Client NAT in CSS11500

Hi,
I'm currently redesigning my portal infrastructure and as such, I need to redo a few things on my CSS. But I can't seem to find anything about Client-NAT on the CSS. I need to NAT/PAT external sessions to an inside RFC1918-address. The config below simply passes the original src.ip through and consequently the session is blocked by the firewall. So far I've been unsuccessfull in finding the proper documentation, so I'm hoping someone here can help me get started.
owner BK019TF
content BK019TF-SSL
vip address a.b.c.d
add service SSLMODULE1
port 443
protocol tcp
active
content BK019TF
redundant-index 142
advanced-balance sticky-srcip
port 4433
protocol tcp
vip address a.b.c.d
add service Misys-ben-3
add service Misys-ben-4
active
Thanks
/Ulrih

Source groups are used on CSS to NAT the source ip addresses. There could be two scenarios
1. A connection is open to the server.
In this case you need 'add destination service'.
2.The server opens the connection. In this case you need 'add service ...'
Following example will give you some idea of how to implement src NAt with CSS
http://www.cisco.com/en/US/products/hw/contnetw/ps789/products_configuration_example09186a0080093dff.shtml
Thanks
Syed Iftekhar Ahmed

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    Chris Higgins

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    Message was edited by:
    Malcolm_F

    Hi lagar,
    Good stuff, I've made some additional progress as well. One thing that I discovered is that the mappings that you establish with the initial handshake are only kept by the client's NAT for a short period of time. In other words, it was working inconsistently for me because I was setting up the mappings early and then not sending the audio until a few minutes later; by that time some of the clients NAT's had lost the mappings that had been previously established and thus some of the clients weren't receiving the audio.
    I modified my application so that right before it starts sending audio it shakes hands with all of the clients, establishing fresh mappings right before the audio is sent. This seems to be working consistently now, I just tested it with someone on a different network who is behind a symettric NAT (the ugly kind) and they were able to send and receive live audio consistently, we had a little online conference call. I tested it from several other remote locations and it has worked fine so far. I have just run into one issue while testing it with someone at a remote location but there might be an issue with their JMF installation.
    Please keep me updated on the status of your project and let me know if you have any questions or ideas.

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