However, while understanding the basics of VoIP and SIP is important, setting one of these systems up will require some general network skills, too. For the best voice quality, your network will need to meet certain minimums levels of throughput for upstream and downstream data. In addition, you'll also need to meet a minimum latency number (that is, the time between when a signal leaves a remote computer and when your system receives it), typically measured in milliseconds. You'll also need a business-grade connection to the Internet if that's where your call traffic is going to go.
The Session Initiation Protocol (SIP) is one of the underlying technologies that make VoIP possible. This is a text-based protocol similar to HTML. It's the most commonly used standard for setting up and controlling phone calls in most VoIP systems. You'll run across references to SIP in almost anything you do with these kinds of phone systems, especially when you're selecting any handset hardware you want to use.

For a VoIP system to work, it needs a means of routing calls between users or to the outside world. In a cloud-based system, this gets handled by a virtual PBX. In the cloud, this routing is managed by your VoIP provider, which is part of why you're paying them. Whatever vendor is supplying that is also running a large PBX operation in a data center somewhere, and slicing off a little of it to dedicate to your organization in exchange for your subscription fees.
The connection of voicemail to custom greetings, ring groups, advanced call routing, etc. can also work within a local system. The biggest difference is that the setup and maintenance of complex voicemail procedures in a PBX will far outreach a hosted setup.