[Asterisk-Users] Medium volume 100% SIP/IAX PBX.

[Kristian Kielhofner]

Chris Shaw wrote:

> * should have no problem keeping up with a setup like that, and VoIP is
> certainly capable.... However... What most interests you? Is it cost savings
> or audio quality. If it's cost savings, you could push 25 calls through a T1
> using GSM encoding, but it would not sound quite the same as a regular line.
> If you use G.711 (mu/A-Law) then you would get toll quality audio but only
> be able to push about 17 calls through at once...

	In my experience G.729a has better quality than GSM (higher latency, 
though).  Plus it is supported on the Cisco 7960G so I can do 
passthrough from 7960G to 7960G and since NuFone also supports G729a, I 
can do passthrough there as well.  Cut down on those pesky licenses and 
* CPU usage.

> Also you must remember that the current RTP implementation in Asterisk is...
> somewhat... lacking, and with a 100% VoIP setup you will need a timing
> source like ztdummy (which requires a UHCI USB controller) or ZapRTC. Or if
> you're using linux 2.6, I don't think you need anything as the internal
> timer resolution is precise enough...

	I would probably have a Wildcard TDM400 or something for POTS backup 
and basic analog anyways.

> Our company is thinking of deploying a setup like this but a bit smaller,
> only 12 extensions and at most 8-9 simultaneous calls. I would certainly
> recommend a setup like this, it's a huge cost savings. I would also do
> plenty of homework and figure out how to do it before actually committing to
> it. Maybe even do a parallel setup where you have some POTS lines as a
> backup. I would also use some kind of failover where your IAX provider can
> forward your incoming calls to another IAX provider's number or a POTS
> number during downtimes...
>