To add to the technical detail of /u/panswere's answer:

ensuring that call quality was minimally clear while allowing as many conversations at the same time as they could manage

These older carrier networks use a signaling protocol called SS7 or Signaling System 7. The protocol basically handles the communication required to initiate, pick up, modify or end a call. The signaling will have a specific set of parameters to determine HOW media is sent from one carrier to the other using identifiers like:

Point Codes - Essentially the location of the carrier's equipment

CIC Codes Essentially the channel to be used

SS7 interconnects are done using T1s (US) or E1s (Europe). I'm mostly familiar with T1s, so I'll be covering those, so just note that there are slight differences between T1s and E1s. It's also important to note that T1s can also fit within slots in larger connection channels (DS3s for those familiar with the terminology).

Anyways, a T1 is capable of passing 193,000 bytes per second and each T1 has 24 channels. Divide 193,000 by 24 and you get 8041 bytes per second, which is the highest bit rate any call going over these networks will have. See the problem?

As mentioned, services like Skype, Line2, Voxox, etc can get around these requirements by sending traffic over VoIP. VoIP uses a protocol called SIP (rarely one called H323) to set up and tear down the calls and sends media over IP. Instead of point codes and CICs, the signaling gives an IP and a port to use. Without the 8,000 bytes per second carrier limit standing between endpoint A and endpoint B they are free to chose whatever bit rate they need.

TL;DR: Carriers have limits to the quality because of systems and methods implemented long ago that are unlikely to change over the next 10 years because there's SO much equipment to support it in place. HD goes over VoIP which doesn't have those limitations.