Specifically, their ability to capture and transmit usable speech is really poor. Between echo and poor noise gating, I'm not exaggerating when I say that I can make out fewer than 1/3 of the words in my team's conference calls.
I don't care how efficient the video codec is, if I can't clearly understand what the other people are saying, it's all useless.
We only ever get reports about problems when people are actually dialing into something over POTS.
To test your network, go to https://www.dslreports.com/speedtest and see if you have significant jitter or bufferbloat. If your rating is an A, then your good to go, otherwise its time to clean up your local network :P
Those cheap Cisco conference phones do sound terrible, and I don't think people replace them very often.
Google, Mozilla and all the organizations that have collaborated on WebRTC have spent 7+ years tuning this component. Much like Opus versus AMR-WB, I doubt there is anything notably better in the proprietary software/hardware world.
The orgs that develop the WebRTC standard have spent years implementing and experimenting with nearly every algorithm that is available for each of the various components (resulting in Opus, AV1, WebRTC AEC, etc), combing research papers daily and often working with researchers directly.
Stuff like WebRTC/OPUS is limited by the fact it runs in the browser or consumer hardware. It's not bad, it's just a different problem domain than commercial audio, and has different constraints. High quality conferencing equipment runs on a dedicated unit that your A/V integrator installs either in the room, or in a server closet and connects over Dante to the analog front end in the room, and usually is a package deal with good microphones (which could be wireless, or the hot thing now is to use beamforming).