[pjsip] Can PJSIP register with hundreds of VoIP providers at one time?

[Klaus Darilion]

Eric Chamberlain schrieb:
> 
> 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.

I would just ask the provider if he allows third-pary registration, or 
just test it (e.g. using sipsak) - of course providers can change the 
setup and then it wont work anymore.

You could also try Joegen's approach, routing all the traffic via a SIP 
proxy, which acts like some kind of egress/ingress proxy (some call it 
SBC), e.g:

Asterisk--
           \
            -------Kamailio -------- Provider
            -------SIP Proxy
           /
pjsip----
reg-clien

As registration-contact you provide the IP:port of the Kamailio proxy. 
Then, the provider will route all requests back to Kamailio and then you 
can forward the request to Asterisk.

regards
klaus