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 <gming...@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 <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
--
Carsten Bock
http://www.ng-voice.com
mailto:cars...@ng-voice.com
Schomburgstr. 80
22767 Hamburg
Germany
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
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