It's a fair question. Tape interfaces had very limited bandwidth. The original Kansas standard used frequency switching (FSK) between four cycles of 1200Hz and eight cycles of 2400Hz. Later faster variations used the same frequencies but cut the number of cycles for faster load speeds.

MP3 compression works by removing frequencies that are (supposedly) too quiet to be audible. If you only have two very loud frequencies in a frame they pass through the compression process unscathed.

The switching hash around them and the tape noise may get munged but that doesn't affect the data stream. So vbr MP3 ends up being an efficient and clean representation, and cbr works too but isn't quite so compact.

Edit: in fact some radio and TV shows in the 70s would transmit software live, and that worked fine too. I never tried it, but I suspect you could probably use a phone line - for early adventures in software piracy.

Lossy codecs use psycho-acoustic models designed to encode music so it sounds okay to humans, by discarding frequencies that we're less likely to hear due to masking etc. -- Just need to avoid those frequencies.

Having said that the newer lossy codecs (e.g Opus and some AAC variants) which are more efficient (sound better at lower bitrates) and have less "problem samples" than MP3 would perhaps also be better for data use, like they are for music.

If you're going to play it over a speaker, you already have way more signal loss than the compression is going to introduce. The solution is redundancy.

The fidelity of standard non high bias tape at compact cassette speeds is already shit compared to even low bitrate mp3. It’s not a problem in practice.