When you make a VoIP telephone call and begin to speak, the first step in any VoIP call is to convert the analogue signal of your voice into digital data. This is done with an Analogue-to-Digital Converter (ADC). As its name suggests, an ADC converts continuous analogue data (e.g. your voice) into smaller pieces which are assigned a numerical value.

Once your voice is digitized, the next step is to compress the audio data using a codec (enCOder/DECoder) which significantly reduces the amount of digital data while maintaining audio quality. The compressed digital data can now be sent over the Internet.

The data stream must be divided into smaller chunks, known as ‘packets’ which, besides containing the audio data, contains information about the origin, the destination, and a timestamp that allows it to be reconstructed in the correct sequence.

Most VoIP uses a transport layer called User Datagram Protocol (UDP) which is faster than TCP. A commonly used application layer is called Real-time Transmission Protocol (RTP) - originally developed for delivering audio and video over the Internet. RTP provides information about the sequence of the data packets so they can be reconstructed in the correct order at their destination.

RTP also has the ability to drop packets if they do not arrive within a certain amount of time. This is necessary for telephone conversations because if the telephone software waited for every packet of information to arrive before reassembling it there would be unacceptable delays in the audio stream.

Even though some of the packets are dropped, there is usually still enough information to make the conversation legible. The number of packets that will be dropped depends on the speed of your Internet connection in the distance between the two parties.