[Asterisk-Users] VoIP Mythbusters Help!! - NATed phone-phone connection without proxy? Possible? Yes/No

[Tomas Florian]

Hello,

 

All I'm looking for is a yes/no answer here.  I have heard that the
following scenario is possible (reasonably easy to implement as well) . but
I just don't get it :-) . if it is possible I'll go ahead and learn on my
own, I just don't want to waste time on something that will not work.

 

Scenario:

 

2x VoIP phones

-          Each phone is configured to register with SIP server
139.142.111.1

-          Each phone is behind a standard NAT device (say regular home
Linksys router - with no ports manually forwarded - it's out of the box
configuration)

-          Each phone is configured to use STUN to find out it's external IP
and the type of NAT it's behind

 

1x Asterisk Server for SIP registration

     - 2 SIP peers defined with extensions 200 and 201

 

 

I already know I can make the phones call each other . NP . but the RTP data
is routed over the Asterisk consuming bandwidth on that server (in+out).

 

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.

 

So can this be done or did I pick up some myth somewhere?

Also, if it can be done, how to I block the VoIP phones from sending their
RTP over the Asterisk in case they can't negotiate a direct connection
between each other?

 

 

Thank you very much,

 

Tomas

 

 

 

 

 

 

 

 

 

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