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

[James Cloos]

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?

-JimC
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James Cloos <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6