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<pre class="moz-quote-pre" wrap="">_______________________________________________
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</blockquote>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com" moz-do-not-send="true">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda" moz-do-not-send="true">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda" moz-do-not-send="true">www.linkedin.com/in/miconda</a>
Funding: <a class="moz-txt-link-freetext" href="https://www.paypal.me/dcmierla" moz-do-not-send="true">https://www.paypal.me/dcmierla</a></pre>
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