[c-nsp] Voice and nat

[Roger Wiklund]

I would mirror desired ports in the switch(LAN, WAN)hook up a PC and
run wireshark. Make a call from/to the wireless clients and capture
the data, in the SIP Invite scroll down to the SDP, you will see the
IP address used for the RTP stream, also you should see you will see
the flow there.

Ensure that the routing is working correctly based on that info, check
NAT translation statistics in the router etc. If you have spare public
IPs, do 1-to-1 NAT on one of the wireless clients, does that fix the
problem? If so check what ports/IPs are used etc etc.

/Roger

On Tue, Aug 17, 2010 at 5:10 PM, j s <batta at moris.org> wrote:
> I hope this is where I have to post this question at.
> I have 2 routers.  r1 is the router connected to the net via an ethernet interface, it is performing nat to exit to the world, and r2 is a router behind it  and connects wireless users.
> r1 and r2 have a switch between them and few servers and end users are connected there.
> The ip addresses of the devices on this switch is 192.168.1.0/24.  The ip  addresses of the wlan customers is from 192.168.2.0/24.
> I have an asterisk Server connected to the switch having ip address 192.168.1.111.
> Customers are able to connect to the asterisk from the outside network and getting sip calls with no problem. Customers from inside the network, coming from 192.168.1.0/24 too.
> The problem is with customers connected to the wifi network. Sip is ok, but rtp isn't.
> Btw, all the customer are natted for getting to the internet.
> There is also some traffic coming from the internet natted to the inside ip addresses of the servers.
> Any Ideas?
> Thank You all.
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