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 <gming...@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:1...@aextddns.dyndns.info>;tag=as032358a3
> To: <sip:102@192.168.1.3:5060>
> Contact: <sip:102@192.168.1.23:5080>
> Call-ID: 416f6e09674ae9671bb7144a1cb11...@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:1...@aextddns.dyndns.info>;tag=1796959074
> To: <sip:1...@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:1...@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:1...@aextddns.dyndns.info>;tag=640933430
> To: <sip:1...@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
>
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