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<title>Re: [Freeswitch-users] SIP Trace Question / NAT one way audio problem.</title>
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<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">Just a follow up with the confirmed resolution.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">I had copied the FS Directory entries for these Polycom Phones from a CISCO phone setup.&nbsp; That setup specified:<o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span style="font-size:9.5pt;font-family:Consolas;color:blue">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;</span><span style="font-size:9.5pt;font-family:Consolas;color:#A31515">param</span><span style="font-size:9.5pt;font-family:Consolas;color:blue">
</span><span style="font-size:9.5pt;font-family:Consolas;color:red">name</span><span style="font-size:9.5pt;font-family:Consolas;color:blue">=</span><span style="font-size:9.5pt;font-family:Consolas">&quot;<span style="color:blue">NDLB-force-rport</span>&quot;<span style="color:blue">
</span><span style="color:red">value</span><span style="color:blue">=</span>&quot;<span style="color:blue">true</span>&quot;<span style="color:blue">/&gt;</span><o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span style="font-size:9.5pt;font-family:Consolas"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">Ken adviced to change that for Polycom Phones to:<o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span style="font-size:9.5pt;font-family:Consolas;color:blue">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;</span><span style="font-size:9.5pt;font-family:Consolas;color:#A31515">param</span><span style="font-size:9.5pt;font-family:Consolas;color:blue">
</span><span style="font-size:9.5pt;font-family:Consolas;color:red">name</span><span style="font-size:9.5pt;font-family:Consolas;color:blue">=</span><span style="font-size:9.5pt;font-family:Consolas">&quot;<span style="color:blue">NDLB-force-rport</span>&quot;<span style="color:blue">
</span><span style="color:red">value</span><span style="color:blue">=</span>&quot;<b><span style="color:blue">safe</span></b>&quot;<span style="color:blue">/&gt;</span><o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span style="font-size:9.5pt;font-family:Consolas"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">The customer emailed me today to say the problem has been resolved.&nbsp; Thanks for all who sent their thoughts.&nbsp; Extra thanks to Ken.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">Sean<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><o:p>&nbsp;</o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Ken Rice<br>
<b>Sent:</b> Thursday, December 05, 2013 9:48 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] SIP Trace Question / NAT one way audio problem.<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;">Re-registering every 23 seconds isnt that big of a deal if the phones are behind nat, sometimes this is needed due to the way NAT happens on UDP with
 various routers. There is no way to track the start/end of the session w/out using an ALG. I have seen some NAT routers require a re-reg interval around 15 seconds which is a bit excessing<br>
<br>
On the issue w/ the non-working ones is you&#8217;ll notice the rport on the one that&#8217;s working...<br>
<br>
On the ones that arent working they arent specifying rport... If those are the polycoms look for NDLB force rport setting and set it to safe... That&#8217;s a special mode for the polycoms...<br>
<br>
Check the wiki for more NAT handling examples<br>
K<br>
<br>
<br>
On 12/5/13 8:36 PM, &quot;Sean P Devoy&quot; &lt;<a href="sdevoy@bizfocused.com">sdevoy@bizfocused.com</a>&gt; wrote:</span><o:p></o:p></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><img border="0" width="31" height="31" id="_x0000_i1025" src="cid:image001.gif@01CEFD76.90387370"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">Hi all,<br>
&nbsp;<br>
I seem to have fallen off the list! &nbsp;No mail since 7/26/2013. &nbsp;Have I made someone mad?? &nbsp;</span><span style="font-family:Wingdings;color:#44546A">J<br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><br>
I have also been blissfully running with ISSUES what so ever since that time.<br>
&nbsp;<br>
I have a NATed client with <b>PERIODIC</b> one way audio. &nbsp;The client end has a Cisco 220W router with SIP ALG disabled. &nbsp;The phones are CISCO 504Gs and Polycom 330s(?), basciallt the same as everywhere else we have phones.<br>
&nbsp;<br>
I did a sofia global sip trace on for a while and captured the output. &nbsp;After writing an app to sort those sip packets into logical stream files, I dug in to the SIP conversations. &nbsp;This has left me with som questions for the people who know these things:<br>
&nbsp;<br>
1. &nbsp;&nbsp;I found all my phones are REGISTERing every 23 seconds! &nbsp;&nbsp;That seems absurd TO ME, so where have I screwed that up/where do I set it and what is a reasonable number?<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">2. &nbsp;&nbsp;The only difference I see in the SIP packets from my working phone and these non-working ones is of course NAT related. &nbsp;The working phone&#8217;s REGISTER looks like this:<br>
<br>
recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203:<br>
------------------------------------------------------------------------ <br>
REGISTER sip: &lt;MYDOMAIN.COM&gt; SIP/2.0<br>
Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport<br>
<br>
But the failing phones look like this:<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691:<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">------------------------------------------------------------------------<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">REGISTER sip: &lt;MYDOMAIN.COM&gt; SIP/2.0<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A">The response is still sent to [AA.BB.CC.38]:5104<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><br>
</span><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#44546A"><br>
I guess first, IS THAT A PROBLEM?<br>
&nbsp;<br>
So, can anyone share any insight into the CISCO 220W config that I might be missing? &nbsp;<br>
&nbsp;<br>
Why would it only be a periodic problem? &nbsp;Could this be a problem where one end is using a wider range of rports than the other supports?<br>
&nbsp;<br>
Thanks,<br>
Sean<br>
&nbsp;<br>
<br>
</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;"><o:p></o:p></span></p>
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</span><span style="font-size:11.0pt;font-family:&quot;Courier New&quot;">-- <br>
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