[Asterisk-Users] P2P RTP without SIP re-invites

[Brancaleoni Matteo]

hi
> 
> I guess this would work if both Alice and Bob were NAT'ed on the inside of the same
> NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes
> and they're on separate NAT'ed networks, the call is broken. So it's a dangerous
> configuration.

nope. I have a public * server (beta server for a free VoIP service),
on a public IP. and some sip phones around , like one in my home,
behind nat, one in my office (another nat) and some others
at my coworkers home... all behind nat. and are different nat
box, do you agree? that works ok, I have RTP passing
directly from one endpoint to the other... no RTP
on the public * server.
No stun is used. The phones are budgetones in this case.
All are configured with nat=yes on asterisk side.
or I missing something?
-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl