Hi Taylor,

see inline:

On 5/14/07, Taylor Carpenter taylor@codecafe.com wrote:

Thanks for your reply Ovidiu.

Is there any need to change any thing with the SIP request because of NAT? Do I need to use rewritehost? Before the NAT requirements I was doing ds_select_domain() for loadbalancing... At some point I will want to do that again, but for the moment I am just wanting private lan traffic to be able to place outbound calls with the PSTN provider.

Yes, you can use rewritehost() to route your request. lcr would work better here.

On that note some (though not all) of the providers require auth and I would like OpenSER to provide the credentials so that the SIP clients only have to know about OpenSER to dial out... I have been using the uac module for that.

You can use the uac for authentication, but there are some restrictions (your provider must accept authenticated INVITEs with the same CSeq number).

So is there anything special (besides RTP traffic) to do with the rest of the SIP packets before doing a t_relay()?

The examples I have seen using rtpproxy used FAEE, FAEI, FAIE, FAII, etc.. I assume that all of those are not needed if everything is coming from the private side out to the internet (and PSTN provider). Is that correct?

You will need to proxy the RTP traffic between the two interfaces.

Taylor

On Mon, May 14, 2007 at 04:06:50PM -0400, Ovidiu Sas wrote:

...

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@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

Openser 1.2 question:

Having a multi-valued AVP $avp(s:uri), does

$ru = $avp(s:uri);

achieve the same as

avp_pushto("$ru", "$avp(s:uri)/g"); ?

Or is there something like

$ru = $avp(s:uri)/g; ?

thanks, Christian