Thank you for your reply Daniel.

OK, let me try to explain better with a diagram.

I want to pass the registration request to SP1.com or SP2.com depending if its a *@sp1.com or *@sp2.com user. If the registration was successful at the service provider, the user are allowed to make phone calls. Remember, each user has their own account which they get billed for at their chosen service provider. So sipclient2@sp2.com cannot make a call on sipclient1@sp1.com's account.

I want the proxy to know the registered users on the network. If a users calls 0214610001 and there are a registered user with the number 0214610001 the call must be routed not to the service provider but directly to the other user.

Hope this makes more sense now.

On 20 January 2014 16:19, Daniel-Constantin Mierla <miconda@gmail.com> wrote:

Hello,

not sure I really understood what you want to achieve, but authentication by kamailio is done only if you call route(AUTH) for requests (in case you based your config on default one) or, in other words, the auth/auth_db functions.

But then, be aware of impacts in security. Be sure the authentication is done by someone, being you or being the provider. For registrations, if they are handled by kamailio, you have to keep doing authentication. So, just use conditions like:

if(is_method("REGISTER")) {
route(AUTH);
}

For rtp, if the clients are on the same network, then don't engage rtpproxy, the audio should work. But if they are behind routers in the same network, you may still need to do rtp relaying.

Cheers,
Daniel

On 18/01/14 19:10, Carel Burger wrote:
Hi there,

I have never used Kamailio before but want to investigate if if will work in my scenario before I invest time to learn it.

I am the administrator of a small Wireless ISP. We do not provide SIP channels to our customers, we rather let them choose a service provider of their choice. Currently our customers are using two different providers.

Since we are using ADSL with only one static IP we sometimes run into issues at the providers side with one way audio when our clients make a call to another client which is using the same service provider. I assume this is because of NAT. Since the RTP traffic actually leaves our network and then comes back to the other client the quality of the call is not as good as phoning a non client from the network.

I want to know if it would be possible to setup Kamailio to keep the internal network calls traffic from leaving the network, and also allow free phone calls on our network. To do this we would require the authentication to take place on the service providers sip server and not on Kamailio. Kamailio would only do routing of the traffic and not worry about authentication.

Would this be possible?

Regards,

Carel Burger
_______________________________________________
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
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
_______________________________________________
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