Lets go thorough ITU 7.11 or PCM
The original standard for converting analog voice to a digital signal is called pulse-code modulation (PCM).
PCM defines that an incoming analog voice signal should be sampled 8000 times per second by the analog-to-digital (A/D) converter( according to Nyquist’s theorem states that you need twice the number of samples as the highest frequency . As before mentioned the required bandwidth of human’s voice is 4000Hz so 4000×2=8000 samples are needed). A/D converters that are used specifically for processing voice are called codecs (meaning encoder/decoder). For each sample, the codec measures the frequency, amplitude, and phase of the analog signal. PCM defines a table of possible values for frequency/amplitude/phase. The codec finds the table entry that most closely matches the measured values. Along with each entry is an 8-bit binary code, which tells the codec what bits to use to represent that single sample. So PCM, sampling at 8000 times per second finds the best match of frequency /amplitude/phase in the table, finds the matching 8-bit code, and sends those 8 bits as a digital signal. Therefore bit rate can be easily calculated
8bits ×8000samples=64Kbps