Just as in classical telephony, the voice signal is sent to the phone microphone handset in the form of an analog signal. An analog to digital converter transforms this signal into a digital one which is then encoded according to an audio format. There are different encoders for compression of a conversation with varying degrees of quality. Depending on the type of compression, some loss of voice signal information occurs, which, however, is mostly subjective and irrelevant.
Voice Data, after the compression, is transferred via the network. For this continuous data stream of compressed voice signal, the conversation is divided into small data packets, before they are sent to the network. Then, these packets arrive at the specific destination network “nodes”, so-called routers, which direct the IP-packets to their final point, maybe via different paths.
The voice packets, before they are sent to the network line, they are first temporarily stored in a memory location called the buffer. At the destination end, voice packets pass through a digital to analog converter so that to be transformed back to human speech. For example, the voice packets delivery can be compared with normal mailing post, which addresses, mails and delivers individual packets of data.
To transfer data using VoIP we currently use the so-called Internet protocol version 4 (IPv4). The next version – IPv6 – was specified in the mid 90-ies, but the need to expand its use has not been taken seriously until the last few years. Unfortunately, the IP protocol works on the principle of “Best Effort” and thus it does not guarantee 100% packet delivery. This is the main reason that we do not always have good quality in VoIP. The newest IP Protocol version 6 offers the so-called “quality of service” under which the voice data transmission will have better quality.
In order to establish a connection between a VoIP telephone system and the traditional PSTN telephone network, we need to use the so-called Gateways. These Gateways are connected to the IP Data network as well as with the PSTN telephone network and transmit requests in both directions. In this case, IP-packets from one side are converted into digital voice stream on the PSTN side.
Since everything now runs over IP (our computer data, voice, video etc) we can have the integration of different types of data on to a single IP network. This concept is usually called “network convergence”. The meaning of convergence is that we have one common network (the IP Network) which transmits all kinds of information – voice, data, video, text and images.