add a fixed jitterbuffer to rtp (settable from a channel variable) (libSTFU)

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5090 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/default_context.xml

@@ -53,6 +53,7 @@
       <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
       <action application="set" data="ringback=${us-ring}"/>
       <!--<action application="set" data="ringback=/home/ring.wav"/>-->
+      <!--<action application="set" data="jitterbuffer_msec=400"/>-->
       <action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
     </condition>
   </extension>