Replacing PBXes with Open Source

[Kenneth E. Lussier]

On Wed, 2004-08-25 at 16:43, Ken D'Ambrosio wrote:
> klussier said:
>  > I agree that VoIP will be huge in emerging economies...
> 
> Not sure... after all, you need the Internet infrastructure -- with a 
> fair bit of bandwidth -- in place to take advantage of it.  

This is sort of true. VoIP can leverage existing infrastructure. If the
end user is using a regular phone line that goes into a class V switch,
the switch can then translate the call to IP, route the call to the
class V switch at the other end over an IP network, which will translate
it back and send it to the analog phone at the other end. If you want to
go VoIP end to end, you need a lot more infrastructure. But if the area
has old, antiquated networks and equipment, it would still be cheaper
and faster to build a VoIP infrastructure in parallel then cut over then
it would be to upgrade/replace their old networks and equipment.
 
> I believe 
> calls are ~80kb/s, including TCP overhead, which is a fair bit more than 
> analog can cope with.  

The bandwidth needed for voice calls depend on the compression that you
use and the signaling type. G.723 can go as low as 5.6kb/s, G.729 is
about 8kb/s, and G.711 is about 64kb/s (which is what standard analog
phones use). 

> On the other hand, my pesky Asterisk hardware 
> STILL hasn't shown up a month after I ordered it, so I can't speak from 
> experience.  (No fault of Digium; I had to go through a third-party 
> vendor for purchasing reasons.)
> 
> One question I have, though: how does H.323 cope with NAT, firewalls, 
> etc., for incoming calls?  Anyone know?

Well, if you choose to use H.323, you will most likely need a
gatekeeper, etc. However, I would recommend using SIP instead. It's a
faster and more flexible protocol. How it handles incoming calls depends
on how they are coming in. Are you planning on having calls come in over
IP via an IP Telephony provider, or are the calls coming in over PSTN
lines? If they are coming in via the `net, then you would probably need
to allow traffic from the outside to port 5060 and do the IAX
registration that way. Check out http://www.voip-info.org/wiki-Asterisk
for all of your asterisk needs. They have a lot of info for connecting
to various providers, as well as everything you needed to know about
Asterisk (but were afraid to ask on the users mailing list ;-)

C-Ya,
Kenny