Hello All, So I had some success using topology_hiding and rtpproxy but found few problems.
After implementing topology_hiding(), SIP INVITE was much better but still showing following: INVITE sip:aaabbbc...@outboundprovider.com:5060 SIP/2.0 Call-ID: 4ed41738da10faa5@172.16.16.250 *<<<-- showing originators Device LAN IP —>>>* Content-Length: 329 CSeq: 8002 INVITE From: <sip:zzzzzzz...@outboundprovider.com>;tag=SP39b79130abfb7487f Max-Forwards: 69 To: <sip: aaabbbc...@3.xxx.xxx.49> Via: SIP/2.0/UDP 3.xxx.xxx.49:5060;branch=z9hG4bK1dcb.5bb78035.0 User-Agent: OBIHAI/OBi302-3.2.2.6259 *<<<-- showing originators User-Agent —>>>* Contact: <sip:3.xxx.xxx.49;did=6a7.5e849703> Expires: 60 Supported: replaces Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE Content-Type: application/sdp === 1) How can I remove IP from Call-ID and rewrite Originators User-Agent to local OpenSIPS User-Agent? === Now issue with rtpproxy - I'm running this OpenSIPS on AWS cloud... AWS cloud does natting by default, so my Public IP is 3.xxx.xxx.49 and actual VM IP is *172.31.29.47. * After implement rtpproxy (https://www.rtpproxy.org/), it is running on local IP: └─183589 /usr/local/bin/rtpproxy -s udp:172.31.29.47 22222 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 172.31.29.47 -m 1000 -M 2000 -d INFO LOG_LOCAL5 As it shows from SIP INVITE and due to that no audio or RTP because IP is not reachable... v=0 o=- 16210664 1 IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>* s=- c=IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>* t=0 0 m=audio 1958 RTP/AVP 0 8 18 104 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:104 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 a=xg726bitorder:big-endian a=nortpproxy:yes === 2. How can I configure rtpproxy with Public IP? Or do I start rtpproxy with Public IP 3.xxx.xxx.49 and reconfigure OpenSIPS with Public IP? modparam("rtpproxy", "rtpproxy_sock", "udp:172.31.29.47:22222") Thanking in advance... Cheers, Nitesh On Wed, Oct 19, 2022 at 10:17 AM Nitesh Divecha <aviator.nites...@gmail.com> wrote: > Hello, > > Thank y'all for the input... I will try to read the documentation and work > on implementing these modules. > > By any chance do either of you have any working examples which I can refer > to? I'm a work in progress and every time I change something I break > OpenSIPS and it takes me hours to troubleshoot! :-) > > Thanking in advance... > > Cheers, > Nitesh > > > > On Wed, Oct 19, 2022 at 2:20 AM Bogdan-Andrei Iancu <bog...@opensips.org> > wrote: > >> Hi there, >> >> Actually you do not need the B2B, you can achieve the same kind of >> privacy (at SIP level) with dialog module and topology_hiding module >> together. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 5-16 Dec 2022, online >> https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 10/19/22 1:23 AM, Abdul Basit wrote: >> >> Nitesh, >> >> You need a B2BUA function >> <https://www.tutorialspoint.com/session_initiation_protocol/session_initiation_protocol_b2bua.htm> >> with >> the help of a topo-hiding module with opensips as Bela shared in his email. >> Also, install the RTP proxy on the same opensips box (not necessary if >> you need separate signaling and media boxes). >> >> Far end party will not be able to see the A-party information. >> >> https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 >> >> I hope this will help. >> >> -- >> regards, >> >> abdul basit >> >> On Wed, 19 Oct 2022 at 03:14, Bela H <hob...@hotmail.com> wrote: >> >>> Hi Nitesh, >>> >>> >>> >>> 1. Check the topology hiding function: >>> https://opensips.org/docs/modules/3.2.x/topology_hiding.html >>> 2. Use e.g. rtpproxy: >>> >>> https://opensips.org/docs/modules/3.2.x/rtpproxy.html#func_rtpproxy_offer >>> >>> >>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English >>> >>> https://github.com/sippy/rtpproxy >>> >>> >>> >>> I hope these help! >>> >>> >>> >>> Cheers, >>> >>> Bela >>> >>> >>> >>> *From: *Nitesh Divecha <aviator.nites...@gmail.com> >>> *Sent: *Wednesday, 19 October 2022 04:26 >>> *To: *OpenSIPS users mailling list <users@lists.opensips.org> >>> *Subject: *[OpenSIPS-Users] - INVITE (SDP) includes Originators IP info >>> >>> >>> >>> Hello All, >>> >>> >>> >>> This is my first OpenSIPS project so I'm a newbie! >>> >>> >>> >>> After going back and forth with "uac_replace_from()", I was successfully >>> able to make a call from my ATA -> OpenSIPS -> Outbound Provider -> >>> CellPhone. All worked fine with two-way audio except few issues: >>> >>> >>> >>> 1) Outbound Provider was able to see my ATA (Originator's >>> IP/User-Agent/etc) in SIP INVITE (SDP) which kinda raised some eyebrows >>> with Outbound provider. How can I block or strip all the Originator's >>> contact info in SIP INVITE (SDP) and only send OpenSIPS info? Meaning I >>> want to protect my Originators and don't want to show anything to the >>> Outbound Provider. Outbound providers should only communicate to the >>> OpenSIPS server. >>> >>> >>> >>> 2) When the call was up I failed to capture any media/RTP on the >>> OpenSIPS server. I want to involve OpenSIPS in media/RTP between ATA and >>> outbound providers. How can I force media/RTP to pass-thru OpenSIPS IP so >>> I'm not exposing Originator's IP. >>> >>> >>> >>> Any insights will be highly appreciated. >>> >>> >>> >>> Cheers, >>> >>> Nitesh >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> _______________________________________________ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >>
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