[SR-Users] Non-authentication setup of Kamailio

[Carel Burger]

Thank you for your reply Daniel.

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

[image: Inline images 1]

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 at sp2.com cannot make a call on
sipclient1 at 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 at 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 listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
> _______________________________________________
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