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<DIV>For a hosted environment, where you're not in control of the users devices/routers you should do the following:</DIV>
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<DIV>In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true. </DIV>
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<DIV>That should work every time.</DIV>
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<DIV>--Dave</DIV><BR>
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<B>From:</B> Carlo Dimaggio [mailto:jaasmailing@gmail.com]<BR><B>To:</B> freeswitch-users@lists.freeswitch.org<BR><B>Sent:</B> Wed, 06 Jun 2012 13:13:08 -0700<BR><B>Subject:</B> Re: [Freeswitch-users] RTP NAT issue<BR><BR>Ok, I'll try the parameter NDLB. Is it suitable in an hosted environment (thousand of extensions)?<BR><BR><BR>Anyway, the problem is that I can't use STUN (phone side configuration) because I have two needs:<BR><BR>1) Phone A (192.168.0.100) calls Phone B (192.168.0.101), in this case the RTP flow should be sent directly between two endpoints. If I use STUN the SDP "A" contains the public IP and not the private IP (that must be used for device reachability).<BR><BR>2) Phone A (192.168.0.100) calls Freeswitch (Public IP), in this case the RTP flow shoud be sent to Freeswitch Public IP and Freeswitch shoud send the RTP to the Phone A NAT IP. Here the STUN is the best solution.<BR><BR>What is the best practice in this scenario?<BR> <BR>Regards,<BR><BR><BR>Il 06/06/12 18.28, Brian Foster ha scritto:
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<P>The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.</P>
<P>Brian Foster<BR>Endigo Computer LLC</P>
<P>Sent from a mobile device.</P>
<DIV class=gmail_quote>On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <<A href="mailto:jaasmailing@gmail.com">jaasmailing@gmail.com</A>> wrote:<BR type="attribution">
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<DIV bgcolor="#FFFFFF" text="#000000"><FONT size=-1><FONT face=Arial>Hi all,<BR><BR>I have a problem with RTP and NAT.<BR>The scenario is Hosted PBX and Natted phones (yealink):<BR><BR>Phones (192.168.0.x) - NAT -> FS (public IP) <BR><BR>When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP.<BR>The registration is ok:</FONT></FONT><BR></DIV></BLOCKQUOTE></DIV></BLOCKQUOTE><BR></BLOCKQUOTE>
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