I would recommend updating to a later version... There are many known problems in 1.2.3 up to an including a remote unauthenticate triggerable crash that¹s at least theoretically exploitable... As far as NAT FS should handle that... You might need to use one of the NDLB flags.... On 7/12/13 8:34 AM, "Nuno Reis" <nreis@wavecom.pt> wrote:
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@wavecom.pt <mailto:nreis@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=202333747613191340808.0004b4b227a6... 44f0df88> | www.wavecom.pt <http://www.wavecom.pt/> <http://www.wavecom.pt/>
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