SRTP trunk svn.digium.com/svn/asterisk/team/jpeeler/srtp

I tested this trunk with Fedora C7. Wireshark reports SRTP stream, but there is not AUDIO. I see if one endpoint is set as rtp only, asterisk send AUDIO to this endpoint but nothing to SRTP direction.
My question is: What is state of art of SRTP ?
Is there a trunk really working ?
I can work to this trunk but i need help from other developers, because I’m new with asterisk source code.
thanks everybody.