[Asterisk-Users] Project Critique

[Philipp von Klitzing]

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