<div dir="ltr">Hello.<br><br>When using a call or conference through sip — freeswitch with external provider there is a problem – the call is interrupted in 30 seconds. Though the sound goes all right.<br>I think that it caused by the NAT settings for freeswitch, but I don&#39;t understand how to adjust it correctly.<br>At start of freeswitch I see the following mistakes in the tracking data:<br><font size="2">2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT<br>2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5<br>2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5<br>2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5<br>2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5<br>2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5<br>2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP [general error]<br>2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP<br>2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected!</font><br><br>Despite of this mistake, conference communication between two internal users works normally. The problem arises at a call through external provider.<br><br>We have the following architecture:<br>In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch, both for NAT for external clients, but have an opportunity to work with each other directly.<br>opensips has the internal address 172.31.0.169 and external 52. *.*.177<br>freeswitch has the internal address 172.31.22.124 and external 52. *.*.198<br><br>In fact, freeswitch acts only for conferences, and is ready for use of a remote DB on opensips.<br>The auto-nat settings by default didn&#39;t work. The problem is in the external profile settings as far as I understand.<br><br>I have filled and created the following configuration:<br>vars.xml <br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;bind_server_ip=auto”/&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_rtp_ip=52.*.*.198”/&gt; &lt;!— public freeswitch ip —&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_sip_ip=52.*.*.198”/&gt; &lt;!— public freeswitch ip —&gt;<br>  &lt;!-- External SIP Profile --&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_auth_calls=true&quot;/&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_sip_port=5060&quot;/&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_tls_port=5061&quot;/&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_ssl_enable=true&quot;/&gt;<br>  &lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;external_ssl_dir=$${base_dir}/conf/tls&quot;/&gt;<br><br>sip_profile/external.xml<br>    &lt;param name=&quot;rtp-ip&quot; value=&quot;$${local_ip_v4}&quot;/&gt;<br>    &lt;param name=&quot;sip-ip&quot; value=&quot;$${local_ip_v4}&quot;/&gt;<br><br>    &lt;param name=&quot;ext-rtp-ip&quot; value=“52.*.*.198”/&gt; &lt;!— public freeswitch ip —&gt;<br>    &lt;param name=&quot;ext-sip-ip&quot; value=“52.*.*.198”/&gt; &lt;!— public freeswitch ip —&gt;<br>In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server (that would be logical), but in that case conferences didn&#39;t work at all and errors below appeared:<br>[ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...<br>Also I tried to put such configuration:<br><span style="color:rgb(0,0,0)">    &lt;param name=&quot;rtp-ip&quot; value=&quot;auto&quot;/&gt;<br>    &lt;param name=&quot;sip-ip&quot; value=&quot;52.*.*.198”/&gt;</span><br>but it also hasn&#39;t helped to solve the problem.<br><br>autoload_configs/switch.conf.xml <br>    &lt;param name=&quot;rtp-start-port&quot; value=&quot;16384&quot;/&gt;<br>    &lt;param name=&quot;rtp-end-port&quot; value=&quot;32768&quot;/&gt;<br><br>&quot;sofia status&quot; looks as follows:<br>                     Name       Type                                          Data    State<br>=================================================================================================<br>            172.31.22.124      alias                                      internal    ALIASED<br>                 external    profile               sip:mod_sofia@52.*.*.198:5060    RUNNING (0)<br>                 external    profile               sip:mod_sofia@52.*.*.198:5061    RUNNING (0) (TLS)<br> external::*********.com    gateway                      sip:USER@*********.com    REGED<br>                 internal    profile               sip:mod_sofia@52.*.*.198:5080    RUNNING (0)<br>                 internal    profile               sip:mod_sofia@52.*.*.198:5081    RUNNING (0) (TLS)<br>=================================================================================================<br>2 profiles 1 alias<br><br>&quot;sofia status profile external&quot; looks as follows:<br>=================================================================================================<br>Name                 external<br>Domain Name          N/A<br>Auto-NAT             false<br>DBName               sofia_reg_external<br>Pres Hosts           <br>Dialplan             XML<br>Context              public<br>Challenge Realm      auto_to<br>RTP-IP               172.31.22.124<br>Ext-RTP-IP           52.*.*.198<br>SIP-IP               172.31.22.124<br>Ext-SIP-IP           52.*.*.198<br>URL                  sip:mod_sofia@52.*.*.198:5060<br>BIND-URL             sip:mod_sofia@52.*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp<br>TLS-URL              sip:mod_sofia@52.*.*.198:5061<br>TLS-BIND-URL         sips:mod_sofia@52.*.*.198:5061;maddr=172.31.22.124;transport=tls<br>HOLD-MUSIC           local_stream://moh<br>OUTBOUND-PROXY       N/A<br>CODECS IN            PCMA<br>CODECS OUT           PCMA<br>TEL-EVENT            101<br>DTMF-MODE            rfc2833<br>CNG                  13<br>SESSION-TO           0<br>MAX-DIALOG           0<br>NOMEDIA              false<br>LATE-NEG             true<br>PROXY-MEDIA          false<br>ZRTP-PASSTHRU        true<br>AGGRESSIVENAT        false<br>CALLS-IN             0<br>FAILED-CALLS-IN      0<br>CALLS-OUT            0<br>FAILED-CALLS-OUT     0<br>REGISTRATIONS        0<br><br><br><br>What do I adjust wrong? Whether there is some opportunity, to tell freeswitch not to break off a call in 30 seconds even if NAT isn&#39;t adjusted?<br></div>