When you're doing voip you're doing so over UDP because TCP has far too much latency to be agreeable, but this means a lost packet will be a digital distortion. The nature of the internet is that packets get lost or arrive out of sequence- that's why TCP exists in the first place.
Just like in the old days of 9600 baud leased lines it was hard to get acceptable console performance, and now people have megabit connections and CLI performance is not unusually constrained over the megabit connections. Of course its only using a fraction of 1% of the capacity.
Also remember that end users might have megs of bandwidth to carry K of voice traffic, but somewhere along the line there's someone constrained. Just like work offices don't have to be dumps, but its hard to put a beancounter $ value on not working in a dump, leads to a lot of open office foolishness.
Remember, the codec has to handle less than 20 ms latency. Running a sizable ANN (fast) or HMM in that time is not easy or even possible. A small one will be no good for various speakers.
Instead the codecs use various specialised error hiding tricks and adaptive strategies.
Heck, Codec2, one of the strongest voice codecs around could be used if all else fails. It is beginning to be used in SDR and amateur radio - optimised for availability and intelligibility not quality. It would be nice to have a variant integrated with CELT for higher quality.
Our pbx is hosted by http://sipworxx.com/ and it works great.
I use this with good Polycom "HD voice" devices. I'm also rather picky about voice quality.
Grandstream, Yealink, etc. all provide pretty easy to administer Asterisk-based PBX appliances that can be paired with a SIP trunk of your choice (shout out to Flowroute here, they're great).
If you're feeling more adventurous sipXecs from sipfoundry is pretty easy to configure relative to the additional complexity it brings over the more simplified alternatives. I went from having 0 real world experience with SIP to managing a small installation for my aunts shop as well as one at home in a couple days.
For proprietary offerings there's of course Skype for Business, etc. and I would argue they aren't terribly hard to configure or maintain either.
Most of the work is just in network configuration once you have multiple sites and you need to ensure QOS going over your MPLS or whatever to ensure your voice packets take priority - that can be more difficult but that's something a competent network admin can help you with.
Edit: "canbe"
Lots of wifis, corporate especially block all UDP traffic. Sometimes they even block all non http/https traffic. Apps like Skype or Biocoded will find a way to route the call, but that can sometimes mean it has to go over https!