<div dir="ltr">Hello, <div><br></div><div>After reading the rtpproxy documentation again, I was able to resolve the rtpproxy NAT issue. </div><div><br></div><div><dt style="color:rgb(0,0,0)"><span class="gmail-term"><code class="gmail-option">-A</code><span class="gmail-Apple-converted-space"> </span><em class="gmail-replaceable"><code>advaddr1[<span class="gmail-optional">/advaddr2</span>]</code></em></span></dt><dd style="color:rgb(0,0,0)"><p>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<span class="gmail-Apple-converted-space"> </span><code class="gmail-option">-A</code><span class="gmail-Apple-converted-space"> </span>option.</p></dd></div><div><br></div>CGroup: /system.slice/rtpproxy.service<br><br> └─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<br><br><div>====</div><div><br></div><div>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).</div><div><br></div><div>Cheers,</div><div>Nitesh</div><div><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Oct 19, 2022 at 5:06 PM Nitesh Divecha <<a href="mailto:aviator.nitesh.d@gmail.com">aviator.nitesh.d@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hello All, <div><br></div><div>So I had some success using topology_hiding and rtpproxy but found few problems. </div><div><br></div><div>After implementing topology_hiding(), SIP INVITE was much better but still showing following: </div><div><br></div><div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px">INVITE <a href="http://sip:aaabbbcccc@outboundprovider.com:5060" target="_blank">sip:aaabbbcccc@outboundprovider.com:5060</a> SIP/2.0</div><div style="font-family:Helvetica;font-size:12px"><font color="#000000">Call-ID: <a href="mailto:4ed41738da10faa5@172.16.16.250" target="_blank">4ed41738da10faa5@172.16.16.250</a> </font><b><font color="#ff0000"><<<-- showing originators Device LAN IP —>>></font></b></div><div style="font-family:Helvetica;font-size:12px"><font color="#000000">Content-Length: 329</font><br><font color="#000000">CSeq: 8002 INVITE</font><br><font color="#000000">From: <<a href="mailto:sip%3Azzzzzzzzzz@outboundprovider.com" target="_blank">sip:zzzzzzzzzz@outboundprovider.com</a>>;tag=SP39b79130abfb7487f</font><br><font color="#000000">Max-Forwards: 69</font><br><font color="#000000">To: <sip: aaabbbcccc@3.xxx.xxx.49></font><br><font color="#000000">Via: SIP/2.0/UDP 3.xxx.xxx.49:5060;branch=z9hG4bK1dcb.5bb78035.0</font><br><font color="#000000">User-Agent: OBIHAI/OBi302-3.2.2.6259 </font><b><font color="#ff0000"><<<-- showing originators User-Agent —>>></font></b><br><font color="#000000">Contact: <sip:3.xxx.xxx.49;did=6a7.5e849703></font><br><font color="#000000">Expires: 60</font><br><font color="#000000">Supported: replaces</font><br><font color="#000000">Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE</font><br><font color="#000000">Content-Type: application/sdp</font><br><br>===</div><div style="font-family:Helvetica;font-size:12px">1) How can I remove IP from Call-ID and rewrite Originators User-Agent to local OpenSIPS User-Agent?</div><div style="font-family:Helvetica;font-size:12px">===</div><div style="font-family:Helvetica;font-size:12px"><br></div><div style="font-family:Helvetica;font-size:12px"><br></div><div style="font-family:Helvetica;font-size:12px">Now issue with rtpproxy - I'm running this OpenSIPS on AWS cloud... AWS cloud does natting by default, so my Public IP is <span style="color:rgb(0,0,0)">3.xxx.xxx.49 and actual VM IP is </span><b style="color:rgb(0,0,0)">172.31.29.47. </b></div><div style="font-family:Helvetica;font-size:12px"><b style="color:rgb(0,0,0)"><br></b></div><div><font face="Helvetica"><span style="font-size:12px">After implement rtpproxy (<a href="https://www.rtpproxy.org/" target="_blank">https://www.rtpproxy.org/</a>), it is running on local IP: </span></font> </div><div>└─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<br><br></div><div>As it shows from SIP INVITE and due to that no audio or RTP because IP is not reachable... </div><div><br></div><div style="font-family:Helvetica;font-size:12px"><font color="#000000">v=0</font><br><font color="#000000">o=- 16210664 1 IN IP4 <b>172.31.29.47 <<<-- OpenSIPS NAT IP —>>></b></font><br><font color="#000000">s=-</font><br><font color="#000000">c=IN IP4 <b>172.31.29.47 <<<-- OpenSIPS NAT IP —>>></b></font><br><font color="#000000">t=0 0</font><br><font color="#000000">m=audio 1958 RTP/AVP 0 8 18 104 101</font><br><font color="#000000">a=rtpmap:0 PCMU/8000</font><br><font color="#000000">a=rtpmap:8 PCMA/8000</font><br><font color="#000000">a=rtpmap:18 G729/8000</font><br><font color="#000000">a=rtpmap:104 G726-32/8000</font><br><font color="#000000">a=rtpmap:101 telephone-event/8000</font><br><font color="#000000">a=fmtp:101 0-15</font><br><font color="#000000">a=sendrecv</font><br><font color="#000000">a=ptime:20</font><br><font color="#000000">a=xg726bitorder:big-endian</font><br><font color="#000000">a=nortpproxy:yes</font><br></div></div><div><br></div><div>===</div><div>2. How can I configure rtpproxy with Public IP? Or do I start rtpproxy with <span style="font-family:Helvetica;font-size:12px">Public IP </span><span style="font-family:Helvetica;font-size:12px;color:rgb(0,0,0)">3.xxx.xxx.49 and reconfigure OpenSIPS with Public IP?</span></div>modparam("rtpproxy", "rtpproxy_sock", "udp:<a href="http://172.31.29.47:22222" target="_blank">172.31.29.47:22222</a>") <div><br></div><div>Thanking in advance... </div><div><br></div><div>Cheers,</div><div>Nitesh</div><div><br></div><div><br></div><div><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Oct 19, 2022 at 10:17 AM Nitesh Divecha <<a href="mailto:aviator.nitesh.d@gmail.com" target="_blank">aviator.nitesh.d@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hello, <div><br></div><div>Thank y'all for the input... I will try to read the documentation and work on implementing these modules. </div><div><br></div><div>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! :-) </div><div><br></div><div>Thanking in advance... </div><div><br></div><div>Cheers,</div><div>Nitesh</div><div><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Oct 19, 2022 at 2:20 AM Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<div>
<font face="monospace">Hi there,<br>
<br>
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.<br>
<br>
Regards,<br>
</font>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 5-16 Dec 2022, online
<a href="https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/" target="_blank">https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/</a></pre>
<div>On 10/19/22 1:23 AM, Abdul Basit wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Nitesh,
<div><br>
</div>
<div>You need a <a href="https://www.tutorialspoint.com/session_initiation_protocol/session_initiation_protocol_b2bua.htm" target="_blank">B2BUA function</a> with the help of a
topo-hiding module with opensips as Bela shared in his email.</div>
<div>Also, install the RTP proxy on the same opensips box (not
necessary if you need separate signaling and media boxes).</div>
<div><br>
</div>
<div>Far end party will not be able to see the A-party
information. </div>
<div> </div>
<div><a href="https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2" target="_blank">https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2<br>
</a></div>
<div><br>
</div>
<div>I hope this will help. </div>
<div><br clear="all">
<div>
<div dir="ltr">
<div><font size="2"><span style="font-family:verdana,sans-serif"><span style="color:rgb(39,78,19)">--<br>
regards,</span></span></font></div>
<font size="2"><span style="font-family:verdana,sans-serif"><span style="color:rgb(39,78,19)"><br>
abdul basit</span></span></font></div>
</div>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Wed, 19 Oct 2022 at 03:14,
Bela H <<a href="mailto:hobe69@hotmail.com" target="_blank">hobe69@hotmail.com</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<div>
<div lang="EN-NZ">
<div>
<p class="MsoNormal">Hi Nitesh,</p>
<p class="MsoNormal"> </p>
<ol style="margin-top:0cm" type="1" start="1">
<li style="margin-left:0cm">Check the topology hiding
function: <a href="https://opensips.org/docs/modules/3.2.x/topology_hiding.html" target="_blank">https://opensips.org/docs/modules/3.2.x/topology_hiding.html</a></li>
<li style="margin-left:0cm">Use e.g. rtpproxy:
</li>
</ol>
<p><a href="https://opensips.org/docs/modules/3.2.x/rtpproxy.html#func_rtpproxy_offer" target="_blank">https://opensips.org/docs/modules/3.2.x/rtpproxy.html#func_rtpproxy_offer</a></p>
<p><a href="http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English" target="_blank">http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English</a></p>
<p><a href="https://github.com/sippy/rtpproxy" target="_blank">https://github.com/sippy/rtpproxy</a></p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I hope these help!</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Cheers,</p>
<p class="MsoNormal">Bela</p>
<p class="MsoNormal"> </p>
<div style="border-style:solid none none;border-top-width:1pt;border-top-color:rgb(225,225,225);padding:3pt 0cm 0cm">
<p class="MsoNormal" style="border:none;padding:0cm"><b>From:
</b><a href="mailto:aviator.nitesh.d@gmail.com" target="_blank">Nitesh
Divecha</a><br>
<b>Sent: </b>Wednesday, 19 October 2022 04:26<br>
<b>To: </b><a href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS
users mailling list</a><br>
<b>Subject: </b>[OpenSIPS-Users] - INVITE (SDP)
includes Originators IP info</p>
</div>
<p class="MsoNormal"> </p>
<div>
<p class="MsoNormal">Hello All, </p>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">This is my first OpenSIPS
project so I'm a newbie! </p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">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: </p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">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. </p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">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.</p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">Any insights will be highly
appreciated. </p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">Cheers, </p>
</div>
</div>
<p class="MsoNormal">Nitesh</p>
<p class="MsoNormal"> </p>
</div>
</div>
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</div>
</blockquote>
</div>
<br>
<fieldset></fieldset>
<pre>_______________________________________________
Users mailing list
<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
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</pre>
</blockquote>
<br>
</div>
</blockquote></div>
</blockquote></div>
</blockquote></div>