20 Jan
2014
20 Jan
'14
9:46 p.m.
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 *(a)sp1.com or *(a)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(a)sp2.com cannot make a call on
sipclient1(a)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(a)gmail.com>wrote;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
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