Fix VoIP audio and auto-reconnect reliability#34
Conversation
- Update backend to handle VoIP disconnects explicitly and broadcast status. - Refactor frontend WebRTC connection logic into `initializePeerConnections` for reuse. - Fix frontend `voip_my_call` to re-establish WebRTC connections on page reload. - Fix frontend socket reconnection to re-join the signaling room via `voip_join_call`.
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This change addresses issues where VoIP calls would have no audio after a reconnection or page refresh, and where auto-reconnects (after network drop) would fail to restore the signaling channel.
Backend:
backend/socketio_handlers.py: Added logic tohandle_disconnectto detectvoip_rooms, mark the user as disconnected in theVoipCallmodel, and broadcastvoip_user_disconnected.Frontend:
frontend/contexts/VoipContext.tsx:initializePeerConnections.voip_call_joinedto use this new function.voip_my_call(used on page refresh) to callinitializePeerConnections, fixing the "can't hear" issue after refresh.handleReconnectto emitvoip_join_callinstead of justvoip_heartbeat, ensuring the socket is re-added to the signaling room on the server.PR created automatically by Jules for task 12874636545607941746 started by @8JP8