Data compression is a technique frequently used in VoIP in order to transport and store data more efficiently. In the early days of VoIP available bandwidth was at a premium and VoIP applications had to contest bandwidth with other sometimes greedy applications. Voice packets in their natural format are sampled and stored in 64kb samples. This is how voice packets are transported over the telephone network, using time division multiplexing in time slots of 64kbps. However that is very inefficient over the internet so there was a need to find methods to compress the size of the packet whilst retaining the quality.
Data compression is not something new to VoIP it has been around and used in many other IT applications it was just with VoIP it was more noticeable, as the results could be detected by humans. The software used to compress voice packets is called codecs.
Codecs encodes an analogue voice signal into digital packets and compresses the data in order to make it lighter and more efficient to transport across the internet. For example the codec may reduce the packet size from 64kb to 32, 16 or even 8kb. Once transmitted in their new lightweight size the destination has the job of decompressing the packets and converting them back into the original analogue signals that we can hear.
The quality and efficiency of the codec software therefore has a bearing on the VoIP quality. There are several type of codecs all designed for specific use or circumstance. Some codec are very ‘lossy’ what this means is that they deliberately lose a lot of the data during compression/decompression With voice this is not such a big problem and doesn’t really effect call quality as most of the loss is at frequencies out-with the capability of the human ear.
Data compression is not only used for VoIP but also for Fax and raw data using the correct compression software is important to ensure good quality, so always ensure that the codec being used matches the requirements.