<div dir="ltr">Thank you for your reply Daniel.<div><br></div><div>OK, let me try to explain better with a diagram.</div><div><br></div><div><img src="cid:ii_143b1285f29bc890" alt="Inline images 1" width="420" height="315"><br>
</div><div><br></div><div>I want to pass the registration request to SP1.com or SP2.com depending if its a *@<a href="http://sp1.com">sp1.com</a> or *@<a href="http://sp2.com">sp2.com</a> 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 <a href="mailto:sipclient2@sp2.com">sipclient2@sp2.com</a> cannot make a call on <a href="mailto:sipclient1@sp1.com">sipclient1@sp1.com</a>'s account.</div>
<div><br></div><div>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.</div>
<div><br></div><div>Hope this makes more sense now.</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 20 January 2014 16:19, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Hello,<br>
<br>
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.<br>
<br>
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:<br>
<br>
if(is_method("REGISTER")) {<br>
route(AUTH);<br>
}<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 18/01/14 19:10, Carel Burger wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">Hi there,
<div><br>
</div>
<div>I have never used Kamailio before but want to investigate
if if will work in my scenario before I invest time to learn
it.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>Would this be possible?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Carel Burger</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
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<pre cols="72">--
Daniel-Constantin Mierla - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
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