[Freeswitch-users] RTP NAT problem in Freeswitch 1.2.3

[Nuno Reis]

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*

M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Soluções Rádio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

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