44.1kHz sampling is sufficient to perfectly describe all analog waves with no frequency component above 22050Hz, which is substantially above human hearing. You can then upsample this band limited signal (0-22050Hz) to any sampling rate you wish, perfectly, because the 44.1kHz sampling is lossless with respect to the analog waveform. (The 16 bits per sample is not, though for the purposes of human hearing it is sufficient for 99% of use cases.)
22050 Hz is an ideal unreachable limit, like the speed of light for velocities.
You cannot make filters that would stop everything above 22050 Hz and pass everything below. You can barely make very expensive analog filters that pass everything below 20 kHz while stopping everything above 22 kHz.
Many early CD recordings used cheaper filters with a pass-band smaller than 20 kHz.
For 48 kHz it is much easier to make filters that pass 20 kHz and whose output falls gradually until 24 kHz, but it is still not easy.
Modern audio equipment circumvents this problem by sampling at much higher frequencies, e.g. at least 96 kHz or 192 kHz, which allows much cheaper analog filters that pass 20 kHz but which do not attenuate well enough the higher frequencies, then using digital filters to remove everything above 20 kHz that has passed through the analog filters, and then downsampling to 48 kHz.
The original CD sampling frequency of 44.1 kHz was very tight, despite the high cost of the required filters, because at that time, making 16-bit ADCs and DACs for a higher sampling frequency was even more difficult and expensive. Today, making a 24-bit ADC sampling at 192 kHz is much simpler and cheaper than making an audio anti-aliasing filter for 44.1 kHz.
https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampli...