[asterisk-users] srtp/dtls when sip is clear over lo

[Joshua Colp]

James Cloos wrote:
> Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or
> chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly,
> will ast negotiate srtp or dtls even ast and the proxy speak sip in
> the clear over the lo interface?
>
> Avoiding encryption over lo can aid debugging, but will doing so also
> block secure media?

The media is not carried over the SIP signaling, it is negotiated using 
SDP and flows over different ports. Unless you also do media 
manipulation in the SIP proxy then it won't touch that.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org