>Store the call audio at 24Kb/s Opus - that's 180KB per minute

Why OPUS though? There's dedicated audio codecs in the VoiP/telecom industry that are specifically designed for the best size/quality for voice call encoding.

Opus pretty much blows all those codecs out of the water, in every conceivable metric. It's actually pretty impressive that a media codec is able to universally exceed (or match) every previous one in every axis.

Still, it's based on ideas from those earlier codecs of course :)

Opus is one of those codecs. Older codecs like g711 have better latency and steady bitrate, but they compress terribly. (Essentially just bandwidth and amplitude remapping).

Opus is great for a lot of things and realtime speech over sip or webrtc is just one.