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Automatically establishes a lightweight, dedicated SSE connection (/stream?call_id=<callId>) for each active call session (call.start(), call.accept(), call.joinSession(), and client.createCallSession()).
Subscribes to the Per-Call channel and receive real-time speech-to-text transcriptions, recordings events, and telemetry without flooding the global workspace event stream.
Automatically disconnects and cleans up the stream upon call termination (call.destroy()).
Removed duplicate event emissions across FiretellClient._initEventStream() and Call._handleSseMessage().
Decoupled per-call transcription events from client.events, strictly routing all live subtitles, chunks, and summaries directly to the owning Call instance (call.on(...)).
Decoupled WebSocket signaling (_handleWsMessage) to handle exclusively Firetell WebRTC signaling (call.offer, call.answered, call.state, etc.), isolating SSE transport for transcription.
Example Page Updated:
Added transcription.started and transcription.completed event logging to example/index.html.