How to improve bitrate/quality for audio coming from backend service sender?

[DiggidyDave]

(Note: if it isn't obvious, I'm sort of new to lower-level webrtc stuff, so a) sorry if this is a bad question and b) bear in mind this might be my first time encountering some aspects of this :-))

I have a client consuming generated audio from a server, through an SFU, and quality is poor. Looking on chrome webrtc internals, I see the bitrate of that received audio stream is ~25 kbps, while the source audio wav I checked on back is ~384 kbps. Is there something I can set on the server/pion side to make this use more bandwidth? Or is this a client setting?

On server, I'm using:

audioTrack, audioTrackErr := webrtc.NewTrackLocalStaticSample(
    webrtc.RTPCodecCapability{
        MimeType: webrtc.MimeTypeOpus
    }, 
    uuid.NewString(),
    uuid.NewString())

at the moment, I'm getting generated audio in the form of wav files, and I'm using github.com/hraban/opus and github.com/youpy/go-wav to transcode wav to opus, then every 20 ms if there is audio available I'm submitting 20 ms opus "pages" to that track using track.WriteSample where track is a webrtc.TrackLocalStaticSample created by the above call.

Anything amiss? Anything I can do here?

[DiggidyDave]

I didn't find that the opus encoder has a much lower default bitrate, so I changed that to see how it went, and I think it might have helped a bit. But it seems to differ depending on client, so wondering if this is mostly a client issue.