I have the following scenario: SIP Phone (RTP/SIP UDP) ---> Freeswitch ---> SRTP/TLS --> ITSP Currently when I place a call Freeswitch is not sending SRTP SDES in SDP. It does the TLS connection but SDP contains only AVP codecs. I have used the rtp_secure_media set to true and also export but crypto caps are not attached. I can see them listed, but nothing happens. Logs: http://pastebin.com/DgZ1Lejp sofia_status external::itsp-tls gateway sip:gonzalo@tls.itsp.com:5061:5061 NOREG dialplan/public.xml <extension name="public_itsp_tls"> <condition field="destination_number" expression="^(80\d{11})$"> <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> <action application="log" data="INFO Processing call to itsp"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> dialplan/default.xml <extension name="US_ITSP_Secure"> <action application="log" data="INFO Processing REGEX"/> <condition field="destination_number" expression="^80(\d{11})$"> <action application="set" data="dialed_number=+$1"/> <action application="log" data="INFO Processing US call to ITSP Secure ${dialed_number}"/> <action application="bridge" data="sofia/gateway/itsp-tls/${dialed_number}"/> </condition> </extension> sip_profiles/external/sipsitspTrunk.xml <include> <gateway name="itsp-tls"> <param name="tls-version" value="tlsv1"/> <param name="register-transport" value="tls"/> <param name="contact-params" value="transport=tls"/> <param name="realm" value="itsp-tls"/> <param name="proxy" value="tls.itsp.com:5061"/> <!-- (your configured Trunk SIP URI) --> <param name="register" value="false"/> </gateway> </include>