We should be able to build WebRTC-based voice servers, with legacy conversion on the server side between the old voice codec and the new one. That would also let us build web-based clients, running directly in the browser. It's also possible to use WebRTC standalone, so existing clients could use the standalone WebRTC library to connect to the voice servers.
In that case, all the people logged in with "new" clients would have good audio, those with "old" clients would have bad audio, and those with "new" clients would receive bad audio from those with "old" clients.
I was going to try to