Why is telephone audio sampled at 8 kHz?
If anyone cares to dig I think they will find that before Bell Telephone started multiplexing voice lines they did a lot of research on the frequency content of the human voice.  They originally used test groups to develop the bel audio unit and the power distribution of the voice as well as the sensitivity of the human ear to various frequencies.  They developed a band pass characteristic that peaked around 2.1 KHz and rolled off below 300 and over 3000 HZ.  That gave a good human sounding voice when done correctly.  All that was analog.
AM radio expanded that to 5 KHz to include music that was acceptable to most folks when we were young and had good ears.  Television flyback transformers were designed to run at ~17.5 KHz because there was a magic number for picture reproduction and most folks could not hear the whine.  Single sideband radio was commercialized in the 1960's and needed very sharp cutoff frequencies.  I used radios with filters at 2.1 and 3.1 KHz. 2.1 had some Donald Duck characteristics.  3.1 sounded fine, again with young ears.  Audio bandpass was increased to 20 KHZ or better with FM because the higher carrier frequencies could handle higher bandwidth for better music reproduction.  Stack up some xylophones or bells or other high pitched instruments and they can get enough harmonic energy into the higher frequencies.  OTOH, as was state, most folks cannot hear it.  

The bottom line is that anyone claiming they need 20 KHz bandwidth for voice is not paying attention.  3 KHz will do it, 5 will give you some margin.  If it does not sound right then something other than the bandwidth is the issue.

When digital signaling was being developed folks who know figured out that no matter how weird a waveform looked it could be broken down into a set of sine waves.  The harmonic mixing of those waves produced the typical spikey pattern of voice or music.  Lastly, Nyquist did research on the digital sampling rate needed to reproduce a sine wave at a given frequency.  It turns out that it takes 2 samples to make a sine wave so the highest frequency that will be reproduced is half the sample rate.  You want 5 KHz of audio then sample at 10 KHz.  Fine for voice.  You want higher fidelity music than most folks can hear then sample at 40 KHz or so to get 20+ KHz.  

One more tidbit is sampling vs bitrate.  If you sample at a given frequency then multiply that by the word length you will get the minimum bitrate needed to produce the desired signal.  Reduce bitrate and the size of the sample word will get cut to meet the new bit rate at a given sample rate.  That is all "lossless" encoding. This is all from memory and trying to find current data.  It's there if somebody looks for cites.  I'm not going to bother as I am getting too old to care.  I just got tired of wading through a lot of obvious mythical issues when I got interested in doing some audio capture.