There're tons of ways to fine-tune WebRTC that it wouldn't corrupt audio in poor network - it has all of the controls to smoothly trade-off latency vs quality. Not just NACKs - FEC, disable PLC/Acceleration/Deceleration, larger JB (tons of parameters) etc.

Most of the glitches I heard with OpenAI's Voice were not WebRTC related - but rather, to my ear, they sounded more like realtime issues with their inference - which is a very different component to optimize.