[VoiceOps] Preventing random SIP connections to handsets

[Carlos Alvarez]

We're starting to see customers who get random arbitrary ringing caused by
a random connection attempt from the internet.  Most of our customers have
Cisco routers with full-cone NAT, so it's easy to do that.  We don't
reinvite handsets, we proxy the media, so we've considered using restricted
NAT instead.  If we can figure out how, we can't find any documentation on
how to do it, and don't have a response to our Cisco TAC case on it yet.

But I figured I'd ask if others have come up with better solutions.  I know
there are a few authentication options in the phones themselves, but they
seem to vary greatly by vendor and even by model.  I like to do things as
simply and system-wide as possible.  We primarily sell Grandstream, and we
support Cisco/Linksys SPA as well as Polycom IP series (not VVX).

We're an Asterisk-based hosted service provider.
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