Add live WebRTC voice conversations to the TUI

[copyberry]

Add live WebRTC voice conversations to the TUI

What changed

• Add feature-gated /voice, /voice mute, and /voice stop commands with local WebRTC audio and app-server signaling.
• Show live transcripts, conversation status, and microphone and speaker levels. Preserve captions and undelivered answers across thread switches.
• Speak final answers from voice handoffs while keeping delegated reasoning and commentary hidden and typed answers unspoken.
• Retry eligible startup failures once and clean up voice sessions on thread switches and disconnects.
• Stop voice and block late handoffs after a misalignment policy violation. Remove realtime event payloads and spoken text from receipt and debug logs.

Testing

Add coverage for voice command mapping, rejected starts, disconnect cleanup, delegated final-answer speech, late handoff rejection, and transcript and footer rendering.

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