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 -- James Cloos <[email protected]> OpenPGP: 0x997A9F17ED7DAEA6 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
