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

Hi, Note: The methods of rtpproxy-module will only replace the IP, if the Kamailio can access the RTPProxy. How is your RTPProxy connected to your Kamailio? Socket or TCP? Do you have the Kamailio FIFO enabeld? If you have the fifo enabled, you should check the following: kamctl fifo nh_show_rtpp You should see, that the Kamailio is connected to the RTPProxy. If no, then that is your problem. If the RTPProxy is connected and is listening on the TCP socket, then you can do an ngrep to see the communication between Kamailio and RTPProxy, which might help you further with your investigation. Carsten 2011/7/6 MingHon &lt;gminghon(a)gmail.com&gt;om>: > Hi Carsten, > no is not about just rewriting the SDP. > i need my UACs media to relay on my rtpproxy > currently my UACs are sending the media to a private ip. > my rtpproxy is in behind nat and UACs behind another nat. > > On Wed, Jul 6, 2011 at 3:15 PM, Carsten Bock &lt;carsten(a)ng-voice.com&gt; > wrote: >> >> 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 > > > > -- > 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 ~~~~~ Checkout SIP-Provider CE: http://www.sipwise.com/products/spce/overview/