Good day all.
I'm experiencing the following sinptom when using some softphones behind nat on a private LAN, sometimes the same happen with hardphones.
Here's the scenario:
FS : <public IP> _______ <public IP>LAN ROUTER <private LAN> --- softphone
Basically when softphone makes an INVITE to FS it always sends the private IP on the SDP and when the media flow starts it's being sent out by FS to the public lan address resulting on a audioless call. However if the phone sends the public IP on the SDP there's no issue at all.
I know there's a variable available disable_rtp_auto_adjust that shoud make freeswitch ignore the SDP IP and use the INVITE IP instead, but it isn't working for me.
Here's what i currently have on my internal SIP profile:
<profile name="internal">
<aliases>
</aliases>
<gateways>
</gateways>
<domains>
<domain name="all" alias="true" parse="false"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<param name="watchdog-enabled" value="no"/>
<param name="watchdog-step-timeout" value="30000"/>
<param name="watchdog-event-timeout" value="30000"/>
<param name="log-auth-failures" value="true"/>
<param name="forward-unsolicited-mwi-notify" value="false"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="H264,G722,PCMA,GSM"/>
<param name="outbound-codec-prefs" value="H264,G722,PCMA,GSM"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="<PUBLIC_IP>"/>
<param name="sip-ip" value="<PUBLIC_IP>"/>
<param name="hold-music" value="local_stream://moh"/>
<param name="apply-inbound-acl" value="domains"/>
<param name="apply-nat-acl" value="rfc1918"/>
<param name="local-network-acl" value="localnet.auto"/>
<param name="record-path" value="/opt/freeswitch/recordings"/>
<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<param name="manage-presence" value="true"/>
<param name="presence-privacy" value=""/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="tls" value="true"/>
<param name="tls-only" value="false"/>
<param name="tls-bind-params" value="transport=tls"/>
<param name="tls-sip-port" value="5061"/>
<param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/>
<param name="tls-passphrase" value=""/>
<param name="tls-verify-date" value="true"/>
<param name="tls-verify-policy" value="none"/>
<param name="tls-verify-depth" value="2"/>
<param name="tls-verify-in-subjects" value=""/>
<param name="tls-version" value="sslv23"/>
<param name="odbc-dsn" value="freeswitch:user:password"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="inbound-reg-force-matching-username" value="true"/>
<param name="auth-all-packets" value="false"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<param name="challenge-realm" value="auto_from"/>
<param name="ext-rtp-ip" value="<PUBLIC_IP>"/>
<param name="ext-sip-ip" value="<PUBLIC_IP>"/>
<param name="presence-hosts" value="_DISABLED_"/>
<param name="NDLB-received-in-nat-reg-contact" value="true"/>
<param name="NDLB-broken-auth-hash" value="true"/>
<param name="dbname" value="share_presence"/>
<param name="send-presence-on-register" value="true"/>
<param name="manage-shared-appearance" value="true"/>
<param name="registration-thread-frequency" value="30"/>
<param name="enable-timer" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="send-message-query-on-register" value="true"/>
<param name="all-reg-options-ping" value="true"/>
<param name="sip-force-expires" value="3600"/>
<param name="sip-expires-max-deviation" value="300"/>
<param name="multiple-registrations" value="contact"/>
</settings>
</profile>
Any suggestions on how to make FS use the INVITE IP for RTP instead of using the IP on the SDP?
Looking forward to hear from you.
Best Regards,
Nuno Miguel Reis | Unified Communication Systems
