Hi All,
I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.

I have followed the steps mentioned in WIKI (https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.

Also, I am bit confused with following condition in "default.xml" when wiki page suggests that
late coded negotiation must NOT to be turned on.

    <!--
         Since we have inbound-late-negotation on by default now the
         above behavior isn't the same so you have to do one extra step.
        -->

      <condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/>
      <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="sip_secure_media=true"/>
      </condition>

By commenting/un-commenting "sip_secure_media=true" above, it did not worked.

Any help with this would be greatly appreciated.

--

Thanks

CWM