[asterisk-users] SJphone behind NAT/Firewall without sound

[Vincent]

On Thu, 3 Apr 2008 22:30:10 -0500, kazabe <kazabe at gmail.com> wrote:
>I need connect some LAN stations with SJphone to an Asterisk Server
>published on Internet. [...] I dont manage the asterisk server. 
> I just manage my proxy/firewall, and i need to my users can
> connect to that server.

SIP works like FTP: One channel to manage calls, and a second one for
data (audio):

http://freshmeat.net/articles/view/2079/

Since Asterisk doesn't (yet) support STUN, to get audio packets to be
received, you must configure the NAT firewall to let them in, and
route them inside to the Asterisk server.
This must match whatever is listed under /etc/asterisk/rtp.conf (you
can reduce the range from 10000-20000 to eg. 10000-10010; I could be
wrong, but I think RTP actually needs two channels per call.)

The same thing is required for the client hosts running the SJPhone
application, but from what I read, most firewalls will work without
having to map ports, and STUN-capable applications like SJPhone will
keep the UDP ports open by sending out dummy packets regularly.

If you can't modify the NAT firewall in front of the Asterisk server,
I don't see how to solve this.