Even if you have a VoIP telephone system, your call will eventually go across the PSTN (like the traditional landline calls), where network operators are able to do whatever they like. Plus VoIP and mobile operators use high-compression codecs that degrade the overall quality of the sound. A good codec can take a 9600bps audio stream (toll quality ISDN voice) and take it down to the required bandwidth by a factor of 10 or more.
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You’ll notice that there’s not too much of a difference between the track when played at CD quality and over the phone. All the instruments can still be heard clearly, and the same techniques I used to introduce clipping on the other two tracks had no effect (I could have kept amplifying the track until it did, but it proved the point). And because it’s fairly bland and inoffensive, you’re not going to upset anyone too much if it’s not their cup of tea.
For those still using analogue or digital phone systems, voice on ISDN circuits has a bandwidth of around 4000Hz. The audio from an analogue phone is sampled at 8000Hz, and because of the nyquist theorem, must be low-pass filtered to 4000Hz. So any frequencies in the audio beyond 4000Hz are lost. Interestingly, the human ear and mind are good at recognising human speech, so this is considered "toll-quality" voice, i.e. good enough to pay for. This is the reason why certain vocalised letters are difficult or impossible to tell apart on the phone. For example, the differences between B, P, T, C, etc. are almost entirely above 4000Hz.