I hope this helps: https://opensips.org/docs/modules/3.2.x/sipmsgops.html#func_remove_hf
From: Nitesh Divecha<mailto:aviator.nites...@gmail.com> Sent: Thursday, 20 October 2022 12:29 To: OpenSIPS users mailling list<mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP info Bela, Much appreciated! Changing topolgy_hiding("C"); fixed the Call-ID issue. Call-ID shows clean Call-ID: DLGCH_W0xtTFgVXWleUV1fVgFvEiVSRVdabgccAltXbUFf Now gotta figure out how to stop sending Originator User-Agent to outbound provider and how to configure rtpproxy behind NAT. Cheers, Nitesh On Wed, Oct 19, 2022 at 5:35 PM Bela H <hob...@hotmail.com<mailto:hob...@hotmail.com>> wrote: For the first problem check this: C - Encode the callid header Note: Changing the callid of the call using the "C" flag is only available when doing topology_hiding with dialog support. Using this flag without dialog support will not change the callid at all!. From: Nitesh Divecha<mailto:aviator.nites...@gmail.com> Sent: Thursday, 20 October 2022 10:09 To: Bogdan-Andrei Iancu<mailto:bog...@opensips.org> Cc: OpenSIPS users mailling list<mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP info 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<http://sip:aaabbbc...@outboundprovider.com:5060> SIP/2.0 Call-ID: 4ed41738da10faa5@172.16.16.250<mailto:4ed41738da10faa5@172.16.16.250> <<<-- showing originators Device LAN IP —>>> Content-Length: 329 CSeq: 8002 INVITE From: <sip:zzzzzzz...@outboundprovider.com<mailto:sip%3azzzzzzz...@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<http://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<mailto: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<mailto: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<mailto: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<mailto:aviator.nites...@gmail.com> Sent: Wednesday, 19 October 2022 04:26 To: OpenSIPS users mailling list<mailto: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<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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