[asterisk-users] SIP/RTP Nat problem, can't solute it.

[Rudolf Ladyzhenskii]

NAT changes address of the packet, but does not go inside of the SIP
packet itself. And SIP packet contains address as well. If you look at
debug output, you will see that SIP packets have remote host local
address in them, not the public IP as one would expect. At least this
is the problem I have.
Basically one needs some software to "NAT" the addresses inside of SIP
packets. STUN server is one alternative. I am about to put one in.

Rudolf

On 1/7/07, C F <shmaltz at gmail.com> wrote:
> Change To canreinvite=no
>
> On 1/6/07, Facundo Barrera - GMail <facubarrera at gmail.com> wrote:
> > Dear list:
> >             I have the typical one way audio problem, as far as i know
> > it's a nating problem, my hosts inside my lan can call to outside
> > internet hosts, but can't listen a thing, i read a lot about sip and
> > rtp and protocols and the problem it seems to be with NAT, this is the
> > config i put on my sip.conf file about nat:
> >
> > externhost=sip.server.com.ar > my server name on the internet
> > localnet=192.168.5.0/255.255.0.0 > my LAN
> > nat=yes
> > canreinvite=yes
> >
> > And this are the ports i opened on my firewall script
> >
> > iptables -A INPUT  -p udp -m udp --dport 8766:35000 -j ACCEPT
> > iptables -A INPUT  -p udp -m udp --dport 5004:5082 -j ACCEPT
> >
> >
> > But still can't hear a thing from an outside call, any hel will be
> > appreciate
> >
> > Thanks a lot
> >
> > --
> > _________________________
> >    Facundo Agustin Barrera
> >   --------------------------------------
> >      www.openlabs.com.ar
> > "Let the penguins do the work"
> > ---------------------------------------------
> >    Buenos Aires - Argentina
> > _________________________
> > _______________________________________________
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