[Asterisk-Users] Project Critique

[Andrew Joakimsen]

If setting up multiple users in the same location overseas, I would
strongly suggest setting up another asterisk box at that location. IAX
can transverse nat and you will have the change to use some low
bandwidth codecs that will not cost you anything more, as I have found
that most devices do not support much more than G711 (too much
bandwidth) or G729 ($10 per channel license) or G723 (no official
support, I have tested it with asterisk but for anything other than
testing you want to have the software licensed, starts at $30,000)



> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Philipp von Klitzing
> Sent: Saturday, December 06, 2003 7:54 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Project Critique
> 
> Hi!
> 
> > using NetMeeting, or GnomeMeeting, or some other H323, or SIP
> > compatible client
> 
> For H323 you'll need to install a gatekeeper next to Asterisk and
fiddle
> with h323 or oh323 (I love to live dangerously, hit me Jeremy). :->
> Moreover NetMeeting doesn't work through NAT.
> 
> SIP is easier to set up, no extra gatekeeper required.
> 
> For permanent use you better get hardware IP phones that can ring,
> have better audio quality, and work while the computer is switched
off.
> However, consider if either GSM (software & SNOM hardware phone) or
g729
> / g723.1 codecs (found in most hardware phones, need to buy Digium
> license for g729) are the path for you to go, or if you can live with
the
> bandwidth monster g711.
> 
> > Another question is:  If the workers are in an office-space, and in
> > order to prevent the need for an additional asterisk PBX in India,
is
> > it easy to run each of the workers' connections on separate ports?
> > (ie. port 5060 for phone # 555-1212, port 5061 for 555-1213, and
port
> > 5062 for 555-1214 ?  This way each NetMeeting, or GnomeMeeting
> > connection coming from India can simply run behind a NAT router,
> > instead of setting up a separate Asterisk PBX.
> 
> Yes, but you'll also need to take care of the RTP (upd) traffic
> forwarding, not just the negotiation protocol. So go for SIP, or
consider
> using IAX (only if you want to stick to software clients, though)
unless
> you require video support for your clients.
> 
> Cheers, Philipp
> 
> 
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