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

[Eric Chamberlain]

On Oct 8, 2009, at 1:35 AM, Klaus Darilion wrote:

>
>
> 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.
>

Klaus & Joegen,

Thank you for the input.  I'll take a look at the sip proxy approach.

--
Eric