[Freeswitch-users] SRTP and bridge woes

[Daniel Ivanov]

I am building an ostn node. I followed a recipe, but got caught up with
learning FS basics. I had this trouble that calls were being signalled, but
no audio was actually put through.
I thought it was nat problems, but connectivity was going fine.
This is my dialplan snippet:
 <extension name="Local_Extension">
      <condition field="destination_number" expression="^(\d{3})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
       <!-- <bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s]
<app> -->
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
<!--        <action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<!--       <action application="answer"/>  -->
        <action application="info"/>
         <action application="answer"/>
        <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
    <!--    <action application="answer"/> -->
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
    </extension>

This works, but you can see i have an answer app right before the bridge.
If i don't answer the call, i don't get to the point where i get Activating
SRTP RECV/SRTP SEND. Yet both channels get answered and RTP goes both ways,
but i have a feeling it's not SRTP and thus is not being decrypted.
I have this in the global context:
 <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_80)$" 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>

 which should activate SRTP on both legs.

I have pasted in to the bin a debug log with and without the answer and
media sending here :
http://pastebin.freeswitch.org/19937
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120925/c8298c08/attachment-0001.html