[Users] openser setup as a telephony gateway (to PSTN) for NAT'd clients

[Ovidiu Sas]

Hi Taylor,

>From what you describe here, this should be an easy setup:
 - configure all your SIP clients to talk to openser (on the private interface)
 - in the openser config, as soon as you get an INVITE, route the
INVITE to the appropriate PSTN GW (based on your rules).  You can
hardcode them or use lcr.
 - use rtpproxy to bridge the media


Hope this helps,
Ovidiu Sas

On 5/14/07, Taylor Carpenter <taylor at codecafe.com> wrote:
> I may be misunderstanding things (very probably so), but all the
> examples for both SER and OpenSER that I have seen either do not do
> NAT or NAT but are not the "end point" for a UAC to the PSTN.  From
> what I can tell they are just sending on the SIP request to the next
> destination as is.  The setup I am trying to accomplish is
>
>     * SIP clients are all on a private network with connectivity to
> OpenSER directly on one interface (on the same private network)
>     * OpenSER's other interface is on the external network (internet
> facing)
>     * SIP clients are only sending telephone numbers
> (sip:telephone_number@*... do not know about PSTN providers)
>     * OpenSER connects to the PSTN provider and send the number to
> dial that came from the SIP client
>     * OpenSER "proxies" the entire call (with the help of rtpproxy or
> mediaproxy for RTP of course)
>     * No incoming calls from the internet to OpenSER (no support for
> that is needed)
>     * Registration not required for sip clients (they are all on same
> private network and authorized)
>
> I have found several posts, example configs, documents that have
> pieces of what I need (from what I can tell).. and I have tried to
> put it together, but it does not quite work...
>
> So is there some example that fits this type of usage?  If not one
> then possibly several pieces from a few documents?  I have been
> thinking that OpenSER setup as a outbound proxy configured for
> multihome, but everything I have seen on that just routes the calls
> through to where ever the SIP client was requesting as a final
> destination and I need to send to one destination (the PSTN
> provider).  Any help is greatly appreciated.
>
> BTW, I was thinking the NAThelper or outbound proxy example from
>
>         http://www.voip-info.org/wiki-SER+tips+and+tricks
>
> Looked close to what is needed, but have a bunch of stuff (seemingly)
> unneeded for my scenario and have nothing about PSTN connectivity.
>
> Thanks for taking the time to read this long post (if you made it
> this far).
>
> Taylor
>
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