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 gminghon@gmail.com:
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 carsten@ng-voice.com 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 gminghon@gmail.com:
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 gminghon@gmail.com 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@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@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@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