[Kamailio-Users] Scaling up asterisk using kamailio while keeping features (WAS: re-invite mid call?)

[Antonio Goméz Soto]

Op 02-01-10 19:48, Olle E. Johansson schreef:
>
> 2 jan 2010 kl. 16.31 skrev Antonio Goméz Soto:
>
>> Hi all,
>>

[snip]

> Well, to scale you DO need a proxy. The issue here is what you use the proxy for and how you distribute
 > the telephony services you need. You always have to start with the services you use. If Asterisk is a simple PSTN gateway for SIP,
 > then it is very easy, since there's no state involved you are interested in sharing. If you are using Asterisk for PBX services
 > and you do need blinking lamps, there is a state you are interested in.
>
> Now, with a decent server, you can manage thousands of simulatenous calls on one Asterisk server. You can use OpenSER for failover
 > between two Asterisk servers, but only use one in production in order to preserve states on one machine. That's is one way of
 > making it easy for yourself.
>
> If you really need to distribute states, there are now multiple ways to do that between multiple Asterisk servers, with custom
 > device states as one example.
>
> So the people who pointed you to OpenSER/kamailio was not wrong at all. If they said that it was easy to scale a PBX using
 > OpenSER, they might have been wrong. It's not easy yet, but many of us are working on making it easier.
>
> /O

Olle,

this sounds like decent advice. Looking at the asterisk source, I think I could safely say that with
all lists converted to astobj2 (in 1.8) that asterisk can handle tens of thousands of phone registrations,
and thousands of channels. I think asterisk is making great strides in scalability at the moment,
don't you agree? It might even scale up to OpenSER/Kaimilio numbers.

It may not be able to handle the media for all those calls, but that is an entirely different matter.

OTOH, it might just be wishful thinking..

Thanks,
Antonio