[asterisk-dev] SIP TLS: traffic flow phone-server-phone

[Lukas]

  Dear Asterisk developers,     I am currently writing my bachelor thesis in
computer science. It's all about asterisk with a special focus on
security (SIP over TLS & SRTP). So far so good, but now I have a question
regarding SIP encryption in Asterisk. I noticed that voice traffic (RTP)
flows from the caller's phone directly to the callee's phone without
passing through Asterisk server. Now that sounds pretty logic to me.  But
the moment I activate SIP over TLS the route of the voice traffic changes so
that it now all flows from the caller's phone to the server and from the
server further on to the callee's phone. Why did this route change? Are
there some legal reasons for that?  Thanks a lot for your efforts.    
Further information regarding my environment:     server:  --------------- 
Asterisk 1.6.0-beta (srtp version from jpeeler:
http://svn.digium.com/view/asterisk/team/jpeeler/srtp/
(http://svn.digium.com/view/asterisk/team/jpeeler/srtp/))  IP: 192.168.0.1 
   Phones:  ---------------  2x snom 300, firmware version 7.1.30  outbound
proxy: 192.168.0.1:5061;transport=tls  IP: 192.168.0.42  IP: 192.168.0.43   
    Best regards,  Lukas  FHNW University of Applied Sciences, Switzerland  

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