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

Wed Jul 6 09:29:16 CEST 2011

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 at 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 at 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 at 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 at 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 at aextddns.dyndns.info>;tag=as032358a3
> >> To: <sip:102 at 192.168.1.3:5060>
> >> Contact: <sip:102 at 192.168.1.23:5080>
> >> Call-ID: 416f6e09674ae9671bb7144a1cb11137 at 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 at aextddns.dyndns.info>;tag=1796959074
> >> To: <sip:102 at aextddns.dyndns.info>;tag=as2e4c0125
> >> Call-ID: 1985782590 at 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 at 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 at 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 at aextddns.dyndns.info>;tag=640933430
> >> To: <sip:102 at aextddns.dyndns.info>
> >> Call-ID: 1909950509 at 192.168.2.200
> >> CSeq: 21 INVITE
> >> Contact: <sip:101 at 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 at ng-voice.com
>
> Schomburgstr. 80
> 22767 Hamburg
> Germany
>
> Mobile +49 179 2021244
> Office +49 40 34927219
> Fax +49 40 34927220
>

-- 
Regards,

MingHon
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