Hi list,

We have two eth,  one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private).

With mhomed configured shouldn't Record Route be updated as VIA header?

".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..."

This is part of ngrep

U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060

INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0.

Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.

From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1.

To: <sip:1111111111@172.16.213.38:5060>.

Call-ID: 1-13024@172.16.213.21.

CSeq: 1 INVITE.

Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>.

....

U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060

SIP/2.0 100 trying -- your call is important to us.

Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.

....

U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060

INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0.

-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----

Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f

Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6

This is my cfg

mhomed=1

request_route {

        # per request initial checks

        route(REQINIT);

        # NAT detection

        #route(NATDETECT);

        # CANCEL processing

        if (is_method("CANCEL")) {

                if (t_check_trans()) {

                        route(RELAY);

                }

                exit;

        }

        # handle requests within SIP dialogs

        route(WITHINDLG);

        ### only initial requests (no To tag)

        # handle retransmissions

        if(t_precheck_trans()) {

                t_check_trans();

                exit;

        }

        t_check_trans();

        # authentication

        route(AUTH);

        # record routing for dialog forming requests (in case they are routed)

        # - remove preloaded route headers

        remove_hf("Route");

        if (is_method("INVITE|SUBSCRIBE")) {

                record_route();

        }

....

Thanks in advance.

Diego

2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares@gmail.com>:

Hi list,

We have two eth,  one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private).

With mhomed configured shouldn't Record Route be updated as VIA header 

".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..."

This is part of ngrep

U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060

INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0.

Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.

From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1.

To: <sip:1111111111@172.16.213.38:5060>.

Call-ID: 1-13024@172.16.213.21.

CSeq: 1 INVITE.

Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>.

....

U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060

SIP/2.0 100 trying -- your call is important to us.

Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.

....

U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060

INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0.

-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----

Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f

Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6

This is my cfg

mhomed=1

request_route {

        # per request initial checks

        route(REQINIT);

        # NAT detection

        #route(NATDETECT);

        # CANCEL processing

        if (is_method("CANCEL")) {

                if (t_check_trans()) {

                        route(RELAY);

                }

                exit;

        }

        # handle requests within SIP dialogs

        route(WITHINDLG);

        ### only initial requests (no To tag)

        # handle retransmissions

        if(t_precheck_trans()) {

                t_check_trans();

                exit;

        }

        t_check_trans();

        # authentication

        route(AUTH);

        # record routing for dialog forming requests (in case they are routed)

        # - remove preloaded route headers

        remove_hf("Route");

        if (is_method("INVITE|SUBSCRIBE")) {

                record_route();

        }

Thanks in advance.

Diego.