Back in the early days of the Internet, I met some of the Bell Labs people working in that area. What they didn't like about an IP-based network was the jitter. That was totally unacceptable for voice. They wanted something with reliable clocking, and had come up with Datakit. That sends all packets for a given call over the same path, in order. You don't open a virtual circuit unless all the nodes have enough bandwidth for it. Telcos still use Datakit, and Asynchronous Transfer Mode is a successor to it.
I never dreamed that people would accept the degradation of telephony to 1 second of delay jitter with random dropouts and echoes.
I remember when cross-oceanic long distance calls required both parties to shout as loud as they could into the phone -- and often still not be heard well enough to make out what they were saying.
Modern IP telephony often has very high quality voice reproduction (my wi-fi to wi-fi Fi calls sound fantastic), in exchange for some timing issues. Echoes usually get solved in software (usually), and dropouts seem to be the main complaint.
In exchange, my wife can call her family in South Korea for approximately nothing, using the same data backbone as we use to watch movies and read web pages.
You can have ip over time multiplexing data link layer, like the IP over ATM. I believe they were still used in the core voice networking.
IP won because they are more flexible, and open. And the Internet cement the win because of that.
That's way planning ahead too far works less and less successful for bigger and bigger project. Too much dynamic is embed in the long wiring process. It becomes impossible to have a plan work out correctly.
1 second of delay jitter with random dropouts and echoes are typically caused by problems at the ends (poor wifi/cellular signal, too much load on cpu during a high res video call, laptop overheating, etc). UDP protocol allows to make a call even in the presence of such issues, whereas circuit switching with reliable clocking would simply not work at all.
Other networking protocols coming from a telco background, in particular ATM and ISDN, were all circuit switched, and had suitable resource reservation for QoS. Acceptance of telephony degradation was probably driven by cost: VoIP was free and that made a difference, especially for international calls. In my experience, in 2020 the VoIP calls I make are really high quality, even better than 1980s-style ISDN calls, and the main cause of audio quality degradation are people using "hands-free" setups with their laptops.
I never dreamed that people would accept the degradation of telephony to 1 second of delay jitter with random dropouts and echoes.