[*] SIP and nat

[Sean Walberg]

On 12/9/06, Bill Reid <billreid at shaw.ca> wrote:
>
> Am I correct in understanding that this is a call between two local phones
> behind the Asterisk server? If that is the case then I suspect what you
> are
> saying is correct. Things seem to work the best to always have Asterisk in
> the
> RTP stream.
>

No, this is an internal phone calling the echo test over at
fwd.pulver.comvia an Asterisk server that's also behind the NAT GW.

nat=yes seems to mean "If I want to set up audio between that peer and
myself, do I need NAT?".  It doesn't answer "If that peer wants to talk to
someone else, will he need my help to NAT?"

Sean

-- 
Sean Walberg <sean at ertw.com>    http://ertw.com/
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