<div dir="ltr">Sounds like your gateway does SIP ALG which in my experience breaks SIP in 99 percent of all communication (because it fixes where nothing is broken).<div>Had such problems when I specified a STUN server for clients - NAT traversal should be left to the server alone. It's perfectly fine to use private IPs in SIP/SDN and let RPORT handle it all.</div><div><br></div><div>Kevin<br><div><br><br><div class="gmail_quote"><div dir="ltr">Am Do., 15. Nov. 2018 um 23:51 Uhr schrieb Paul Muaddib <<a href="mailto:paul.muaddib83@gmail.com">paul.muaddib83@gmail.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div dir="ltr"><div dir="ltr">
<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>
</div></div></div>
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