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

Thu Oct 20 15:38:49 UTC 2022

Hello,

After reading the rtpproxy documentation again, I was able to resolve the
rtpproxy NAT issue.

-A *advaddr1[/advaddr2]*

Set advertised address of rtpproxy. Useful if the rtpproxy is behind a NAT
firewall. (Amazon EC2) When the rtpproxy receives a session request from a
SIP controller it will return the IP address(es) specified by the -A option.

CGroup: /system.slice/rtpproxy.service

             └─247521 /usr/local/bin/rtpproxy -s udp:172.31.29.47 22222 -u
rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -A 3.xxx.xxx.49 -l
172.31.29.47 -m 1000 -M 2000 -d INFO LOG_LOCAL5

====

Just for my understanding... What is the difference between rtpproxy and
Mediaproxy Module in OpenSIPS? Do I need both or can I achieve the same
with Mediaproxy? I have to monitor two services now (rtpproxy and OpenSIPS).

Cheers,
Nitesh

On Wed, Oct 19, 2022 at 5:06 PM Nitesh Divecha <aviator.nitesh.d at gmail.com>
wrote:

> 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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