Why Opus became the default

Opus is the default in WebRTC, in most SIP stacks, and in every browser. Not because it wins a benchmark, but because it is the only codec that does not need to be swapped when the network changes.

The problem it solves

A call starts on wifi and moves to cellular. Bitrate collapses from 64 kbps to 24 kbps. With G.711 you are already uncompressed; there is nothing to give up. With a fixed-rate codec you either drop packets or drop the call.

Opus switches modes within the same stream. Narrowband to fullband, 6 kbps to 510 kbps, without renegotiating.

What that costs

Latency. Opus at low bitrates looks further ahead than G.711 ever did. The default 20 ms frame is a compromise; at 2.5 ms frames the algorithm has less to work with and quality drops faster on lossy links.

When not to use it

If both endpoints are on a controlled LAN and the audio is music rather than speech, a transform codec at a fixed high bitrate will sound better. For everything else, Opus.