Audio routes through call speaker instead of media speaker when joining as a viewer-only participant
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Nadie ha tomado este issue todavía.
Evaluación
- Dificultad
- 4/5
- Tiempo estimado
- 3-5 días
- Aptitud para principiantes
- 38/100
- Tipo de issue
- Nueva funcionalidad
- Claridad
- Bastante claro
- Estado de actividad
- Tranquilo
- Stack tecnológico
- flutter
- Área
- audio-video-rtc, mobile
Línea de trabajo
Start by reading the audio-routing setup around RoomOptions and Hardware.instance, then trace how the SDK handles audio sessions on Android and iOS when a viewer joins without publishing tracks. The issue does not name files or tests, so first locate those entry points and relevant platform tests. Done means receive-only participants use media playback routing without breaking audio for participants who publish tracks.
Escrito por el modelo de indexación a partir del texto del issue.
Descripción
Is your feature request related to a problem? Please describe.
When joining a livestream session as a pure viewer (no microphone, no camera), the audio is incorrectly routed through the device's call/earpiece speaker instead of the media/loudspeaker — the same routing used for phone calls.
This occurs even though the viewer is not publishing any audio or video track. The expected behavior is that a participant in a receive-only role should use media playback audio routing, consistent with how any video player or media app behaves. Instead, LiveKit appears to initialize WebRTC in MODE_IN_COMMUNICATION (Android) / AVAudioSessionCategory.playAndRecord (iOS) regardless of whether the local participant is actually publishing.
Describe the solution you'd like
When a participant joins with no published tracks (viewer-only / subscriber-only role), LiveKit should automatically apply media playback audio routing:
• Android: Use AudioManager.MODE_NORMAL with loudspeaker, not MODE_IN_COMMUNICATION
• iOS: Use AVAudioSessionCategory.playback instead of playAndRecord
Ideally, expose a participant role hint (e.g. ParticipantKind.viewer or a audioMode option in RoomOptions) so developers can explicitly declare the intended audio routing behavior at connection time, before WebRTC initializes the audio session.
Describe alternatives you've considered
• Calling Hardware.instance.setSpeakerphoneOn(true) before room.connect() — overridden internally when WebRTC initializes the audio session on first track subscription.
• Manually reconfiguring AudioSession after the first remote audio track is subscribed — partially works but requires a delay and is fragile, as LiveKit re-acquires the audio session asynchronously.
• Using AudioOutputOptions(speakerOn: true) in RoomOptions — does not change the underlying audio mode from communication to media playback.
Additional context
• SDK: livekit_client (Flutter)
• Platforms affected: Android & iOS
• Use case: Livestreaming with a single host and many viewer-only participants
• Root cause: WebRTC initializes the audio stack in voice communication mode unconditionally, even for receive-only participants. This is by design in WebRTC but should be overridden by LiveKit when no local tracks are published.
- Lenguaje dominante
- Dart
- Estrellas
- 421
- Forks
- 260
- Merge medio
- 2 d 5 h
- PR fusionados (30 d)
- 17
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