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>

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