When businesses understand that VoIP should be embraced and not rejected then the benefits just start rolling in . To embrace VoIP the business should really understand how it works and what they need to do to give it the best chance of working . Talking to a VoIP/cloud/hosted telephony specialist such as Phoenix Link ( www.phoenixlink.co.uk ) is a good place to start.

Voice signals , frequencies are  analogue….. not digital. Voice communication can only go onto a data network if it is converted into a digital equivalent that can be converted back into “voice” when it reaches its destination.

The conversion of audible sound into data is called quantization. The process of quantizing voice is typically done by sampling the sound at eight thousand times per second, assigning one of eight bits to each sample. This yields a 64-thousand bit-per-second data stream that can be reproduced at the other end.

The additional added packet information (overhead ) required to get the voice data stream across the data network when there are several simultaneous conversations can cause issues on the network , and therefore voice is generally compressed unless there is lots of available bandwidth or the voice data network is a separate network to the general data network.

Compression as a technology in not something that is new – it has been used for many years with fax machines. Voice communication can be compressed into a much smaller data stream ( than the original ) with almost undetectable loss of quality. Compression standards are expressed as CODEC ( Coder / Decoder ) , of which a common one is called G.729 which cuts the actual voice data stream down to 1/8 of the original size (the overhead remains the same). The highest quality voice has the CODEC G722 although the most common CODEC for voice is G711 which uses 64 kbps over bandwidth ( plus overhead)

A basic element of a data network is the ability to recover from lost packets by tracking and retransmitting any portions of the data that do not arrive intact and on time. The packets that handle this are a variety called Transmission Control Protocol, or TCP. These TCP packets cannot be used for voice communication. It would result in poor quality when missing packets were re-sent and inserted into the sound at the other end in the wrong place. It’s much better to just drop any defective or missing packets and listen to the part that does arrive as expected. Remember, you have 8,000 samples per second. You can lose some and still understand what is said. To accomplish this, voice uses exclusively packets designated User Datagram Protocol, or UDP. These packets are “best effort” delivery.

A subset of UDP packets is used to further enhance the delivery of voice packets in a timely manner: Real-time Transport Protocol, or RTP. This streaming-specific protocol improves the quality of sound by giving special attention to packets that may arrive late or out of sequence (called jitter).

In summary ; as long as the data network is configured correctly , with sufficient bandwidth and the right CODECs are used there is no reason why a user would even realise that their conversations are taking place over VoIP rather than the PSTN