When converting analog voice signals to digital signals, what must the sampling rate be?

When converting analog voice signals to digital signals, the sampling rate must be at least twice the highest frequency component present in the signal. This principle is derived from the Nyquist theorem, which states that to accurately capture and reproduce a signal without any loss of information, the sampling frequency needs to be at least twice the maximum frequency of the signal being sampled. For typical audio signals, which in human voice can go up to approximately 3-4 kHz in terms of essential frequencies, a sampling rate of at least 8 kHz (which is twice the highest frequency of 4 kHz) is required to ensure that all the nuances and variations of the original signal are captured without aliasing. This practice is essential in fields such as telecommunications, where digital representations of voice signals are a standard for communication systems.