Hi FreeSWITCHers: I configured a sip profile as bellow: ================================================================================================= Name sipinterface_1 Domain Name N/A Auto-NAT false DBName sofia_reg_sipinterface_1 Pres Hosts Dialplan XML Context context_1 Challenge Realm auto_to RTP-IP 192.168.10.59 Ext-RTP-IP X.Y.Z.W SIP-IP 192.168.10.59 Ext-SIP-IP X.Y.Z.W URL sip:mod_sofia@X.Y.Z.W:5060 BIND-URL sip:mod_sofia@X.Y.Z.W :5060;maddr=192.168.10.59;transport=udp,tcp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT true CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 for every INVITE that comes in via this profile, FS replys with this SDP: v=0 o=FreeSWITCH 1407905843 1407905844 IN IP4 192.168.10.59 s=FreeSWITCH c=IN IP4 192.168.10.59 t=0 0 m=audio 25142 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Although the Contact header of 200 OK is correct : Contact: <sip:4000@X.Y.Z.W:5060;transport=udp> Appreciate all comments, -- Afshin