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

Hi MingHon, what do you want to achieve? If it is only about rewritibng the SDP, then this will help you: fix_nated_sdp("10", "<your-ip-here>"); => 0x02 rewrite media IP address (c=) with the provided IP address => 0x08 rewrite IP from origin description (o=) with the provided IP address Kind regards, Carsten 2011/7/6 MingHon &lt;gminghon(a)gmail.com&gt;om>: > 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 -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220