Hello,
We are trying to get Kamailio working as an SBC using rtpproxy();
The config seems to work except with some carriers, kamailio does not
recognize the ACK because it does not match the contact header of the OK
I've been trying to figure out how to get kamailio to send the OK back
using the public IP address but I cannot figure out how.
I've tried using various deployments of the nathelper module such as
fix_nated_contact()
set_contact_alias()
No matter what though, Kamailio will not change the contact header it
receives from FS before sending it back to the carriers.
This has become frustrating as I read tutorials that showed this is
possible, yet I am unable.
Thank you.
/V
U 2015/06/25 15:40:09.095946 99.12.125.162:5060 -> 2.43.7.157:5060
INVITE sip:500@3.local SIP/2.0.
CSeq: 1 INVITE.
Via: SIP/2.0/UDP
192.168.16.120:5060;branch=z9hG4bKb0421938-f90f-1910-9836-8851fb4369e9;rport.
User-Agent: Ekiga/4.0.2.
From: "TEST" <sip:1000@3.local>;tag=643e1938-f90f-1910-9834-8851fb4369e9.
Call-ID: 643e1938-f90f-1910-9835-8851fb4369e9@EV.
Supported: 100rel,replaces.
To: <sip:500@3.local>.
Contact: "TEST" <sip:ekiga@192.168.16.120>.
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK.
Content-Length: 256.
Content-Type: application/sdp.
Max-Forwards: 70.
.
v=0.
o=- 1435261202 1 IN IP4 192.168.16.120.
s=Ekiga/4.0.2.
c=IN IP4 192.168.16.120.
t=0 0.
m=audio 5082 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16,32,36.
a=maxptime:240.
U 2015/06/25 15:40:09.098884 2.43.7.157:5060 -> 99.12.125.162:5060
SIP/2.0 100 trying -- your call is important to us.
CSeq: 1 INVITE.
Via: SIP/2.0/UDP
192.168.16.120:5060;branch=z9hG4bKb0421938-f90f-1910-9836-8851fb4369e9;rport=5060;received=99.12.125.162.
From: "TEST" <sip:1000@3.local>;tag=643e1938-f90f-1910-9834-8851fb4369e9.
Call-ID: 643e1938-f90f-1910-9835-8851fb4369e9@EV.
To: <sip:500@3.local>.
Server: kamailio (4.4.0-dev1 (x86_64/linux)).
Content-Length: 0.
.
U 2015/06/25 15:40:09.115025 2.43.7.157:5060 -> 99.12.125.162:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.16.120:5060;received=99.12.125.162;branch=z9hG4bKb0421938-f90f-1910-9836-8851fb4369e9;rport=5060.
Record-Route:
<sip:2.43.7.157;lr;ftag=643e1938-f90f-1910-9834-8851fb4369e9;nat=yes>.
From: "TEST" <sip:1000@3.local>;tag=643e1938-f90f-1910-9834-8851fb4369e9.
To: <sip:500@3.local>;tag=5cF0vyUt2KQSH.
Call-ID: 643e1938-f90f-1910-9835-8851fb4369e9@EV.
CSeq: 1 INVITE.
Contact: <sip:500@192.168.0.11:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150611T050046Z~67ed8f4260~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 240.
Remote-Party-ID: "500" <sip:500@3.local>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1435241875 1435241876 IN IP4 2.43.7.157.
s=FreeSWITCH.
c=IN IP4 2.43.7.157.
t=0 0.
m=audio 35522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=nortpproxy:yes.
U 2015/06/25 15:40:09.125027 99.12.125.162:5060 -> 2.43.7.157:5060
ACK sip:500@192.168.0.11:5060;transport=udp SIP/2.0.
Route: <sip:2.43.7.157;lr;nat=yes;ftag=643e1938-f90f-1910-9834-8851fb4369e9>.
CSeq: 1 ACK.
Via: SIP/2.0/UDP
192.168.16.120:5060;branch=z9hG4bK304f1938-f90f-1910-9836-8851fb4369e9;rport.
From: "TEST" <sip:1000@3.local>;tag=643e1938-f90f-1910-9834-8851fb4369e9.
Call-ID: 643e1938-f90f-1910-9835-8851fb4369e9@EV.
To: <sip:500@3.local>;tag=5cF0vyUt2KQSH.
Contact: "TEST" <sip:ekiga@192.168.16.120>.
Content-Length: 0.
Max-Forwards: 70.
.
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