Bela Much appreciated!
That fixed the "User-Agent" problem. Cheers, Nitesh On Wed, Oct 19, 2022 at 8:03 PM Bela H <hob...@hotmail.com> wrote: > I hope this helps: > > https://opensips.org/docs/modules/3.2.x/sipmsgops.html#func_remove_hf > > > > > > *From: *Nitesh Divecha <aviator.nites...@gmail.com> > *Sent: *Thursday, 20 October 2022 12:29 > *To: *OpenSIPS users mailling list <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> 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 <aviator.nites...@gmail.com> > *Sent: *Thursday, 20 October 2022 10:09 > *To: *Bogdan-Andrei Iancu <bog...@opensips.org> > *Cc: *OpenSIPS users mailling list <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 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 list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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