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

[Russell Bryant]

On May 26, 2008, at 7:26 AM, Lukas wrote:
> 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?
>
I can not think of any reason that this should happen by _only_  
enabling TLS for SIP.  There are many other things that will make  
Asterisk not send a re-INVITE to the phones, but changing the  
transport is not one of them.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.