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@gmail.com>:

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@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@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@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

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--

Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/

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-- 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
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