Hello,

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.

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.

Cheers,
Daniel

On 06.11.20 10:35, Kjeld Flarup wrote:

Hi Daniel

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.

My question is still, which port is the BYE from the server supposed to be sent to?

The original 37148

The new 37150

or the advertised 5071

Regards Kjeld

Den fre. 6. nov. 2020 kl. 10.18 skrev Daniel-Constantin Mierla <miconda@gmail.com>:

Hello,

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.

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:

SIP/2.0 481 Unknown Dialog.

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:

From: "Front Door" <sip:32221660@194.255.22.44:5071>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.
To: <sip:004540294149@127.0.0.1:5071>;tag=12003375157297.
Call-ID: ***FgXBdt966gypC5V1T5VGUtLILtzxsJJ57NRSL5UMUiq*.

And the BYE:

From: "Front Door" <sip:u0@192.168.2.9>;tag=thm9OFNQemH0IsqgRR1jDGF4rjVivTOK.
To: sip:195.249.145.198:5060;transport=udp;line=sr-z-yMngm27FwI73qx0CQo6gm2n3ao03LMn5UILt2NziWIO3ooTDc*;tag=12003375157297.
Call-ID: ***FgXBdt966gypC5V1T5VGUtLILtzxsJJ57NRSL5UMUiq*.

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.

Cheers,
Daniel

On 06.11.20 09:31, Kjeld Flarup wrote:

Thanks Juha

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.

It receives a 200 OK on port 37148, and then establishes 37150 to reply with an ACK. 

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.

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. 

Now my next question is, what makes a Kamailio detect this change?

Is it a problem that I only rewrite To and From in the INVITE, thus the ACK contains some other values. 

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).

Can I do anything to prevent a NAT timeout from the client side?

Another thing. I assume that sending my internal port in the From field, or any kind of advertising, should be ignored by the server.

Regards Kjeld

Den fre. 6. nov. 2020 kl. 07.45 skrev Juha Heinanen <jh@tutpro.com>:

Kjeld Flarup writes:

> How is TCP SIP actually supposed to handle a BYE, when the client is
> behind NAT.

Client behind NAT is supposed to keep its TCP connection to SIP Proxy
alive and use it for all requests of the call.  If the connection breaks
for some reason, the client sets up a new one for the remaining
requests.

-- Juha

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--------------------- Med Liberalistiske Hilsner ----------------------

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--

--------------------- 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 - www.liberalismen.dk

_______________________________________________
Kamailio (SER) - Users Mailing List
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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla