9 Nov
2020
9 Nov
'20
2:06 p.m.
Hello,
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...
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.
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).
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.
Cheers,
Daniel
On 09.11.20 10:13, Kjeld Flarup wrote:
Hello
I have attached a pcap received from the provider.
It is quite informative as it shows bits of how they forward the call.
We send to 194.247.61.26 which is a Kamailio proxy, that forwards the
call to a SBC 194.247.61.32
My guess is that the 194.247.61.26 kamailio gets confused, and does
not match the BYE with the ongoing TCP session.
Instead it sees it as a new session, and forwards it according to the
route information.
Can anybody help explaining what fields Kamailio uses to match an
ongoing TCP session.
Regards Kjeld
Den fre. 6. nov. 2020 kl. 10.50 skrev Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>>:
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:
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
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 <mailto: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:
--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Funding: https://www.paypal.me/dcmierla <https://www.paypal.me/dcmierla>
--
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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 <mailto:jh@tutpro.com>>:
Kjeld Flarup writes:
--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Funding: https://www.paypal.me/dcmierla <https://www.paypal.me/dcmierla>
--
--------------------- 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
<http://www.liberalismen.dk>
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<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> 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
----------------------
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
<http://www.liberalismen.dk>
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