asterisk users - Sep 2003 - SIP Phone -> Asterisk -> SIP LD Provider question

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?

> 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?
I was using a sniffer a few minutes ago to identify an issue between
a cisco 7960 and ata186. The call setup occurs between the phones and
asterisk on udp 5060 (both source and destination ports), but the 
actual conversation was directly between the phones (in at least this
one example) on udp ports 23570 and 10000, with 180 byte data payloads
occuring approximately every 20 milliseconds.

Another call between XLite and a Snom 200 used udp ports 8000 and 10018
directly between the phones.

The above is only intended to point out the NATing issues associated with
using voip through a firewall.

Rich

Peter Pauly wrote:
> 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?
SIP control connection usualy goes thru firewall. RTP - no.
Just put the Asterisk on the machine with the firewall and it will work.