14 May
2007
14 May
'07
11:06 p.m.
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(a)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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