Our system is: SIPClients <-> Kamailio <->Asterisk<->external access points SIP clients are connected to Kamailio. All calls (from clients) are forwarded to Asterisk, and Asterisk either send call back to Kamailio, or send call to external world through paid SIP trunk. One of Asterisk servers is registered with VoIP provider. Calls to access points are going directly to Asterisk server, although it is behind Kamailio - Asterisk servers have only private IP address. Some of calls are forwarded to Kamailio and clients connected to it. Kamailio itself, as I understand is not prepared as gateway, but as sip router with some other functions. We use also rtpproxy module, and we do not need STUN for clients. PS I can send you some pieces of code, or even whole conf file. It is far far away from perfectness because I am working from very short time with Kamailio, until last 5 months we used only Asterisk.

On Tue, Apr 3, 2012 at 4:00 PM, Rob Watkin robwatkin@gmail.com wrote:

Hi Stoyan,

Does that mean that you use Asterisk as pure SIP PSTN Gateways? I imagined Asterisk as a physical PSTN gateway but thought that Kamailio/RTPProxy would scale out better for pure SIP. I was planning to use Asterisk or FreeSWITCH as a media server for hold, VM, conference and IVR.

Rob

On 3 April 2012 13:23, Stoyan Mihaylov stoyan.v.mihaylov@gmail.comwrote:

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We do just that way - Kamailo to handle load balancing and clients, and Asterisk servers for routing, gateway etc.... You can forward calls to your main gateway which then will work with other gateways or calls from gateways to clients registered in Kamailio.

On Tue, Apr 3, 2012 at 3:15 PM, Rob Watkin robwatkin@gmail.com wrote:

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Hi,

I am just getting started with Kamailio and have been following the book "Building Telephony Systems with OpenSER" by Flavio E. Goncalves. The book describes an architecture with a SIP Proxy handling registrations and handing calls to a PSTN Gateway. I now have a basic test network running where calls are routed via the SIP Proxy (Kamailio) to a third party PSTN Gateway. I feel that a better design would be to implement my own PSTN Gateway using Kamailio. This single gateway would then handle all third party PSTN gateways. Thus one Kamailio server would be facing my clients while another would be facing my suppliers.

Is this a sensible architecture and are there any sample configurations for Kamailio performing this role?

Thanks Rob Watkin

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Your remark about digest authentication explains why I have had difficulties. I was considering merging SIP interconnects with digest authenticated SIP trunks in a single gateway. I would prefer not to use a B2BUA as a PSTN gateway in order to avoid carrier media where possible. I suspect it might be better to separate out any digest authenticated SIP trunks to a B2BUA just for this.

My main reason for wishing to run a back-end gateway is to simplify our own network and to be able to reconcile against third-party CDRs all in one place.

Can you point me to any documentation for running Kamailio as a PSTN gateway?

Thanks Rob

On 3 April 2012 14:00, Klaus Darilion klaus.mailinglists@pernau.at wrote:

<snip>

It depends on what features you need facing the gateway providers. Eg. if the gateway provider uses digest authentication then you may run into problems with Kamailio. Many people use a B2BUA as "fake" gateway to have full control over the call (e.g. Asterisk, sems, sippy)

regards Klaus

Ah yes, I think you're correct that a B2BUA fits my needs. I'll look at sippy too!

Thanks Rob

On 3 April 2012 15:18, Stefan Sayer stefan.sayer@googlemail.com wrote:

o Rob Watkin on 04/03/2012 03:20 PM:

Your remark about digest authentication explains why I have had

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difficulties. I was considering merging SIP interconnects with digest authenticated SIP trunks in a single gateway. I would prefer not to use a B2BUA as a PSTN gateway in order to avoid carrier media where possible. I suspect it might be better to separate out any digest

a B2BUA doesn't necessarily need to be in the media path.

if SEMS SBC may fit your needs: you do not set enable_rtprelay in the SBC profile, you can set SEMS SBC to be signaling-only. you can enable SIP auth in the profile. you can use syslog CDR to write CDRs to syslog and then post-process from there; alternatively you can program your own CDR generation on top of call control interface (or, for 1.4, prepaid interface), adapt the rest module and use some web app server or similar. additionally you get stuff like session timers, codec filter, header manipulation etc.

Stefan