On June 14, 2017 at 5:19 PM John Petrini <jpetr...@coredial.com> wrote:You might try using the fix_nated_sdp function from the nathelper module to rewrite the c line in the SDP body.___
John Petrini
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On Wed, Jun 14, 2017 at 5:00 PM, Pascal Poudrier <pascal.poudr...@quebecinternet.net> wrote:Hi, first of all, here's the architecture of what I'm trying to do :
Public client --> Kamailio and there is a private freeswitch behind kamailio. I use rtpproxy to connect the client and the freeswitch that is used at media server.
If a call is not answered from userA to userB and userB doesn't have a voicemail configured, I route the Invite to FreeSwitch and I change the destination user to 666666 that take care of playing a nice message that the user doesn't have a voicemail.
My problem is that the SDP Connection doesn't get updated to the right IP. It get changed to the external IP instead of the internal.
Here's the original INVITE Packet :
13:21:22.824290 IP 8.8.8.16.30893 > 8.8.8.18.5060: SIP: INVITE sip:1...@mydomain.com SIP/2.0
Eh....@.?...........x......mINVITE sip:1...@mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK-3e3b9e22
From: 102;tag=cacbdfac40ddd17ao0
To: <sip:1...@mydomain.com>
Remote-Party-ID: 102;screen=yes;party=calling
Call-ID: 3ae5b914-591b8732@192.168.0.236
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 102
Expires: 240
User-Agent: Cisco/SPA122-1.3.5r(003)
Content-Length: 263
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 16833207 16833207 IN IP4 192.168.0.236
s=-
c=IN IP4 192.168.0.236
t=0 0
m=audio 16396 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecvNote that 192.168.0.x in the internal network of the clients and 8.8.8.16 is the public IP of the client and 8.8.8.18 is the kamailio ip.
Here's the INVITE packet sent from kamailio to Freeswitch
13:21:32.947196 IP 172.16.0.18.5060 > 172.16.0.19.5080: SIP: INVITE sip:666666666666666@172.16.0.
19:5080 SIP/2.0
E... ...@. ...e...e.......P1INVITE sip:666666666666666@172.16.0.19:5080 SIP/2.0
Record-Route: <sip:8.8.8.18;lr=on;ftag=cacbdfac40ddd17ao0;did=f3f. 205>
Test-Pascal: SDP avec freeswitch
Via: SIP/2.0/UDP 172.16.0.18;branch=z9hG4bK31ea. e52b370f8248032680211fc7c25390 2a.1
Via: SIP/2.0/UDP 192.168.0.236:5060;rport=30893;received=8.8.8.16; branch=z9hG4bK-3e3b9e22
From: 102;tag=cacbdfac40ddd17ao0
To: <sip:1...@mydomain.com>
Remote-Party-ID: 102;screen=yes;party=calling
Call-ID: 3ae5b914-591b8732@192.168.0.236
CSeq: 101 INVITE
Max-Forwards: 69
Contact: 102
Expires: 240
User-Agent: Cisco/SPA122-1.3.5r(003)
Content-Length: 279
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 16833207 16833207 IN IP4 8.8.8.18
s=-
c=IN IP4 8.8.8.18
t=0 0
m=audio 46134 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=nortpproxy:yesNote that 172.16.0.18 is the internal IP of kamailio and 172.16.0.19 is the internal IP of FreeSwitch
Here's the revealing par of kamailio configuration that us use when falling back to freeswitch :
failure_route[TOVOICEMAIL] {
$ru = "sip:666666666666666@" + $sel(cfg_get.freeswitch.
bindip)
+ ":" + $sel(cfg_get.freeswitch.bindport);
$du = "sip:666666666666666@"+ $sel(cfg_get.freeswitch.bindip)
+ ":" + $sel(cfg_get.freeswitch.bindport);
force_send_socket(udp:172.16.0.18:5060); if (is_request()) {
xlog("L_INFO","$ft -- NATMANAGE -- Request");
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLT_NATS);
}
}
}
if(nat_uac_test("8")) {
xlog("L_INFO","---------------------------------- -----------------");
xlog("L_INFO","---------------------------------- -----------------");
xlog("L_INFO","SDP Body avant : $fu --> $du | $ru");
xlog("L_INFO","---------------------------------- -----------------");
xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : NAT SDP");
rtpproxy_manage("1cowei",'192.168.101.18');
insert_hf("Test-Pascal: SDP avec freeswitch\r\n");
xlog("L_INFO","---------------------------------- -----------------");
}
else {
xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : PAS DE NAT SDP");
rtpproxy_manage("cowrei");
}
if (is_reply()) {
xlog("L_INFO","$ft -- NATMANAGE -- Reply");
if(isbflagset(FLT_NATS)) {
if(is_first_hop())
set_contact_alias();
}
if ( $si == $sel(cfg_get.freeswitch.bindip) && is_method("SUBSCRIBE") ) {
remove_hf("Contact");
insert_hf("Contact: \$fU\\r\n","Expires");
}
else {
fix_nated_contact();
}
}
if ( is_method("INVITE") )
setflag(FLT_DIALOG);
if (!t_relay()) {
xlog("L_INFO","$ft -- ERROR Relay : $rc");
sl_reply_error();
}
------------------------------------------------ This configuration is working if the call is not going through failure_route
I use Kamailio version 5.0.1 on Debian.
Thank you
Pascal
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