Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.

Hi List, im facing an issue that my kamailio proxy did not replace the ip address in the invite and 200OK sdp body. my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user my kamailio is listening on 192.168.1.3, also define: advertised_address="175.136.223.112"; & advertised_port=5060; and my asterisk is on 192.168.1.23. sip signalling and rtp port forwarded to kamailio. uacs from another nat register successfully. if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112"); i will get double ip addr in c and o but kamailio ignore my ip addr. example i will get c=IN IP4 192.168.1.3192.168.1.3 here is part of my simple script. hope you can help. thank you very much. ---------------cfg------------------- route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy("fcow","175.136.223.112"); #force_rtp_proxy("fcow","175.136.223.112"); xlog("L_INFO","offer"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -------------------------------------- and here is the wireshark for uac INVITE and OK. -----------INVITE----------------- ve0 EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0 Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080 Max-Forwards: 69 From: "101" <sip:102@aextddns.dyndns.info>;tag=as032358a3 To: <sip:102@192.168.1.3:5060> Contact: <sip:102@192.168.1.23:5080> Call-ID: 416f6e09674ae9671bb7144a1cb11137(a)aextddns.dyndns.info CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 05 Jul 2011 07:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 327 v=0 o=root 1639709788 1639709788 IN IP4 192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.3 t=0 0 m=audio 10072 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -----------200OK--------------- e90 ElE;pX4tSIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416 Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes> From: "101" <sip:101@aextddns.dyndns.info>;tag=1796959074 To: <sip:102@aextddns.dyndns.info>;tag=as2e4c0125 Call-ID: 1985782590(a)192.168.2.200 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:102@192.168.1.23:5080> Content-Type: application/sdp Content-Length: 286 v=0 o=root 403900934 403900934 IN IP4 192.168.1.23 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.23 t=0 0 m=audio 14420 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------ My kamailio log. -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3 INFO: <script>: offer ------------------------------------- double force_rtp_proxy --------kamailio -> asterisk [INVITE]--------- Pyi-}E7V@:#pINVITE sip:102@aextddns.dyndns.info SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0 Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648 From: "101" <sip:101@aextddns.dyndns.info>;tag=640933430 To: <sip:102@aextddns.dyndns.info> Call-ID: 1909950509(a)192.168.2.200 CSeq: 21 INVITE Contact: <sip:101@175.138.21.31:2788> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 69 User-Agent: T20 9.41.0.80 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 334 v=0 o=20073 20073 IN IP4 192.168.1.3192.168.1.3 s=SDP data c=IN IP4 192.168.1.3192.168.1.3 t=0 0 m=audio 1006410064 RTP/AVP 0 8 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 INFO: <script>: offer -----------LOG------------------ -- Regards, MingHon