stale server sockets, passive non-initiators, and race conditions
during peer connection setup.
Fix users unable to see or hear each other in voice channels due to
stale server sockets, passive non-initiators, and race conditions
during peer connection setup.
Server:
- Close stale WebSocket connections sharing the same oderId in
handleIdentify instead of letting them linger up to 45s
- Make user_joined/user_left broadcasts identity-aware so duplicate
sockets don't produce phantom join/leave events
- Include serverIds in user_left payload for multi-room presence
- Simplify findUserByOderId now that stale sockets are cleaned up
Client - signaling:
- Add fallback offer system with 1s timer for missed user_joined races
- Add non-initiator takeover after 5s when the initiator fails to send
an offer (NON_INITIATOR_GIVE_UP_MS)
- Scope peerServerMap per signaling URL to prevent cross-server
collisions
- Add socket identity guards on all signaling event handlers
- Replace canReusePeerConnection with hasActivePeerConnection and
isPeerConnectionNegotiating with extended grace periods
Client - peer connections:
- Extract replaceUnusablePeer helper to deduplicate stale peer
replacement in offer and ICE handlers
- Add stale connectionstatechange guard to ignore events from replaced
RTCPeerConnection instances
- Use deterministic initiator election in peer recovery reconnects
- Track createdAt on PeerData for staleness detection
Client - presence:
- Add multi-room presence tracking via presenceServerIds on User
- Replace clearUsers + individual userJoined with syncServerPresence
for atomic server roster updates
- Make userLeft handle partial server removal instead of full eviction
Documentation:
- Add server-side connection hygiene, non-initiator takeover, and stale
peer replacement sections to the realtime README
Experimental fix for solving replacement of mic audio when enabling screenshare audio on windows electron
On Windows Electron, startWithElectronDesktopCapturer uses getUserMedia({ chromeMediaSource: 'desktop' }) for both video and audio. This getUserMedia desktop audio call can interfere with / replace the existing mic getUserMedia stream, killing voice audio.
BREAKING CHANGE: possibly streaming
1. Server: WebSocket ping/pong heartbeat (index.ts)
Added a 30-second ping interval that pings all connected clients
Connections without a pong response for 45 seconds are terminated and cleaned up
Extracted removeDeadConnection() to deduplicate the cleanup logic between close events and dead connection reaping
2. Server: Fixed sendServerUsers filter bug (handler.ts:13)
Removed && cu.displayName from the filter — users who joined a server before their identify message was processed were silently invisible to everyone. This was the direct cause of "can't see each other" in session 2.
3. Client: Typing message now includes serverId
Added serverId: this.webrtc.currentServerId to the typing payload
Added a currentServerId getter on WebRTCService
