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

hello List, anyone could give some hints?? im still unable to rewrite the sdp body. hope to hear from you all. thanks -- Regards, MingHon On Tue, Jul 5, 2011 at 3:49 PM, MingHon &lt;gminghon(a)gmail.com&gt; wrote: > > 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