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<TITLE>Re: [Freeswitch-users] RTP Packet Routing</TITLE>
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<FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Just remember that bypass_media is a special mode and not everything will work when a call is in that mode... Don’t expect to do anything that remotely relies on media being in the FS box...<BR>
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<HR ALIGN=CENTER SIZE="3" WIDTH="95%"><B>From: </B>Michael Collins <msc@freeswitch.org><BR>
<B>Reply-To: </B><freeswitch-users@lists.freeswitch.org><BR>
<B>Date: </B>Tue, 25 Aug 2009 11:42:53 -0700<BR>
<B>To: </B><freeswitch-users@lists.freeswitch.org><BR>
<B>Subject: </B>Re: [Freeswitch-users] RTP Packet Routing<BR>
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On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards <jerry.richards@teotech.com> wrote:<BR>
</SPAN></FONT></FONT><BLOCKQUOTE><FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Hello All,<BR>
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I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an<BR>
active call. How do I configure it so it allows the two SIP endpoints to<BR>
send RTP packet to each other directly?<BR>
</SPAN></FONT></FONT></BLOCKQUOTE><FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'><BR>
Check out bypass media mode:<BR>
<a href="http://wiki.freeswitch.org/wiki/Bypass_Media">http://wiki.freeswitch.org/wiki/Bypass_Media</a><BR>
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-MC<BR>
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