Hello<div><br></div><div>I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway</div><div><br></div><div>And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles.</div>
<div>The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly.</div><div><br></div><div>Currently I have this setup:</div><div><br></div><div><div>API CALL [sofia(status profile external)] output:</div>
<div>========================================================</div><div>Name external</div><div>Domain Name N/A</div><div>Context public</div><div>Challenge Realm auto_to</div>
<div>RTP-IP 192.168.1.110</div><div>Ext-RTP-IP 85.89.XX.XX</div><div>SIP-IP 192.168.1.110</div><div>Ext-SIP-IP 85.89.XX.XX</div><div>OUTBOUND-PROXY N/A</div>
<div>PROXY-MEDIA false</div><div>AGGRESSIVENAT false</div><div>STUN-ENABLED true</div><div>STUN-AUTO-DISABLE false</div><div></div></div><div><br></div><div><div>API CALL [sofia(status profile default)] output:</div>
<div>========================================================</div><div>Name default</div><div>Domain Name N/A</div><div>Alias Of internal</div><div>Context public</div>
<div>Challenge Realm auto_from</div><div>RTP-IP 192.168.1.110</div><div>Ext-RTP-IP 85.89.XX.XX</div><div>SIP-IP 192.168.1.110</div><div>OUTBOUND-PROXY N/A</div>
<div>PROXY-MEDIA false</div><div>AGGRESSIVENAT false</div><div>STUN-ENABLED false</div><div>STUN-AUTO-DISABLE false</div><div><br></div><div>Sample phone registration:</div><div><div>
Call-ID: <a href="mailto:Xmbw9PyQ5Q6L2MnQ@192.168.1.121">Xmbw9PyQ5Q6L2MnQ@192.168.1.121</a></div><div>User: u1000009@default</div><div>Contact: "u1000009" <<a href="http://sip:u1000009@192.168.1.121:6094">sip:u1000009@192.168.1.121:6094</a>></div>
<div>Agent: IP PHONE 3 V1.58.004 CFG0</div><div>Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40)</div><div>Host: jonas-PC</div><div>IP: 192.168.1.121</div><div>Port: 6094</div>
<div>Auth-User: u1000009</div><div>Auth-Realm: default</div><div>MWI-Account: u1000009@default</div><div><br></div><div>Outbound INVITE:</div><div><div>send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000:</div>
<div> ------------------------------------------------------------------------</div><div> INVITE <a href="mailto:sip%3A0706930XXX@sipgw2.XXXXX.se">sip:0706930XXX@sipgw2.XXXXX.se</a> SIP/2.0</div><div> Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp</div>
<div> Max-Forwards: 69</div><div> From: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;tag=B7pve7F6eeH7c</div><div> To: <<a href="mailto:sip%3A0706930821@sipgw2.XXXXX.se">sip:0706930821@sipgw2.XXXXX.se</a>></div>
<div> Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23</div><div> CSeq: 123379614 INVITE</div><div> Contact: <<a href="http://sip:mod_sofia@192.168.1.110:5060">sip:mod_sofia@192.168.1.110:5060</a>></div><div> Call-Info: <answer-after=400></div>
<div> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY</div><div> Supported: timer, precondition, path, replaces</div>
<div> Allow-Events: talk, refer</div><div> Content-Type: application/sdp</div><div> Content-Disposition: session</div><div> Content-Length: 293</div><div> X-FS-Support: update_display</div><div> Remote-Party-ID: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;party=calling;screen=yes;privacy=off</div>
<div><br></div><div> v=0</div><div> o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110</div><div> s=FreeSWITCH</div><div> c=IN IP4 192.168.1.110</div><div> t=0 0</div><div> m=audio 24986 RTP/AVP 0 8 3 101 13</div>
<div> a=rtpmap:0 PCMU/8000</div><div> a=rtpmap:8 PCMA/8000</div><div> a=rtpmap:3 GSM/8000</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=rtpmap:13 CN/8000</div><div> a=ptime:20</div>
</div><div><br></div><div>Many thanks,</div><div> Jonas</div></div></div>