[Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question

[Peter Pauly]

If Asterisk registers with a SIP long distance provider and
I make a call from an IP phone through Asterisk to that
LD provider, does the RTP (audio) traffic flow between the two
end points directly (normally the IP phone and the LD provider) or
does it flow through Asterisk?

I'm asking because I have Asterisk running behind a NAT firewall
along with an IP Phone (software) and I'm trying to get it
working with Iconnecthere (ICH). I am able to register, connect
, but no audio. I have ports opened up on the firewall, but
they point to the Asterisk machine and not the IP phone machine. 
In this scenario, any audio traffic would have to go through the
asterisk box to reach the IP phone. Is that how it works?