VoIP – Voice Over Internet
VoIP stands for Voice over IP, which means telephone voice calls over the Internet, or over an IP network
POTS to VoIP
In the old days, our plain old telephone service (POTS) was analog. As you spoke on the phone, the microphone would vibrate, and send electrical waves down the wire to cause the receiver to vibrate exactly the same at the other end to reproduce your voice. This worked quite well for many years since old AG Bell invented it, but the voice signal was subject to interference, losses, and distortion, and often did not come out quite right at the other end. Only one conversation could be held on one copper line at a time until they learned to multiplex several calls together, but even this took up a lot of resources.
In today’s digital world, almost all voice telephone traffic is digitized in some way before it is sent down the line and then converted back to an audible signal at the distant end.
If enough bandwidth is used, this method can provide crystal clear reproduction of your voice. To make voice traffic more cost-effective, the voice is sampled and then digitized and added together with many other voice conversations on the same transmission medium, whether it be copper wire, fiber, satellite or Internet.
To be even more cost-effective, the digital signal is compressed before it goes down the wire and then decompressed at the other end into good quality voice.
As transmission costs have got more expensive, they have developed clever methods of sampling the voice in ways that closely resemble the original, but are actually only approximations of the original sounds. In this way, they have achieved quality voice in bandwidths as low as 4kbps. In the old days, a regular analog phone call would take 64kbps of bandwidth.
The most common G.729 ACELP voice algorithm samples the voice in 10mS slices and uses 8kbps of bandwidth. When sent over the Internet, the voice packets need to be addressed and labeled so that they know where to go and what do with them at the other end. This addressing scheme takes up an additional 16kb of the header (or overhead) making each voice circuit 16+8=24kbps. So your typical VoIP call takes up 24kbps of your bandwidth.
Skype is another, free, universal form of VoIP calling, but this tends to take up bandwidths of about 40-80 kbps. If you have the bandwidth to spare, this is no problem, but if your bandwidth is scarce, you would be better off using a proper VoIP modem like a Linksys SP2012 through your satellite providers gateway.
Some documents state that Skype claims bandwidth usage 24-128 kbps. During some tests the typical bandwidths were about 40 kbps so 24-128 kbps would be correct; even tested on the low end. Even more reasonable during the tests it was found that call quality was reasonable even at bandwidths like 16 kbps but almost unintelligible at 12 kbps.
It is important that you have that bandwidth available during the call, otherwise you will have distorted sound, drop outs, dropped calls, or one way calls where only one party can hear.