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.fr...@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 _______________________________________________ juniper-nsp mailing list juniper-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/juniper-nsp