asterisk users - Jun 2004 - X-Lite/Firefly behind NAT connecting to Asterisk not receiving RTP

I have an asterisk server up and running, using Firefly in IAX mode 
works great, even with Firefly behind a NAT (as expected, since IAX 
works really well with NAT).

Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from 
behind the NAT, and I can't seem to get there.

At this point, the phone will successfully register with Asterisk, and 
the Asterisk qualify messages get delivered to the phone. I can initiate 
a call with the phone, or Asterisk can send a call to the phone. In 
either case, the RTP traffic does not get accepted by the NAT in front 
of the phone, because X-Lite (and Firefly) are not sending the _first_ 
RTP packet to get the conversation started.

In X-Lite I have specified the Asterisk server as an "outbound proxy",
and its logs shows that setting has taken effect. In Firefly there is no 
means to make a similar setting. STUN does not make any difference, 
because Asterisk is already sending the RTP packets to the correct IP 
address, the firewall is just not allowing them through because it 
doesn't know the client (phone) is expecting them.

Obviously I can reconfigure the firewall to open up ports for RTP, and 
port-forward them to the computer that X-Lite/Firefly is running on, but 
that's not going to work with two phones behind the NAT :-(

I just don't understand what I'm missing here... aren't SIP clients
that
know they are behind NAT supposed to initiate the RTP conversation 
themselves, so their NAT firewall will know to accept the RTP packets 
coming from Asterisk?

Kevin P. Fleming wrote:
> I have an asterisk server up and running, using Firefly in IAX mode
> works great, even with Firefly behind a NAT (as expected, since IAX
> works really well with NAT).
I have the same scenario, but after about 4 hours, the Firefly phones can
still make calls, but asterisk won't send them calls, and the caller gets
the unavailable message.  In the asterisk CLI, I still see the Firefly phone
registering about every 30 seconds, so * knows it's there.  Restarting
Firefly gets it back up and running again for about 4 more hours.

> Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from
> behind the NAT, and I can't seem to get there.
>

Are there any VOIP providers that offer Local Number Portability in Houston?

Cheers,
Rich

Rich,
My organization most likely can assist you with your need. Please
contact me off list at brett@utex.net.

Thanks,
Brett


-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Dr. Rich
Murphey
Sent: Wednesday, June 16, 2004 3:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] LNP local number portability in Houston (713,
281, 832)


Are there any VOIP providers that offer Local Number Portability in
Houston?

Cheers,
Rich


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