Voice codecs  are designed to International Telecommunication Union (ITU) standards, which specify how segments of analog voice signals are to be encoded into digital data streams. The design of the particular codec used to digitize voice signals carried via a packet switched network determines both the minimum number of bytes that can be reasonably included in a voice packet and the throughput of packet bits that must be achieved in order to transmit a digitized voice signal. Below picture of Table shows, for example, the characteristics of three of the codecs that are most widely considered for possible use in setting up VoIP services.

All three are based on an 8000-hertz (Hz) sampling rate for analog voice signals. However, as shown in the table, the differences in encoding techniques create substantial differences in both the minimum duration of the segment of voice that is sampled and the amount of data transmitted to support regeneration of the analog signals at the distant end. The codec characteristics shown in the table, then, directly affect two characteristics of the voice signals heard by users over a digitized voice connection: delays and signal fidelity.