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          "quote": "...jumping in mid-sentence or waiting too long. Switching from fixed silence thresholds to semantic end-of-turn detection was night and day. One dimension I'd add: geography matters even more when your...",
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          "quote": "...him PTT by default as the problems of the model accurately guessing the end of utterance are just too great. I think it can be solved in the future but,...",
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          "quote": "...talking, I tried in a room with absolute silence too. ### Expected Behavior Silero VAD commits the turn after detecting user silence. Agent respond to the user message already timely...",
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          "url": "https://github.com/livekit/agents/issues/4087"
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          "handle": "Jeeva-MV",
          "handle_url": "https://github.com/Jeeva-MV",
          "quote": "...any suggestions on how to tackle this will be helpful. Attaching the STT and VAD configs along with a sample log file for reference. Azure-configs (Tried with multi-lingual detection as...",
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          "url": "https://github.com/livekit/agents/issues/1893"
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        "note": "Printed as a crosstalk line carrying the coherence and the lag: the caller channel looks like a lag-shifted copy of the agent's own audio, so the barge-in may be leaked output rather than a caller."
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          "handle": "1967xbill",
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          "source": "github",
          "url": "https://github.com/livekit/agents/issues/6956"
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          "date": "2026-02-10",
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          "quote": "...`livekit-plugins-silero==1.6.10` * `livekit-plugins-openai==1.6.10` * WebRTC browser voice session * Silero VAD used for turn detection/interruption ### Checkpoints tested Previous checkpoint: ```text 597d30b3ec076608d059477bb14cfeffdf951bf5cae370d38f65d33bbfe82004 ``` Current bundled checkpoint: ```text 1a153a22f4509e292a94e67d6f9b85e8deb25b4988682b7e174c65279d8788e3 ``` The...",
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          "quote": "...duplex audio: if you don’t subtract your own TTS audio (or at least gate VAD while TTS is playing), you’ll get false user-starts. Even a crude ‘TTS playing → raise...",
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          "quote": "...no BotStartedSpeakingFrame within 3.0s of pausing frame processing (e.g. a TTS context completed with no audio) — force-resuming ERROR | pipecat.processors.frame_processor:push_error_frame:732 - ElevenLabsTTSService#1 error: ElevenLabsTTSService#1 no BotStartedSpeakingFrame within 3.0s of...",
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          "quote": "...the turn succeeded, and no `ErrorFrame` is pushed. The bot produces an empty turn (dead air) and nothing is logged, so error accounting, failover wrappers (e.g. `ServiceSwitcherStrategy`), and observability all...",
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          "quote": "...follows. assist_satellite.announce Service call succeeds. Satellite changes responding → idle . LED ring activates. No audio. Tested with preannounce: true as well. Direct tts.speak Tested directly against the Voice PE...",
          "source": "home_assistant_forum",
          "url": "https://community.home-assistant.io/t/voice-pe-responds-with-led-ring-but-produces-no-audio-announce-tts-and-play-media-all-stay-silent/1018477"
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          "handle": "mannyb223",
          "handle_url": "https://github.com/mannyb223",
          "quote": "...during the incident): - The greeting is \"spoken\" (transcript logged, usage chars counted) but no audio exists. - With `pause_frame_processing`, the TTS service pauses its process task after the utterance...",
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      "statement": "The agent's speech synthesis silently produces no audio even though it logs the turn as spoken, and the safeguard meant to detect and recover from this never triggers, leaving the caller in permanent silence.",
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          "handle": "mannyb223",
          "handle_url": "https://github.com/mannyb223",
          "quote": "...by the LLM token stream and end-of-turn finalisation, fully decoupled from playback. - On interruption, the aggregation is committed rather than discarded. On a turn ending with `EndFrame` (e.g. a...",
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          "quote": "...completes normally: without it, the assistant context would lose text that was spoken. On interruption, however, the same fallback can add text that was not delivered to the listener. For...",
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      "statement": "When the caller interrupts mid-response and the system lacks word-level delivery timestamps, it falls back to recording the full intended text as spoken, so the conversation history includes words the caller never actually heard.",
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      "references": [
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          "date": "2026-08-27",
          "handle": "vignesh-koruverse",
          "role": "report",
          "source": "github",
          "url": "https://github.com/livekit/agents/issues/7010"
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          "quote": "...end-of-turn bounce, which sleeps the endpointing delay before delivering the turn. With `turn_detection=\"manual\"`, a VAD `START_OF_SPEECH` inside that window cancels the pending task unconditionally (`audio_recognition.py`, SOS handler). Audio tail around...",
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          "quote": "...line diagnosis instead of an afternoon. Env: pipecat main, `gemini-3.1-flash-live-preview`, SmallWebRTC transport, server side VAD, python 3.13 on macOS. Happy to send a PR. The bit I'd want your call...",
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          "quote": "## Summary The AMD classifier consumes two **independent** input streams — a transcript from AMD's own STT and speech-start/end boundaries from the session's VAD. When those streams desync, a live...",
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          "quote": "...was dispatched ~8s after the INVITE — and ~2.5s after the caller had already hung up. Why we think the delay is dispatch-side: this was 1 of 208 inbound calls...",
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          "url": "https://community.livekit.io/t/inbound-sip-call-agent-job-dispatched-8s-after-the-invite-reached-livekit-cloud-caller-had-already-hung-up/1393"
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