Hi Daniel
The Record route in the INVITE from 194.247.61.26 sets this pair
Record-Route:
<sip:194.255.22.44:7071;transport=tcp;r2=on;lr=on;ftag=6acjlRdN~;did=836.f1b1>
Record-Route:
<sip:194.255.22.44:7071;r2=on;lr=on;ftag=6acjlRdN~;did=836.f1b1>
The Bye requests this route
Route:
<sip:194.255.22.44:7071;transport=tcp;r2=on;lr=on;ftag=6acjlRdN~;did=836.f1b1>
Route: <sip:194.255.22.44:7071;r2=on;lr=on;ftag=6acjlRdN~;did=836.f1b1>
But the real port on 194.255.22.44 is 36059
It is my invite to 194.247.61.26 that sets the 7071 port, which
automatically comes from the listen statement.
I suspect that it might work if the invite was using 36059, but how can
I know this port, if the NAT router decides to map it to another port.
What is the correct behaviour. Should my Kamailio somehow set the
correct port?
Should the providers Kamailio rewrite the route information?
Or something else?
-------------------- Med Liberalistiske Hilsner ----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
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On 1/11/21 10:18 AM, Daniel-Constantin Mierla wrote:
The From/To/Call-ID are not used to match the connection. The
connection is matched based on target IP and port. For BYE, these are
taken from Route header, if there is one for next hop, otherwise it is
the request URI. Check these two to see if something is not as
expected. Otherwise, you have to discuss with the provider and see the
reason it returns back the 477 response.
Cheers,
Daniel
On 08.01.21 20:36, Kjeld Flarup wrote:
Happy New Year everyone.
I haven't solved this problem yet. Although is it not critical, it is
a bit annoying.
I have tried to simplify things, and have a reference setup that works.
My linphone sends a TCP call and my Asterisk answers, plays a speak
and hangs up.
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.
Status-Line: SIP/2.0 477 Unfortunately error on sending to next
hop occurred (477/SL)
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.
I have attached a capture which the provider send me. This is the setup
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)
A note on the providers Kamailio. It listens on both port 5060 and
5070, and both UDP/TCP.
It is also used as access point for both my PBX and my Asterisk, thus
the trace may be a little confusing to read.
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.
The flow is also clean, there is no change of ports etc.
I have this reference setup which works
linphone -> provider Kamailio -> SBC -> provider Kamailio -> my
Asterisk
The only differences towards the reference I can see these. I do not
have a capture from the provider on this.
* There is an extra Via step.
* Contact points to the Linphone IP, not the Kamailio IP
Any hint will be appreciated.
-------------------- 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
On 11/9/20 12:06 PM, Daniel-Constantin Mierla wrote:
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
<mico...@gmail.com <mailto:mico...@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:
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
<mico...@gmail.com <mailto:mico...@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
<j...@tutpro.com <mailto:j...@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
----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min
tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
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<http://www.liberalismen.dk>
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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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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
--
--------------------- 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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www.twitter.com/miconda --www.linkedin.com/in/miconda
Funding:https://www.paypal.me/dcmierla
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www.twitter.com/miconda --www.linkedin.com/in/miconda
Funding:https://www.paypal.me/dcmierla
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