[j-nsp] ScreenOS and VoIP and NAT

[Ivan c]

thanks Tony, I have read those examples. In the eg. the phone on one
side is a SIP client/server and on the other side there is a SIP proxy
and handset.

My scenario is different as I have a alcatel omni box which talks SIP
to the remote cisco call manager, and then hands off the RTP stream to
the handset which talks direct.

When I iniatiate a call to the remote end the recipes work from the
cookbook, but when the remote end iniates a call its a no go. I can
see it in the SIP trace, the netscreen sees the SIP, but it does not
know what to NAT the incoming stream to...

I think the issue is that I am trying to use the netscreen as a SBC or
proxy type device which obviously it isnt designed for.

On Tue, Nov 24, 2009 at 9:51 AM, Tony Frank <tony.frank at ericsson.com> wrote:
> Hi Ivan,
>
>> it is all direct, the alcatel omni handles the SIP, and then hands off to the phones, which talk direct RTP....
>> But I still can't understand how the firewall would know how to NAT the incoming traffic, first to the SIP server and then to each handset....
>
> Have you read through the description for SIP with NAT, incoming calls covered on page 26?
>
> http://www.juniper.net/techpubs/software/screenos/screenos6.1.0/ce_v6.pdf
>
> The examples from page 33 onwards do seem to describe your scenario, unless I am missing something obvious?
>
> Regards,
> Tony