Voice-over-IP is the current craze. It's not the first, and it probably won't be the last, but it's here to stay. It's slowly but surely changing the way we think about telephony and Ultimate Linux Solutions is adamant to be leading the pack with new and innovative solutions in this field.
In a similar way that open-source software such as firefox, openoffice and many other Linux utilities have already and continues to change the way we think about how software should operate VoIP is changing the way people think about voice communications. With desktop solutions such as skype callout the cost of telephony calls have already started to come down significantly. But VoIP is much, much more than just free and cheap calls.
With standards such as SIP and IAX/2 different implemenentations are significantly more inter-operable than more traditional PABX systems, and by running on top of ethernet it's much, much more scalable.
From a functionality perspective the options are endless. If you can imagine it, it can probably be done - and much more cost effectively than would otherwise be possible with hard-wired equipment.
Why the claim to cheaper telephone bills?
There are many different reasons why VoIP can potentially work out cheaper. The most significant is that due to lower fixed costs on the providers side we can offer calls at a lower rate than what a telco such as Telkom is able to do.
Our capital outlay is lower - instead of having to spend millions (possibly billions) on switching equipment we can spend thousands on server hardware and software (of the shelf for most VoIP providers, custom built or hybrid - such as what ULS is doing), thus we don't need to recover this costs from our customers (we unfortunately do still have to interact with the telco's and their equipment, and this is still quite expensive).
We can run higher call concurrencies on the same bandwidth. This is primarily due to signficantly better voice compression algorithms than when ISDN (what most telco's are still using today) was designed. On an ISDN circuit you require 64Kbps of actual bandwidth per direction per call, excluding overhead. Using newer codecs we can do 8 calls on that same bandwidth (8 Kbps/call, excluding overheads). To extrapolate this, using correct trunking mechanisms on an E1 (PRI) link Telkom can run 30 concurrent calls, on that same link (Actually, the equivalent thereof) we can do slightly over 160.
You can see our call costs here.
Different requirements - different solutions
Whilst VoIP is insanely flexible (thousands of extensions on a single PABX are not unimaginable) these large systems are not for everybody. Whilst we are more than happy to consult on your needs, here are a few guidelines (note that transitioning from smaller to larger solutions is very easy and cost effective):