On Oct 7, 2009, at 8:59 AM, Klaus Darilion wrote:
>
>
> Eric Chamberlain schrieb:
>> On Oct 7, 2009, at 12:09 AM, Klaus Darilion wrote:
>>>
>>>
>>> Eric Chamberlain schrieb:
>>>> Hello,
>>>> We'd like to move registration to upstream VoIP providers off of
>>>> Asterisk and on to something that can be updated dynamically and
>>>> maintained in a database. Could the python pjsip module be used
>>>> to perform registrations on behalf of the Asterisk server
>>>> (contact info would point to the Asterisk server)? If so, can it
>>>> scale up and handle hundreds or thousands of registrations at a
>>>> time?
>>>
>>> Of course this only works if your upstream VoIP provider accepts
>>> the value in the contact header. Many providers (me too) do ignore
>>> Contact completely and use IP:port from where the request was
>>> received instead.
>>>
>> Klaus,
>> Interesting point. How does your or other providers'
>> configurations handle multiple registrations from the same
>> IP:port? Multiple users behind a firewall for example? If you
>> ignore the contact, how does the registering device know where to
>> route an inbound call?
>
> Hi Eric!
>
> I was bit too less verbose. The provided contact will be stored by
> the proxy, and during in incoming call, the RURI of the INVITE will
> be set to the provided contact. However, the INVITE will not be sent
> to the URI in the RURI, but to the IP:port from which the
> registration was performed.
>
> For example in Kamailio (Openser) this is done by storing both data
> during registration.
>
> location table (which holds the registered clients):
>
> AoR | Contact | received_from
> -------------+-------------------------+----------------
> aaa at domain | sip:192.168.0.1;line=1 | 1.2.3.4:5060
> bbb at domain | sip:192.168.0.1;line=2 | 1.2.3.4:5060
>
>
> Thus, an incoming call to aaa at domain will be sent to 1.2.3.4:5060,
> with request line containing: INVITE sip:192.168.0.1;line=1
>
> In Kamailio, this can be activated by using fix_nated_register()
> before calling save("location").
>
> If you do not sending back symmetrically, but instead using the
> provided contact for routing, you have to validate the contact
> before - for example a malicious user could send a REGISTER with
> Contact: sip:0900666666 at ip.address.ofyour.gateway
> and the proxy will forward the request to its own PSTN gateway
> instead of sending it to the user.
>
> regards
> klaus
>
Klaus,
Thanks for the clarification. It sounds like the most compatible
solution would be to run our pjsip code on the Asterisk box, so that
the port and IP addresses would match and Asterisk could answer the
inbound INVITEs.
