Thank you for your question! LiveKit uses adaptive quality management for audio, which includes helpful features like congestion control and Discontinuous Transmission (DTX). These work together to keep your audio streams smooth and uninterrupted, even when the network is a bit congested. The

audioBitrate

setting in the

AudioPublishOptions

class lets you set the maximum audio bitrate, and this value helps configure the WebRTC

RTCRtpEncoding

parameter

maxBitrate

. While LiveKit does adjust audio quality based on network conditions, the current documentation doesn’t explicitly state if the

audioBitrate

value dynamically lowers below the set maximum during tougher network situations in the Flutter SDK. It’s likely that the system tries its best to keep audio quality within that set limit, but if you’d like, we can look into this further together to get a clearer picture of exactly how it behaves. Sources: livekit-android-sdk/src/main/java/io/livekit/android/room/participant/Local... | src/room/participant/LocalParticipant.ts | Bringing Zoom's end-to-end optimizations to WebRTC | Speech & Audio Processing | livekit/livekit_composite | DeepWiki | lib/src/participant/local.dart