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<p>Hi.</p>
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</p>
<p>I have a case when a peer answering a call (INVITE) is behind a NAT.</p>
<p>So, in its SIP OK message I would like to see the SDP containing a valid IP and media Port valid to receive audio from the caller. That is, the caller need to know a valid IP and Port where he/she can send his/her audio packets.</p>
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</p>
<p>1 - Is it possible to "fix" SDP content for such objective?</p>
<p>2 - <span style="font-family: Calibri, Arial, Helvetica, sans-serif, 'Apple Color Emoji', 'Segoe UI Emoji', NotoColorEmoji, 'Segoe UI Symbol', 'Android Emoji', EmojiSymbols; font-size: 16px;">Can OpenSIPS do something for this idea works or must I to use
something more like a stun server?</span></p>
<p><span style="font-size: 12pt;">3 - What is the OpenSIPS module that can help me with this task?</span></p>
<p><span style="font-size: 12pt;"><br>
</span></p>
<p><span style="font-size: 12pt;">I guess I will have to fix 2 fiels in SDP:</span></p>
<p><span style="font-size: 12pt;"></p>
<div> Media Description, name and address (m): audio 55142 RTP/AVP 8 101 ( to fix the port)</div>
<div> Connection Information (c): IN IP4 192.168.100.156 ( to fix the IP)</div>
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<p><span style="font-size: 12pt;">P.S.: I have already a solution (opensips.cfg) that let SIP messages cross NATs without problems. Only SDP has to be fixed. </span></p>
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</span></p>
<p><span style="font-size: 12pt;">Any hint will be very helpful!</span></p>
<p><span style="font-size: 12pt;">Best regards.</span></p>
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