But how often have you connected a VoIP call, only to have it suffer from jitter, lag or latency - or even fail to connect at all? The chances are, you're nodding your head right now. That's because this sort of thing continues to plague some VoIP services, making a red-headed stepchild of the entire concept.
But throwing the baby out with the bathwater is no solution. VoIP continues to hold sway because when done right, it delivers call quality that is every bit as good as that of a hard-wired, copper circuit. The trick is that you need to be sure that every time you pick up the phone, the quality of your call is consistently that good.
However, while your service provider will tell you that this is what you can expect, you probably still have a worry lurking in the back of your head. "But if the service providers say it'll work perfectly, how come sometimes when I connect calls over VoIP, the quality is poor ... "
Yes, it happens. The reasons largely fall into three shortcomings:
Assessment is absolutely critical because data networks weren't originally designed to carry voice traffic. The Internet has always worked on the concept of "best effort"; with "ordinary" data, it doesn't matter if some parts of a file arrive after others, in any old order. With voice (or video), lose some parts and suddenly you develop a digital stutter and communication goes out of the window.
These sorts of issues are more likely to confront small to medium business than they are to strike large corporations (though these are far from immune). That's because SMB networks typically don't have the throughput capacity that bigger companies enjoy. When taking into account that VoIP traffic uses around 10 times more capacity than data traffic - well, you don't need to be a network engineer to appreciate how that translates into dodgy call quality.
Add in some congestion, such as concurrent calls, or simply the combination of data and voice going into the same pipe, and things can very quickly go pear shaped.
Is it just a bandwidth problem, then? Bandwidth is certainly the first port of call, but there is more to it than that. A mere assessment of capacity as a measure of the ability for any given network to carry perfect VoIP calls is just the starting point. If your calls are to be consistently good (and only excellent is good), your service provider should demonstrate an understanding of the necessity for a live network assessment of the entire LAN and WAN paths. This is a necessary first point in understanding network readiness to implement VoIP successfully - or any real-time media application, for that matter.
Your service provider should absolutely start with the process of thoroughly analysing your network. If VoIP isn't going to fly, there's no point in "investing" in it; we all know that unless a call is clear, latency and jitter-free, it simply isn't good enough to communicate over.
Another crucial part of success is the onsite equipment used to terminate voice calls through your service provider's telecommunications network. Many providers use cheap data routers and gateways, with contended (many users of the same channel) connectivity mediums to save costs. Simply put, this causes poor call quality.
Once it is established that your network can handle it and your VoIP system is in place, it should be stress tested, constantly monitored with onsite probes and managed to ensure consistent performance.
After all, among the benefits of digital technology is that records are generated as a matter of course; these provide the necessary data to understand what's happening and how - the raw material to identify and eliminate any challenges before they interrupt your conversations.
