increase the log level and verify if force_rtp_proxy is called during
response processing or not. If yes -> there is a problem in
force_rtp_proxy. If no -> you have to debug your configuration by adding
more xlog() messages, e.g. to log the value of the relevant flags and
the status.
regards
klaus
Am 30.08.2010 13:10, schrieb peter_green lion:
hi all,
i am a new user kamailio.
i have configure kamailio with RTP proxy, but i have a problem in using :
|force_rtp_proxy("c","192.168.1.10")|
because i want to change to force to this ip.
my configure is :
when server receive "200 OK" it change value in "c= IN IP4 <ip rtp
server>" to "c = IN IP4 192.168.1.10"
i configure as :
kamailio.cfg :
#!ifdef WITH_NAT
if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy("c","192.168.1.10");
}
if (isbflagset("6")) {
fix_nated_contact();
}
#!endif
}
when i make call from sip a to sip b, sip b answer .trace as :
U 115.78.129.190:54337 -> <server ip>:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP <server ip>;branch=z9hG4bK00c.daa8fe8.0.
Via: SIP/2.0/UDP
192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930.
Record-Route: <sip:<server ip>;lr;nat=yes>.
Contact: <sip:1...@192.168.1.10:8576;rinstance=8392ffb3fe461110>.
To: <sip:102@<server ip>:5060>;tag=d179a842.
From: <sip:101@<server ip>:5060>;tag=1c61b708.
Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
REGISTER, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: PortGo v6.0, Build 07282010.
Content-Length: 237.
.
v=0.
o=- 30452887 30452887 IN IP4 169.254.202.160.
s=http://www.portsip.com.
c=IN IP4 169.254.202.160.
t=0 0.
m=audio 21480 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.< br>
U <server ip>:5060 -> 115.78.129.190:63930
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930.
Record-Route: <sip:server ip;lr;nat=yes>.
Contact: <sip:1...@115.78.129.190:54337;rinstance=8392ffb3fe4611 10>.
To: <sip:1...@server ip>:5060>;tag=d179a842.
From: <sip:101@<server ip>:5060>;tag=1c61b708.
Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
REGISTER, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: PortGo v6.0, Build 07282010.
Content-Length: 237.
.
v=0.
o=- 30452887 30452887 IN IP4 169.254.202.160.
s=http://www.portsip.com.
c=IN IP4 169.254.202.160.
t=0 0.
m=audio 21480 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
thanks for help me.
regards.
beter_green
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
