[Freeswitch-users] RTP to SRTP Freeswitch 1.4

[Gonzalo Gasca Meza]

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 at 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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