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(a)gmail.com>gt;:
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(a)tutpro.com>om>:
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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