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

Wed Oct 19 21:06:07 UTC 2022

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
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
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