Yes, Opus is the fastest and best option for real-time audio. It was designed to be flexible and to encode/decode at fairly low latencies. It sounds good for narrow-band (speech) at low bitrates but also works well at higher bitrates for music. And forward error correction is part of the codec standard.
It's possible to tweak the Opus settings to reduce that encode/decode latency substantially. Which might actually be worth doing for this use case. But there isn't quite a free lunch, here. The default Opus frame size is 20ms. Smaller frames lower the encoding/decoding latency, but increase the bitrate. The implementation in libwebrtc is very well tested and optimized for the default 20ms frame sizes and maybe not so much at other frame sizes. Experience has made me leery of taking the less-trodden-paths without a lot of manual testing.
It's possible to tweak the Opus settings to reduce that encode/decode latency substantially. Which might actually be worth doing for this use case. But there isn't quite a free lunch, here. The default Opus frame size is 20ms. Smaller frames lower the encoding/decoding latency, but increase the bitrate. The implementation in libwebrtc is very well tested and optimized for the default 20ms frame sizes and maybe not so much at other frame sizes. Experience has made me leery of taking the less-trodden-paths without a lot of manual testing.