The real question is:
*Can I have no RTP bandwidth consumed by the Asterisk server? (SIP
data allowed) Supposedly the 2 VoIP phones can talk to each other
directly through the NAT once STUN and SIP do their ***magic*** to
establish their RTP connection.*
If the NATs are NOT Symmetric then YES. (Google for the different types
of NAT if you don't know this by now)
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