[asterisk-users] Asterisk, NAT, and RTP?

[Vincent]

Hello

I think I finally understood the issue/solution, but I'd like to make
sure I'm correct:

- In Diana Cionoiu's famous article on Freshmeat
(http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol),
regardless of whether SIP end-points use a public IP or are behind a
NAT, RTP packets flow directly between the two SIP end-points because
the SIP server only acts... as an SIP server, meaning it only acts as
a registrar (for SIP end-points to make themselves know with an IP +
RTP ports), and then as a Central office (to ring the other SIP
end-point, and close the connection when an SIP end-point decides to
hangup)

- OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call
transfer, call parking, etc.), it must remain in the loop, and hence,
by default (canreinvite=no), all RTP packets always go through
Asterisk, even if both SIP end-points live in the same network as the
Asterisk server (and hence, since NAT is not involved, there's no need
for any kung-fu with rewriting information in SDP packets and asking
the NAT box to open the relevant ports for RTP)

Is this correct?

Thank you.