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<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">Hi <br></span></p><p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">at the
moment I have the setup<span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">Internal.xml<br>
ext-rtp-ip = auto-nat<br>
ext-sip-ip = auto-nat<span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">External.xml<br>
ext-rtp-ip = “my_local_freeswitch_ip”<br>
ext-sip-ip = auto-nat<span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">RTP ports
(only) are begin opened by UPNP and SIP via TCP is being keep open with “expire-seconds=600”
and “register=true” (Is there an alternative for the expire-seconds ? nat-options-ping is only for endpoints registering to freeswitch, right?)<br></span></p><p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif">The reason why I am doing this is, that I dont want to open up a SIP Port in the Firewall.<br></p><p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US"><span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">But this
setup up doesn’t seem right. Audio is working, inbound and outbound, both ways.
But the echo application is not.<span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">When I
change <br>
External.xml<br>
ext-rtp-ip = “auto-nat”<span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">I get
one-way audio for inbound calls for the callee. The caller can be heard. <span></span></span></p>

<p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><span lang="EN-US">I tried to analyses
it in wireshark, but I don’t understand it. The sip provider keeps sending
INVITE messages while freeswitch is confirming it with OK but keeps going on
until the connection breaks.</span></p><p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif">Thank you for your help <br></p><p class="MsoNormal" style="margin:0cm 0cm 8pt;line-height:107%;font-size:11pt;font-family:"Calibri",sans-serif"><br></p>





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