[Asterisk-Users] Re: sip URL peering

[Wolfgang S. Rupprecht]

Klaus Darilion <klaus.mailinglists at pernau.at> writes:
> It's not that easy. If you want to have open SIP URIs (just like email
> is open for everybody) you will receive SPIT calls. E.g. the SPEER
> group tries to define rules for VoIP peering which allows
> authentication to enable open SIP URIs. (I won't open acces to my SIP
> URI if I can not verify the senders URI).

Keeping spam in mind seems like a really good idea.  I'm also a big
fan of keeping a cryptographic "paper trail" so that one can figure
out who spammed.

On the other hand, is SPAM / SPIT a big enough problem at this point
to warrant scuttling any interconnectivity?  It seems a bit premature
to worry about a problem that may not develop for 5 years and allow
that fear to stop direct sip dialing.

As an amusing aside, I inadvertently added a "captcha" to my phone
line when I had the local number go into an IVR that asks the caller
to press 1 for person XXX and 2 for person YYY and 3 of they are a
telemarketer.  I don't think anyone other than my friends has ever
pressed 3, but the predictive dialers used by the phone-spammers
doesn't seem to pass the turing test and isn't able to press 1 or 2.
;-) I see lots of timeout-hangups in the IVR with caller-id's like
"CAR PROMO" or "VOIP CALL".

If spam/spit is ever a problem, I'm simply routing previously unseen
calls to a turing test of the same type and anyone that has previously
called (and/or been called) gets to bypass the turing test.

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html