<div dir="ltr"><div><div><div>Hi,<br><br></div>Please clear me on this topic I am confused <br><br></div>My log show "switching to native rtp". <br>Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? <br>
<br></div>Am I right or wrong ?<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani <span dir="ltr"><<a href="mailto:mitul@enterux.in" target="_blank">mitul@enterux.in</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">No way to avoid bw charges for any of the client if it is behind any sort of NAT.</p><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On 08-Jul-2014 8:52 PM, "Sameer Rathod" <<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.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 dir="ltr"><div><div><div>Hi Eric,<br><br><br></div>I am behind nat <br><br></div>Is there any solution for the same.<br><br></div>My goal is to deduct the balance <br>for the call but free my asterisk server from audio packet load.<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I think you will find that direct audio between two endpoints does not work when NAT is involved. <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Sameer Rathod<br>
<b>Sent:</b> Tuesday, July 08, 2014 11:18 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] packet2packet bridging<u></u><u></u></span></p></div><div><div>
<p class="MsoNormal"><u></u> <u></u></p><div><div><div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">Hi Joshua,<br><br><u></u><u></u></p></div><p class="MsoNormal">I had disabled <u></u><u></u></p></div>
<p class="MsoNormal">ice support and remover encryption= yes<u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">Then also it is showing the same native_rtp in log<u></u><u></u></p></div><p class="MsoNormal">
Could you help me in bypassing asterisk server for audio?<u></u><u></u></p></div><p class="MsoNormal">please help me I am struggling with it form a long time. <u></u><u></u></p><div><div><div><div><p class="MsoNormal"><u></u> <u></u></p>
</div></div></div></div></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod <<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>> wrote:<u></u><u></u></p>
<div><div><p class="MsoNormal" style="margin-bottom:12.0pt"> -- Channel SIP/1060-0000008e left 'native_rtp' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b><br> -- Channel SIP/1061-0000008f left 'native_rtp' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b><br>
== Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-0000008e'<br><br><u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">here are more generated when I cut the call <br><br><u></u><u></u></p>
</div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod <<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>> wrote:<u></u><u></u></p>
<div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">so In this case If I disable ice support <u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">ie commented the icesuppot=yes from all files <u></u><u></u></p>
</div><p class="MsoNormal">then also I am getting this output<br><br><br>-- Executing [1061@sameer:1] Dial("SIP/1060-0000008e", "SIP/1061") in new stack<u></u><u></u></p><div><p class="MsoNormal"><br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/1061<u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"> -- SIP/1061-0000008f is ringing<br> -- SIP/1061-0000008f answered SIP/1060-0000008e<br> -- Channel SIP/1061-0000008f joined 'simple_bridge' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b><br>
-- Channel SIP/1060-0000008e joined 'simple_bridge' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b><br> > Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp<br>
> 0x7f6800039020 -- Probation passed - setting RTP source address to <a href="http://192.168.1.176:8000" target="_blank">192.168.1.176:8000</a><br> > 0x7f6780045810 -- Probation passed - setting RTP source address to <a href="http://192.168.1.191:8000" target="_blank">192.168.1.191:8000</a><br>
<br><br><br><br><u></u><u></u></p></div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>> wrote:<u></u><u></u></p>
<div><p class="MsoNormal">Sameer Rathod wrote:<u></u><u></u></p><p class="MsoNormal" style="margin-bottom:12.0pt">yes I had configured<br><br>icesupport=yes ;<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p></div>
<p class="MsoNormal">Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.<u></u><u></u></p><div><div><p class="MsoNormal"><br><br>-- <br>Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
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</div></div></div><p class="MsoNormal"><br><br clear="all"><u></u><u></u></p></div></div><div><p class="MsoNormal">-- <u></u><u></u></p><div><p class="MsoNormal">Regards<u></u><u></u></p></div><p class="MsoNormal">Sameer Rathod<u></u><u></u></p>
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<br clear="all"><br>-- <u></u><u></u></p><div><p class="MsoNormal">
Regards<u></u><u></u></p></div><p class="MsoNormal">Sameer Rathod<u></u><u></u></p><div><p class="MsoNormal"><a href="tel:8109413462" value="+918109413462" target="_blank">8109413462</a> <u></u><u></u></p><div><p class="MsoNormal">
<u></u> <u></u></p></div></div></div></div></div></div><p class="MsoNormal">
<br><br clear="all"><br>-- <u></u><u></u></p><div><p class="MsoNormal">Regards<u></u><u></u></p></div><p class="MsoNormal">Sameer Rathod<u></u><u></u></p><div><p class="MsoNormal"><a href="tel:8109413462" value="+918109413462" target="_blank">8109413462</a> <u></u><u></u></p>
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<u></u> <u></u></p></div></div></div></div></div></div></div><br>--<br>
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