[SR-Users] help to use force_rtp_proxy([flags [, ip_address]]).

[Klaus Darilion]

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:102 at 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:102 at 115.78.129.190:54337;rinstance=8392ffb3fe4611 10>.
> To: <sip:102 at 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
>
>
>
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