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Fix VoIP audio and auto-reconnect reliability#34

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8JP8 merged 1 commit intomainfrom
fix-voip-reconnect-12874636545607941746
Jan 19, 2026
Merged

Fix VoIP audio and auto-reconnect reliability#34
8JP8 merged 1 commit intomainfrom
fix-voip-reconnect-12874636545607941746

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@google-labs-jules
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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 to handle_disconnect to detect voip_ rooms, mark the user as disconnected in the VoipCall model, and broadcast voip_user_disconnected.

Frontend:

  • frontend/contexts/VoipContext.tsx:
    • Extracted peer connection creation into initializePeerConnections.
    • Updated voip_call_joined to use this new function.
    • Updated voip_my_call (used on page refresh) to call initializePeerConnections, fixing the "can't hear" issue after refresh.
    • Updated handleReconnect to emit voip_join_call instead of just voip_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

- 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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@8JP8 8JP8 marked this pull request as ready for review January 19, 2026 18:55
@8JP8 8JP8 merged commit 2918ccd into main Jan 19, 2026
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@8JP8 8JP8 deleted the fix-voip-reconnect-12874636545607941746 branch January 19, 2026 18:55
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