<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Using a media relay is the solution for your problem. You are asking for a solution to not use the solution which makes no sense.<div><div><div><br></div><div>Adrian</div><div> <div><div><div><div>On Feb 25, 2013, at 7:24 PM, Roberto Spadim wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr">humm i got the same problem but didn't found a solution<div>my solution was connect internet (public ip) directly to voip server, in other words, i removed the opensip proxy and ntpproxy, but if anyone have the solution could be very nice, i googled many examples but they don't work</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/2/25 Muhammad Shahzad <span dir="ltr"><<a href="mailto:shaheryarkh@gmail.com" target="_blank">shaheryarkh@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>You are missing one fundamental fact, that is you have to handle NAT for both signalling and media. From your description it looks signalling is going perfect (NAT is correctly handled), since you are able to establish call between two clients successfully, clients can register, make call, accept call and hangup call with your server. So main goal of NAT Traversal module is achieved. </div>
<div><br></div><div>However, there is no media on call, so media NAT is not handled. NAT Traversal and / or NAT Helper modules may try to fix media NAT issues as well by manipulating SDP but in so many case they will be simply NOT enough for this purpose. Especially in case of 3g and corporate networks, which may have very very complex network typology with multiple layers of NAT (so called Nested NAT). So rtp / media proxy is the ONLY solution that can handle media across such complex networks.</div>
<div><br></div><div>If you have really good sip clients with support for STUN / TURN / ICE etc. and you somewhat control over client data network environment, them you may fix media NAT issues up to 90% but in about 5-10% cases you will still need a media relay.</div>
<div><br></div>Thank you.<div><div><div class="h5"><br><br><div class="gmail_quote">On Mon, Feb 25, 2013 at 11:51 PM, leo <span dir="ltr"><<a href="mailto:uzcudunl@yahoo.it" target="_blank">uzcudunl@yahoo.it</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
<br>
Unfortunately after reading the forum i've to open a new post about NAT<br>
because i couldn't find a clear solution and information for my problem.<br>
I've also read the NAT Traversal module documentation.<br>
<br>
I've an OpenSIPS server (version 1.8.2) on a Debian system (6.0.7 -<br>
2.6.32-5-686).<br>
OpenSIPS was installed by the apt-get install using the <a href="http://apt.opensips.org/" target="_blank">apt.opensips.org</a><br>
repository and configured with osipsconfig (residential script with ALIASES,<br>
AUTH, DBACC, DBUSRLOC and DIALOG).<br>
<br>
The UAs can register to the OpenSIPS server. They can place the call but i<br>
've no audio no video.<br>
The OpenSIPS server has a public IP address (so, no natted).<br>
The UAs could be natted or with public ip thru 3G.<br>
<br>
I wouldn't like to use rtproxy or mediaproxy cause the rtp traffic would be<br>
passing by those servers (am i correct?) adding jitter and latency.<br>
I would set up the system in the way the the rtp traffic would be P2P. Would<br>
NAT Traversal be the solution? How it should be configured (i've already<br>
enabled the required modules too)?<br>
<br>
Thanks a lot.<br>
<br>
Leo.<br>
<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div></div></div><span class="HOEnZb"><font color="#888888">-- <br><span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">Muhammad Shahzad</span><br style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
<span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">------------------------------</span><span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">-----</span><br style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
<span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">CISCO Rich Media Communication Specialist (CRMCS)</span><br style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
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<span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">Cell: <a href="tel:%2B49%20176%2099%2083%2010%2085" value="+4917699831085" target="_blank">+49 176 99 83 10 85</a></span><br style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
<span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">MSN: </span><a href="mailto:shari_786pk@hotmail.com" style="color:rgb(17,85,204);font-size:13px;font-family:arial,sans-serif" target="_blank">shari_786pk@hotmail.com</a><br style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
<span style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">Email: </span><a href="mailto:shaheryarkh@googlemail.com" style="color:rgb(17,85,204);font-size:13px;font-family:arial,sans-serif" target="_blank">shaheryarkh@googlemail.com</a>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Roberto Spadim<br>SPAEmpresarial</div>
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