## Why
The voice host drained incoming RTP packets without playing them. Route received audio through decoding and speaker output while preserving speaker suppression boundaries.
## What changed
- Add a GStreamer pipeline with RTP jitter buffering, Opus decoding, conversion, and resampling into the native speaker sink.
- Rebuild playout when the speaker is re-enabled, reject packets received before the new pipeline started, and discard queued audio and echo references on speaker transitions while preserving capture history.
- Limit each receive pass to 64 packets so incoming media cannot indefinitely delay audio controls.
- Prevent retired sink failures from failing a new speaker generation, and retain the last coherent playback delay estimate during callback contention.
## Testing
Add tests for real Opus playback and stale packet rejection, combined microphone mute and speaker suppression, retired sink failures, delay contention, and echo-reference reset without losing capture history.
GitOrigin-RevId: c91c94a2ba2347ac98af4e7fd1dc331ae766cc45
Benjamin Carlsson
a9896da3fe