<p>Hi,<br>
Use NAT handling for clients behind NAT and may use media relaying for those clients.<br>
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.</p>
<p>Thanks<br>
Sammy</p>
<div class="gmail_quote">On Sep 11, 2012 10:54 PM, "Faisal Rehman" <<a href="mailto:faisal.rehman22@yahoo.com">faisal.rehman22@yahoo.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:10pt;font-family:tahoma,new york,times,serif"><div><span>Hi Sammy,</span></div><div style="font-style:normal;font-size:13px;background-color:transparent;font-family:tahoma,'new york',times,serif">
<span><br></span></div><div style="font-style:normal;font-size:13px;background-color:transparent;font-family:tahoma,'new york',times,serif"><span>I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show
any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?</span></div><div style="font-style:normal;font-size:13px;background-color:transparent;font-family:tahoma,'new york',times,serif">
<br></div><div> </div><div>Warmest Regards,<br><br></div><div>Faisal Rehman<br> <div style="font-family:tahoma,'new york',times,serif;font-size:10pt"> <div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
<div dir="ltr"> <font face="Arial"> <hr size="1"> <b><span style="font-weight:bold">From:</span></b> SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>><br> <b><span style="font-weight:bold">To:</span></b> OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a>>; Faisal Rehman <<a href="mailto:faisal.rehman22@yahoo.com" target="_blank">faisal.rehman22@yahoo.com</a>> <br>
<b><span style="font-weight:bold">Sent:</span></b> Tuesday, September 11, 2012 9:40 PM<br> <b><span style="font-weight:bold">Subject:</span></b> Re: [OpenSIPS-Users] Audio Issue<br> </font> </div> <br>
<div><div>Hi Faisal,<br>
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?</div>
<div> What is your network topology.? <br>
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other. </div>
<div>Regards,<br>
Sammy </div>
<div>On Sep 11, 2012 8:49 PM, "Faisal Rehman" <<a rel="nofollow" href="mailto:faisal.rehman22@yahoo.com" target="_blank">faisal.rehman22@yahoo.com</a>> wrote:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:10pt;font-family:tahoma,'new york',times,serif"><div><span>Hello Everyone!</span></div><div>
<span><br></span></div><div><span>I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?</span></div>
<div><span><br></span></div><div>
<span><br></span></div><div> </div><div>Regards,<br><br></div><div>Faisal Rehman</div></div></div><br>_______________________________________________<br>
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</div><br><br> </div> </div> </div></div></div></blockquote></div>