22 Dec
2016
22 Dec
'16
6:08 p.m.
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(a)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.