Hello
I have a Kamailio running behind NAT, which sends calls to a VOIP service provider.
I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router.
Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog.
To which port is the server supposed to send the BYE, and what field tells the server this.
Hello,
your message is confusing somehow, what do you mean by "never sees the BYE, because BYE is a new dialog"?
Maybe you can grab the pcap with sip packets for such call on kamailio server and sent it over to see if the headers are as expected.
Cheers, Daniel
On 04.11.20 12:04, Kjeld Flarup wrote:
Hello
I have a Kamailio running behind NAT, which sends calls to a VOIP service provider.
I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router.
Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog.
To which port is the server supposed to send the BYE, and what field tells the server this.
-- -------------------- 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 sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I think you are searching for this parameter:
http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_reuse_port
It basically tells the kernel not to choose a new port at random for a new connection, but it will try to reuse the existing port (in your case 5071).
Thanks, Carsten
-- Carsten Bock I CTO & Founder
ng-voice GmbH
Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock
Am Mi., 4. Nov. 2020 um 12:07 Uhr schrieb Kjeld Flarup < kjeld.flarup@liberalismen.dk>:
Hello
I have a Kamailio running behind NAT, which sends calls to a VOIP service provider.
I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router.
Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog.
To which port is the server supposed to send the BYE, and what field tells the server this.
-- -------------------- 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 sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello again
I have attached a Capture.
My public IP is 194.255.22.44, and I have a port forward of 5071 in my router
The Raspberry Pi running the Kamailio is on 192.168.2.9 with an alias 192.168.2.40
My listening is setup like this
listen=127.0.0.1:5071 listen=eth0:5071 advertise 194.255.22.44:5071 alias=194.255.22.44:5071
Kamailio is forwarding an invite from a door phone thus I also rewrite FROM:
uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@194.255.22.44:5071"); And the failure route to $fu = "sip:"+$dbr(ra=>[0,0])+"@194.255.22.44:5071";
In the route doing the INVITE I also set this:
set_advertised_address("194.255.22.44:5071");
Regarding tcp_reuse_port, if I set this I get this error, which I understand is due to the way TCP works.
Nov 4 18:38:41 scantronpbx /usr/sbin/kamailio[15864]: {1 ACK 16191 ACK C2Az-Xm2b0CEPyV5eQuz7yEf9IJo4PyJ} WARNING: <core> [core/tcp_main.c:1061]: tcp_do_connect(): binding to source address 192.168.2.40:5071 failed: Address already in use [98]
I do have tcp_reuse_port=yes with this capture. But much to my surprise, the 200 OK from is send to 37148, but the ACK to it comes from 37150
-------------------- 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 11/4/20 1:59 PM, Carsten Bock wrote:
Hi,
I think you are searching for this parameter:
http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_reuse_port
It basically tells the kernel not to choose a new port at random for a new connection, but it will try to reuse the existing port (in your case 5071).
Thanks, Carsten
-- Carsten Bock I CTO & Founder
ng-voice GmbH
Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com http://www.ng-voice.com/
Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock
Am Mi., 4. Nov. 2020 um 12:07 Uhr schrieb Kjeld Flarup <kjeld.flarup@liberalismen.dk mailto:kjeld.flarup@liberalismen.dk>:
Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello
I got an answer from the VOIP provider.
They claim that we have an error in our port forwarding. Because we advertise 5071, but traffic comes from another port.
As I understand, it is however impossible for TCP to send all traffic from just one port.
How is TCP SIP actually supposed to handle a BYE, when the client is behind NAT.
-------------------- 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 11/4/20 10:00 PM, Kjeld Flarup wrote:
Hello again
I have attached a Capture.
My public IP is 194.255.22.44, and I have a port forward of 5071 in my router
The Raspberry Pi running the Kamailio is on 192.168.2.9 with an alias 192.168.2.40
My listening is setup like this
listen=127.0.0.1:5071 listen=eth0:5071 advertise 194.255.22.44:5071 alias=194.255.22.44:5071
Kamailio is forwarding an invite from a door phone thus I also rewrite FROM:
uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@194.255.22.44:5071"); And the failure route to $fu = "sip:"+$dbr(ra=>[0,0])+"@194.255.22.44:5071";
In the route doing the INVITE I also set this:
set_advertised_address("194.255.22.44:5071");
Regarding tcp_reuse_port, if I set this I get this error, which I understand is due to the way TCP works.
Nov 4 18:38:41 scantronpbx /usr/sbin/kamailio[15864]: {1 ACK 16191 ACK C2Az-Xm2b0CEPyV5eQuz7yEf9IJo4PyJ} WARNING: <core> [core/tcp_main.c:1061]: tcp_do_connect(): binding to source address 192.168.2.40:5071 failed: Address already in use [98]
I do have tcp_reuse_port=yes with this capture. But much to my surprise, the 200 OK from is send to 37148, but the ACK to it comes from 37150
-------------------- 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 11/4/20 1:59 PM, Carsten Bock wrote:
Hi,
I think you are searching for this parameter:
http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_reuse_port
It basically tells the kernel not to choose a new port at random for a new connection, but it will try to reuse the existing port (in your case 5071).
Thanks, Carsten
-- Carsten Bock I CTO & Founder
ng-voice GmbH
Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com http://www.ng-voice.com/
Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock
Am Mi., 4. Nov. 2020 um 12:07 Uhr schrieb Kjeld Flarup <kjeld.flarup@liberalismen.dk mailto:kjeld.flarup@liberalismen.dk>:
Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>--
--------------------- 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
--------------------- 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 Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
Kjeld Flarup writes:
My question is still, which port is the BYE from the server supposed to be sent to?
If client is behind NAT, the server must use the connection that the client opened to it to send the request to client. There is no way the server opens the connection and uses it.
-- Juha
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 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: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 <mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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@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 < 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
--------------------- 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 Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 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
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
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@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: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 <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: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 <mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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@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: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 <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: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 <mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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 -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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--
--------------------- 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>
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla --www.asipto.com www.twitter.com/miconda --www.linkedin.com/in/miconda Funding:https://www.paypal.me/dcmierla
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 AsteriskThe 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@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: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 <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: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 <mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
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 AsteriskThe 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@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: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 <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: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 <mailto: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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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 -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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--
--------------------- 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>
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla --www.asipto.com www.twitter.com/miconda --www.linkedin.com/in/miconda Funding:https://www.paypal.me/dcmierla
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla --www.asipto.com www.twitter.com/miconda --www.linkedin.com/in/miconda Funding:https://www.paypal.me/dcmierla
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@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 ---------------------- 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 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@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 < 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
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
--------------------- 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 Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 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
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 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
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Yuriy
Thanks for Your suggestions. I looked at them, and it seems to me that they all are supposed to be on the receiving side. My side is the client side behind NAT, and only does INVITE, I never receives INVITES.
The alias concept looks interesting but I doubt that I can convince the provider to use is, as the documentation states it to be dangerous.
When looking up the force_tcp_alias I noticed that fix_natted_contact was recomended for NAT traversal. I do not know if the provider uses, this function. Could that be the cause?
-------------------- 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 1/12/21 8:59 AM, Yuriy Gorlichenko wrote:
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@liberalismen.dk mailto:kjeld.flarup@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 ---------------------- 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> 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 <http://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 <http://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 <http://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 <http://194.247.61.32:5040>, which cannot deliver it further based on Route header. The system at 194.247.61.26:5070 <http://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 <http://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@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: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 <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:> 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 mailto: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 > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > mailto:sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > > --------------------- 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 > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- 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 -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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 -- --------------------- 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> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- 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_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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