12 Jan
2021
12 Jan
'21
2:59 a.m.
It doesn't matter whet port used by provider when it sent initial INVITE to
you.
Record-route and Route headers are Proxy headers. They are announce
addresses of the proxy for the listening. That means when UA sends the
request it has to use port mentioned in the first of the Route headers or
in the Request URI header.
That means that your kamailio has to create new connection to this host
port pair or reuse it if it already exists to reach provider's server. So
there is nothing bad if you will create new connection for BYE to port 7071.
However, If provider initiated INVITE to you and sent it from the different
port ( which is true because for that transaction provider has to behave
atleast as TCP client ) and connection still alive ( socket still exists )
- you can try to use $du variable and put here existing address used for
the connection to provider.
But remember it is a hack.
This is also can be achieved via as mentioned above global param
tcp_accept_aliases =yes
And functions wich has to be called on initial invite:
tcp_keepalive_enable
force_tcp_alias
On Tue, 12 Jan 2021, 07:15 Kjeld Flarup, <kjeld.flarup(a)liberalismen.dk>
wrote:
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 ----------------------
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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 <
miconda(a)gmail.com>gt;:
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;:
--
--------------------- 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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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>:
--
--------------------- Med Liberalistiske Hilsner ----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
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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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Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
_______________________________________________
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Funding: https://www.paypal.me/dcmierla