|
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 |
_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
