23 Dec
2015
23 Dec
'15
3:28 p.m.
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not received in
Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included this in in
NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent by the
vendor. From what I see IP and port is taken from advertised in contact
(PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in order
to received new transactions or should I follow a different procedure???
Thank you
28 Dec
28 Dec
6:17 p.m.
AFAIK bye is usually sent to the address stored in record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)
2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migliaro(a)gmail.com>om>:
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not received in
Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included this in in
NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent by the
vendor. From what I see IP and port is taken from advertised in contact
(PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in
order to received new transactions or should I follow a different
procedure???
Thank you
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
12 Jan
12 Jan
7:13 p.m.
Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the public
port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568691(a)gmail.com>om>:
AFAIK bye is usually sent to the address stored in
record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)
2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migliaro(a)gmail.com>om>:
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not received in
Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included this in in
NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent by the
vendor. From what I see IP and port is taken from advertised in contact
(PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in
order to received new transactions or should I follow a different
procedure???
Thank you
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
13 Jan
13 Jan
12:09 a.m.
Hello,
can you get the SIP trace with all the packets of such dialog outside of
the NAT router? It will help to see the headers and based on that we may
be able to provide a solution.
Cheers,
Daniel
On 12/01/16 19:13, Nelson Migliaro wrote:
Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the
public port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568691(a)gmail.com
<mailto:568691@gmail.com>>:
AFAIK bye is usually sent to the address stored in record_route.
Try setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)
2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migliaro(a)gmail.com
<mailto:eng.migliaro@gmail.com>>:
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet
router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not
received in Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included
this in in NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact
$ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent
by the vendor. From what I see IP and port is taken from
advertised in contact (PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in order to received new transactions or should I
follow a different procedure???
Thank you
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493 <tel:%2B37367398493>
web: http://abs-telecom.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
12:59 p.m.
When you are running Kamailio behind a NAT you should use advertise
parameter of listen address. According to documentation " A typical use
case for advertise address is when running SIP server behind a
NAT/Firewall, when the local IP address (to be used for bind) is different
than the public IP address (to be used for advertising).". For example:
listen=udp:10.10.10.10:5060 advertise 11.11.11.11:5060
For more info refer to http://www.kamailio.org/wiki/cookbooks/3.3.x/core#listen
Regards
Gholamreza Sabery Tabrizy
On Wed, Jan 13, 2016 at 2:39 AM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
Hello,
can you get the SIP trace with all the packets of such dialog outside of
the NAT router? It will help to see the headers and based on that we may be
able to provide a solution.
Cheers,
Daniel
On 12/01/16 19:13, Nelson Migliaro wrote:
Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the
public port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568691(a)gmail.com>om>:
AFAIK bye is usually sent to the address stored
in record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)
2015-12-23 16:28 GMT+02:00 Nelson Migliaro < <eng.migliaro(a)gmail.com>
eng.migliaro(a)gmail.com>gt;:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not received in
Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included this in in
NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent by the
vendor. From what I see IP and port is taken from advertised in contact
(PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in
order to received new transactions or should I follow a different
procedure???
Thank you
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
8:31 p.m.
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT (5060->52548). I
found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060 and
router blocks de request. I assume vendor sends BYE to 5060 because it is a
new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 52548
and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060 advertise
PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact: <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1 SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@norvoz.es>;tag=as3b72a453
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
2016-01-13 12:59 GMT+01:00 Gholamreza Sabery <gr.sabery(a)gmail.com>om>:
When you are running Kamailio behind a NAT you should
use advertise
parameter of listen address. According to documentation " A typical use
case for advertise address is when running SIP server behind a
NAT/Firewall, when the local IP address (to be used for bind) is different
than the public IP address (to be used for advertising).". For example:
listen=udp:10.10.10.10:5060 advertise 11.11.11.11:5060
For more info refer to http://www.kamailio.org/wiki/cookbooks/3.3.x/core#listen
Regards
Gholamreza Sabery Tabrizy
On Wed, Jan 13, 2016 at 2:39 AM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
can you get the SIP trace with all the packets of such dialog outside of
the NAT router? It will help to see the headers and based on that we may be
able to provide a solution.
Cheers,
Daniel
On 12/01/16 19:13, Nelson Migliaro wrote:
Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the
public port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568691(a)gmail.com>om>:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
AFAIK bye is usually sent to the address stored
in record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)
2015-12-23 16:28 GMT+02:00 Nelson Migliaro < <eng.migliaro(a)gmail.com>
eng.migliaro(a)gmail.com>gt;:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I am running Kamailio behind NAT.
Kanailio has a private IP and I am relaying NAT to internet router.
I am using:
- #!define WITH_NAT
- listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
- Patched RTP proxy including the advertise option
And everything goes fine. I can make calls and have two way audio.
The problem begins when the callee ends the call. BYE is not received
in Kamailio (caller)
I included the public IP using "add_contact_alias" because
"set_contact_alias" was not adding the public IP. I included this in in
NATDETECT (pre loaded router)
if(is_first_hop()) {
xlog("L_NOTICE","Metodo: $rm \n");
xlog("L_NOTICE","is first hop\n");
#set_contact_alias();
if (!add_contact_alias("PUBLIC-IP", "$Rp",
"udp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
}
}
I think the problem is related to destination that BYE is sent by the
vendor. From what I see IP and port is taken from advertised in contact
(PUBLIC-IP and 5060).
The problem is that internet router changes the source port.
Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
--- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp",
"udp") in
order to received new transactions or should I follow a different
procedure???
Thank you
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
11:28 p.m.
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or
it doesn't provide the option to set static forwarding, then you are
pretty much left with sniffing the traffic to discover the external port
and advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT
(5060->52548). I found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port
5060 and router blocks de request. I assume vendor sends BYE to 5060
because it is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port
52548 and router forward the request to Kamailio. Since there is an
open connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact: <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@norvoz.es
<mailto:sip%3ANUM-SOURCE@norvoz.es>>;tag=as3b72a453
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
14 Jan
14 Jan
1:33 p.m.
New subject: Fwd: Kamailio and NAT
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a static
NAT.
By sniffing traffic I can see the port is using new but in case it change,
how can automate the process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or it
doesn't provide the option to set static forwarding, then you are pretty
much left with sniffing the traffic to discover the external port and
advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT (5060->52548). I
found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060 and
router blocks de request. I assume vendor sends BYE to 5060 because it is a
new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 52548
and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060 advertise
PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO
;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact: <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO
;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1 SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<sip%3ANUM-SOURCE(a)norvoz.es>VENDOR-IP
;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3:43 p.m.
New subject: Fwd: Kamailio and NAT
Is the kamailio behind nat communicating with another kamailio on a public
IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <eng.migliaro(a)gmail.com>
wrote:
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a
static NAT.
By sniffing traffic I can see the port is using new but in case it change,
how can automate the process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or it
doesn't provide the option to set static forwarding, then you are pretty
much left with sniffing the traffic to discover the external port and
advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT (5060->52548).
I found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060
and router blocks de request. I assume vendor sends BYE to 5060 because it
is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 52548
and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact: <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<sip%3ANUM-SOURCE(a)norvoz.es>VENDOR-IP
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3:47 p.m.
New subject: Fwd: Kamailio and NAT
There is not a public Kamailio, only one Kamailio behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) <->
Internet router (public IP + symmetric na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Is the kamailio behind nat communicating with another
kamailio on a public
IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <eng.migliaro(a)gmail.com>
wrote:
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a
static NAT.
By sniffing traffic I can see the port is using new but in case it
change, how can automate the process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or it
doesn't provide the option to set static forwarding, then you are pretty
much left with sniffing the traffic to discover the external port and
advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT (5060->52548).
I found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060
and router blocks de request. I assume vendor sends BYE to 5060 because it
is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they are
part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 52548
and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<sip%3ANUM-SOURCE(a)norvoz.es>VENDOR-IP
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:07 p.m.
New subject: Fwd: Kamailio and NAT
Do you control the Asterisk? If yes, depending on Asterisk capabilities of
building replies, you may be able to do some automation to detect the
external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro <eng.migliaro(a)gmail.com>
wrote:
There is not a public Kamailio, only one Kamailio
behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) <->
Internet router (public IP + symmetric na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
Is the kamailio behind nat communicating with
another kamailio on a
public IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <eng.migliaro(a)gmail.com>
wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a
static NAT.
By sniffing traffic I can see the port is using new but in case it
change, how can automate the process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or
it doesn't provide the option to set static forwarding, then you are pretty
much left with sniffing the traffic to discover the external port and
advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT
(5060->52548). I found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060
and router blocks de request. I assume vendor sends BYE to 5060 because it
is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 52548
and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<sip%3ANUM-SOURCE(a)norvoz.es>VENDOR-IP
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
5:19 p.m.
New subject: Fwd: Kamailio and NAT
Yes, I manage all devices, even the internet router but it does not allow
static pat.
2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Do you control the Asterisk? If yes, depending on
Asterisk capabilities of
building replies, you may be able to do some automation to detect the
external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro <eng.migliaro(a)gmail.com>
wrote:
There is not a public Kamailio, only one Kamailio
behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy )
<-> Internet router (public IP + symmetric na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Is the kamailio behind nat communicating with
another kamailio on a
public IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <eng.migliaro(a)gmail.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
wrote:
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a
static NAT.
By sniffing traffic I can see the port is using new but in case it
change, how can automate the process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List" <
sr-users(a)lists.sip-router.org>
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or
it doesn't provide the option to set static forwarding, then you are pretty
much left with sniffing the traffic to discover the external port and
advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT
(5060->52548). I found this port sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port 5060
and router blocks de request. I assume vendor sends BYE to 5060 because it
is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine because they
are part of the same transaction
When remote party disconnects the call, BYE goes to PUBLIC-IP port
52548 and router forward the request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to advertise the public port
internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<sip%3ANUM-SOURCE(a)norvoz.es>VENDOR-IP
>;tag=as3b72a453
Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route: <
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded NAT´d port.
I would like to find a way to avoid setting the port in "hard".
Thank you
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
6:32 p.m.
New subject: Fwd: Kamailio and NAT
Not really up to date with all Asterisk features -- do you know if you
can append a custom header to a SIP response that is going to be
generated by Asterisk? Eventually the reply for an OPTIONS request.
Cheers,
Daniel
On 14/01/16 17:19, Nelson Migliaro wrote:
Yes, I manage all devices, even the internet router
but it does not
allow static pat.
2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com
<mailto:miconda@gmail.com>>:
Do you control the Asterisk? If yes, depending on Asterisk
capabilities of building replies, you may be able to do some
automation to detect the external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro
<eng.migliaro(a)gmail.com <mailto:eng.migliaro@gmail.com>> wrote:
There is not a public Kamailio, only one Kamailio behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP
Proxy ) <-> Internet router (public IP + symmetric na) <->
Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>>:
Is the kamailio behind nat communicating with another
kamailio on a public IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro
<eng.migliaro(a)gmail.com <mailto:eng.migliaro@gmail.com>>
wrote:
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router
does not allow a static NAT.
By sniffing traffic I can see the port is using new
but in case it change, how can automate the process of
advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: *Daniel-Constantin Mierla* <miconda(a)gmail.com
<mailto:miconda@gmail.com>>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>>
Hello,
it looks like you have a symmetric nat router, so the
allocated port is randomly selected.
If you don't control the nat router to set a static
forwarding rule or it doesn't provide the option to
set static forwarding, then you are pretty much left
with sniffing the traffic to discover the external
port and advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
Hello,
I finally were able to run my Kamailio behind NAT but
in order to accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when
doing NAT (5060->52548). I found this port sniffing
traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it
does not work :(
When I dial a call, INVITE / ACK / Trying / OK goes
fine because they are part of the same transaction
When remote party disconnects the call, BYE goes to
PUBLIC-IP port 5060 and router blocks de request. I
assume vendor sends BYE to 5060 because it is a new
transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes
fine because they are part of the same transaction
When remote party disconnects the call, BYE goes to
PUBLIC-IP port 52548 and router forward the request
to Kamailio. Since there is an open connection.
I need to find the way to find the way to advertise
the public port internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I
included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060
-> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
BYE
sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <sip:NUM-SOURCE@
<mailto:sip%3ANUM-SOURCE@norvoz.es>VENDOR-IP>;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded
NAT´d port.
I would like to find a way to avoid setting the port
in "hard".
Thank you
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda
<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
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--
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http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/micond <http://www.linkedin.com/in/miconda>
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6:51 p.m.
New subject: Fwd: Kamailio and NAT
Yes it is possible, but is there an easy way to workaround the issue using
Kamailio.
Because I have the port because vendor is sending that info in Trying:
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=
z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548*
I need to find the way to discover the port used by firewall (maybe getting
that info from Trying) and then advertise that port.
2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Not really up to date with all Asterisk features -- do
you know if you can
append a custom header to a SIP response that is going to be generated by
Asterisk? Eventually the reply for an OPTIONS request.
Cheers,
Daniel
On 14/01/16 17:19, Nelson Migliaro wrote:
Yes, I manage all devices, even the internet router but it does not allow
static pat.
2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla < <miconda(a)gmail.com>
miconda(a)gmail.com>gt;:
Do you control the Asterisk? If yes, depending on
Asterisk capabilities
of building replies, you may be able to do some automation to detect the
external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro <
<eng.migliaro@gmail.com>eng.migliaro@gmail.com> wrote:
_______________________________________________
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--
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http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
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There is not a public Kamailio, only one Kamailio
behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy )
<-> Internet router (public IP + symmetric na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
--
Daniel-Constantin Mierla - <http://www.asipto.com>http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
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Is the kamailio behind nat communicating with
another kamailio on a
public IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <
<eng.migliaro@gmail.com>eng.migliaro@gmail.com> wrote:
> Thank you Daniel for your answer,
>
> As you mention, there is a symmetric nat and router does not allow a
> static NAT.
>
> By sniffing traffic I can see the port is using new but in case it
> change, how can automate the process of advertising the correct port?
>
> Cheers!
>
>
> ---------- Forwarded message ----------
> From: Daniel-Constantin Mierla < <miconda@gmail.com>miconda@gmail.com>
> Date: 2016-01-13 23:28 GMT+01:00
> Subject: Re: [SR-Users] Kamailio and NAT
> To: "Kamailio (SER) - Users Mailing List" <
> <sr-users@lists.sip-router.org>sr-users@lists.sip-router.org>
>
>
> Hello,
>
> it looks like you have a symmetric nat router, so the allocated port
> is randomly selected.
>
> If you don't control the nat router to set a static forwarding rule or
> it doesn't provide the option to set static forwarding, then you are pretty
> much left with sniffing the traffic to discover the external port and
> advertise it.
>
> Cheers,
> Daniel
>
>
>
>
> On 13/01/16 20:31, Nelson Migliaro wrote:
>
> Hello,
>
> I finally were able to run my Kamailio behind NAT but in order to
> accomplish that I included:
>
> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
>
> 52548 is the port my internet router change when doing NAT
> (5060->52548). I found this port sniffing traffic
>
> Conclusions at this point are:
>
>
>
---------------------------------------------1--------------------------------------------------------------------------------------------------
> If I use this line:
>
> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work :(
>
> When I dial a call, INVITE / ACK / Trying / OK goes fine because they
> are part of the same transaction
> When remote party disconnects the call, BYE goes to PUBLIC-IP port
> 5060 and router blocks de request. I assume vendor sends BYE to 5060
> because it is a new transaction
>
>
-----------------------------------------------2--------------------------------------------------------------------------------------------------
>
> If I use this line:
>
> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
>
> When I dial a call, INVITE / ACK / Trying / OK goes fine because they
> are part of the same transaction
> When remote party disconnects the call, BYE goes to PUBLIC-IP port
> 52548 and router forward the request to Kamailio. Since there is an open
> connection.
>
> I need to find the way to find the way to advertise the public port
> internet router is doing NAT (PAT).
>
>
>
---------------------------------------------------------------------------------------------------------------------------------------------------
> This trace is a call that worked fine because I included line:
>
> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
>
>
> This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
> advertise PUBLIC-IP:52548
> 2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
> INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
> Record-Route: <
>
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
> A-;nat=yes>
> Via: SIP/2.0/UDP
> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
> Via: SIP/2.0/UDP
>
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
> Max-Forwards: 69
> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
> To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
> Contact:
> <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
> CSeq: 102 INVITE
> User-Agent: Kamailio
> Date: Wed, 13 Jan 2016 19:10:15 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 255
>
>
> Trying.....
>
> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP
> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
> Via: SIP/2.0/UDP
>
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
> To: <sip:NUM-DESTINATION@VENDOR-IP>
> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
> CSeq: 102 INVITE
> Server: kamailio
> Content-Length: 0
>
>
>
>
> And finally a BYE
>
> 2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
> BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
> SIP/2.0
> Via: SIP/2.0/UDP
> VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
> From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
> To: "NUM-SOURCE" < <sip%3ANUM-SOURCE@norvoz.es>sip:NUM-SOURCE@
> VENDOR-IP>;tag=as3b72a453
> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
> CSeq: 28731 BYE
> Max-Forwards: 69
> Route: <
>
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
> yes>
> Reason: Q.850;cause=16
> Content-Length: 0
>
>
>
>
>
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
>
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> Finally, It is finally working because I hardcoded NAT´d port.
> I would like to find a way to avoid setting the port in "hard".
>
> Thank you
>
>
>
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> <sr-users@lists.sip-router.org>sr-users@lists.sip-router.org
> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
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15 Jan
15 Jan
7:58 a.m.
New subject: Fwd: Kamailio and NAT
Ahh, I thought Asterisk is in the public internet, but actually you
connect to a provider (vendor), which seems to run Kamailio as well.
Using information from 100 trying is too late, as the INVITE was already
sent... so one more question before trying to propose a solution. Do you
have to register to the provider?
Cheers,
Daniel
On 14/01/16 18:51, Nelson Migliaro wrote:
Yes it is possible, but is there an easy way to
workaround the issue
using Kamailio.
Because I have the port because vendor is sending that info in Trying:
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=
z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548*
I need to find the way to discover the port used by firewall (maybe
getting that info from Trying) and then advertise that port.
2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com
<mailto:miconda@gmail.com>>:
Not really up to date with all Asterisk features -- do you know if
you can append a custom header to a SIP response that is going to
be generated by Asterisk? Eventually the reply for an OPTIONS request.
Cheers,
Daniel
On 14/01/16 17:19, Nelson Migliaro wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
Yes, I manage all devices, even the internet
router but it does
not allow static pat.
2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>>:
Do you control the Asterisk? If yes, depending on Asterisk
capabilities of building replies, you may be able to do some
automation to detect the external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro
<eng.migliaro(a)gmail.com <mailto:eng.migliaro@gmail.com>> wrote:
There is not a public Kamailio, only one Kamailio behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP +
RTP Proxy ) <-> Internet router (public IP + symmetric
na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>>:
Is the kamailio behind nat communicating with another
kamailio on a public IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro
<eng.migliaro(a)gmail.com
<mailto:eng.migliaro@gmail.com>> wrote:
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and
router does not allow a static NAT.
By sniffing traffic I can see the port is using
new but in case it change, how can automate the
process of advertising the correct port?
Cheers!
---------- Forwarded message ----------
From: *Daniel-Constantin Mierla*
<miconda(a)gmail.com <mailto:miconda@gmail.com>>
Date: 2016-01-13 23:28 GMT+01:00
Subject: Re: [SR-Users] Kamailio and NAT
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>>
Hello,
it looks like you have a symmetric nat router, so
the allocated port is randomly selected.
If you don't control the nat router to set a
static forwarding rule or it doesn't provide the
option to set static forwarding, then you are
pretty much left with sniffing the traffic to
discover the external port and advertise it.
Cheers,
Daniel
On 13/01/16 20:31, Nelson Migliaro wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I finally were able to run my Kamailio behind
NAT but in order to accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when
doing NAT (5060->52548). I found this port
sniffing traffic
Conclusions at this point are:
---------------------------------------------1--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise
PUBLIC-IP:5060 it does not work :(
When I dial a call, INVITE / ACK / Trying / OK
goes fine because they are part of the same
transaction
When remote party disconnects the call, BYE goes
to PUBLIC-IP port 5060 and router blocks de
request. I assume vendor sends BYE to 5060
because it is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise
PUBLIC-IP:52548 it work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK
goes fine because they are part of the same
transaction
When remote party disconnects the call, BYE goes
to PUBLIC-IP port 52548 and router forward the
request to Kamailio. Since there is an open
connection.
I need to find the way to find the way to
advertise the public port internet router is
doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I
included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568
PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To: <sip:NUM-DESTINATION@VENDOR-IP>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
BYE
sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From:
<sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE"
<sip:NUM-SOURCE@VENDOR-IP>;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route:
<sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I
hardcoded NAT´d port.
I would like to find a way to avoid setting the
port in "hard".
Thank you
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda
<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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sr-users mailing list
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<mailto:sr-users@lists.sip-router.org>
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda
<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
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sr-users mailing list
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<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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sr-users mailing list
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<mailto:sr-users@lists.sip-router.org>
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda
<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/micond
<http://www.linkedin.com/in/miconda>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
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<mailto:sr-users@lists.sip-router.org>
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Alexandru Covalschi -
Daniel-Constantin Mierla -
Gholamreza Sabery -
Nelson Migliaro