[Asterisk-Users] Total newbie here looking to do a VoIP conference call?

[Jim Van Meggelen]

asterisk-users-bounces at lists.digium.com wrote:
> Sorry for the misspelling...  Thanks for the replies.  I will
> set it up and start playing.  This is all very exciting.
> I've been using VoIP as my primary phone but this is going a
> bit further.  At the office we have a T1 that is probably
> fairly dead after hours.  Supporting 5-10 users should be
> fine I'd imagine.  I've read 1 VoIP connection uses about 64kbps or
> 8KB/s? 

That's a tough one. The bandwidth of a VoIP connection is a combination
of the bandwidth used by the codec itself, plus the overhead. In some
cases, encapsulation can mean that each VoIP packet is as bad as 50%
overhead or more. 

Generally the overhead will be between 10Kbps and 16Kbps (although I
might be off by a few K - these are ballpark figures). The codecs range
from 5.3Kbps all the way up to 64Kbps. So a VoIP connection could use
between 15K and 80K per channel.

I believe IAX can be more efficient, as it eliminates a lot of overhead
by combining multiple channels into trunks. It is a very interesting
VoIP protocol.

> So... Asterisk can act as a SIP server...

Yeah. It's probably best decribed as a gateway, although it can be just
about anything you need it to be. For heavy, pure SIP work you'd want to
look at SER (SIP Express Router), and use Asterisk as a PSTN/non-SIP
gateway, or an application server (voicemail and such).

> My packet8 "dta310" adapter has the SIP server hardcoded into
> it. If I could change that, I could use that?

If it's SIP, it can talk to Asterisk.

> But since it can't be modified, we'd have to purchase SIP
> adapters for each employee... Something like this?
> http://store.sipphonestore.com/  I guess we need one where I can
> dynamically define the SIP server. 

If your existing phone system has a T1, you might consider putting the
Asterisk in front of it, like this:

[PSTN]---[Asterisk]---[PBX]

That'd allow you to keep your existing phones running and gradually
migrate your users to Asterisk.

The Wiki is full of fun stuff. Read, experiment, and hang out with us
here.

Cheers,

Jim.