[OpenSIPS-Users] - INVITE (SDP) includes Originators IP info

Thu Oct 20 15:31:32 UTC 2022

Bela

Much appreciated!

That fixed the "User-Agent" problem.

Cheers,
Nitesh

On Wed, Oct 19, 2022 at 8:03 PM Bela H <hobe69 at hotmail.com> wrote:

> I hope this helps:
>
> https://opensips.org/docs/modules/3.2.x/sipmsgops.html#func_remove_hf
>
>
>
>
>
> *From: *Nitesh Divecha <aviator.nitesh.d at gmail.com>
> *Sent: *Thursday, 20 October 2022 12:29
> *To: *OpenSIPS users mailling list <users at 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 <hobe69 at 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.nitesh.d at gmail.com>
> *Sent: *Thursday, 20 October 2022 10:09
> *To: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Cc: *OpenSIPS users mailling list <users at 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:aaabbbcccc at outboundprovider.com:5060 SIP/2.0
>
> Call-ID: 4ed41738da10faa5 at 172.16.16.250 *<<<-- showing originators Device
> LAN IP —>>>*
>
> Content-Length: 329
> CSeq: 8002 INVITE
> From: <sip:zzzzzzzzzz at outboundprovider.com>;tag=SP39b79130abfb7487f
> Max-Forwards: 69
> To: <sip: aaabbbcccc at 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.nitesh.d at 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 <bogdan at 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 <hobe69 at 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.nitesh.d at gmail.com>
> *Sent: *Wednesday, 19 October 2022 04:26
> *To: *OpenSIPS users mailling list <users at 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
>
>
>
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