You clearly haven't made any PSTN calls on Skype. If you're looking for higher sample rates, it's relatively easy to use a B channel to transport g.722, or even bond a couple together for 128k AAC/MP3/any other codec you'd like.
Not only do you not have to deal with the problems of internet routing, you aren't stuck with underwater sounding CELP codecs.
Anyway, if you're arguing voip should replace the PSTN, I'd hope you'd at least have something hinting it could be as reliable.
> Why would you use an HD codec to transcode to G711? Call on any direction can sound only as good as G711.
The idea is to use the timeslot g.711 normally would sit in to transport your own codec. Studios use this technique pretty frequently when they hire voiceover talent or do remotes.
No one is saying VoIP is as reliable as PSTN. Not even mobile (GSM, CDMA, etc.) is.