● Then the issue has become clearer. The agent itself is functioning normally. Looking at the logs again: time gap=0ms (960 samples) Timestamp normalization is working correctly. The reason packets are sparse on the second track is simply because the agent is barely responding to the room-client. Suspected Cause: The room-client's microphone input is not being properly recognized by the Agent's STT. Agent Behavior: room-client voice → Not recognized → No response (or short responses like "Please say that again") Playground/Web voice → Recognized → Normal response Possible Causes: Echo issue: Speaker output feeding back into the microphone, confusing the agent. Audio format: The Agent's STT might not handle Stereo Opus well. Voice quality: Quality issues with PipeWire capture. Test Methods: Connect both the room-client and Playground. Speak through the Playground and check if the agent's response is heard on the room-client. Check Agent logs to see how the room-client voice is being processed by the STT. Or a simple test: Try speaking in the Playground while the room-client is running. Does the agent's response come out through the room-client speakers? "When I speak in the Playground while the room-client is active, if the agent responds, I can hear that response on the room-client. It seems like it just isn't receiving my input? Unlike my laptop in 'meet'? What is the reason?" ● Hmm, then to summarize the situation: room-client mic → LiveKit → Playground (Laptop): ✅ Heard clearly room-client mic → LiveKit → Agent: ❌ Not recognized Playground mic → LiveKit → Agent: ✅ Recognized The room-client's audio is being delivered to LiveKit normally (since it can be heard on the laptop). The problem lies in how the Agent processes the audio from the room-client. do you know this case?

Thanks for reaching out, 하대훈. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

Resource

Do you mean which resource (docs/logs/code) you want? Please tell me which of the following you need and I’ll provide it: • Documentation for

pre_connect_audio

/ AudioInputOptions • RoomIO setup examples (including

participant

and start sequence) • STT/VAD troubleshooting guide and logs to check • Example code to subscribe directly to a participant audio track • How to force mono downmix / remix and resample Or paste the specific log or code snippet you want me to analyze.

100 / 58. 3.100 [2025-12-23 232434.686] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Input #0, sdp, from '/tmp/livekit_speaker.sdp': [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Metadata: [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: title : LiveKit Speaker [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Duration: N/A, start: 0.000000, bitrate: N/A [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Stream #00 Audio: opus, 48000 Hz, stereo, fltp [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Stream mapping: [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native)) [2025-12-23 232434.687] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Press [q] to stop, [?] for help [2025-12-23 232434.747] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Output #0, pulse, to 'default': [2025-12-23 232434.748] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Metadata: [2025-12-23 232434.748] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: title : LiveKit Speaker [2025-12-23 232434.749] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: encoder : Lavf61.7.100 [2025-12-23 232434.749] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Stream #00 Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s [2025-12-23 232434.749] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: Metadata: [2025-12-23 232434.750] DEBUG default: [conn.go:598] [main.(*RoomClient).startSpeakerPipeline.func1] ffmpeg speaker: encoder : Lavc61.19.101 pcm_s16le 2025/12/23 232436 pion.ice: "level"=0 "msg"="failed to get server reflexive address udp4 turnsosingapore1b.turn.livekit.cloud443?transport=tcp: read udp4 0.0.0.049969 i/o timeout" 2025/12/23 232436 pion.ice: "level"=0 "msg"="failed to get server reflexive address udp4 turnsosingapore1b.turn.livekit.cloud443?transport=tcp: read udp4 0.0.0.035005 i/o timeout" 2025/12/23 232436 pion.ice: "level"=0 "msg"="failed to get server reflexive address udp4 turnsturbo zb3os80m.turn.livekit.cloud443?transport=tcp: read udp4 0.0.0.055639 i/o timeout" 2025/12/23 232436 pion.ice: "level"=0 "msg"="failed to get server reflexive address udp4 turnsturbo zb3os80m.turn.livekit.cloud443?transport=tcp: read udp4 0.0.0.051787 i/o timeout" [2025-12-23 232436.939] INFO default: [conn.go:296] [main.(*RoomClient).bridgeVideoTrack] Bridged 2000 RTP packets [2025-12-23 232439.536] INFO default: [conn.go:296] [main.(*RoomClient).bridgeVideoTrack] Bridged 3000 RTP packets [2025-12-23 232440.022] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #100 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AMQ6R8EHBkKHxC, size: 134 bytes) [2025-12-23 232441.660] DEBUG default: [conn.go:471] [main.(*RoomClient).runFFmpegAudioPipeline.func2] Audio bridged 500 samples [2025-12-23 232442.006] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #200 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AMQ6R8EHBkKHxC, size: 142 bytes) [2025-12-23 232442.242] INFO default: [conn.go:296] [main.(*RoomClient).bridgeVideoTrack] Bridged 4000 RTP packets 2025/12/23 232442 pion.pc: "level"=0 "msg"="signaling state changed to have-remote-offer" 2025/12/23 232442 pion.pc: "level"=0 "msg"="signaling state changed to stable" 2025/12/23 232442 "level"=0 "msg"="track subscribed" "participant"="agent-AJ_BjGgBFTpGeTH" "trackID"="TR_AFbdKZAn79gq6" "kind"="audio" [2025-12-23 232442.749] INFO default: [conn.go:128] [main.(*RoomClient).Connect.func1] Track subscribed: audio from agent-AJ_BjGgBFTpGeTH [2025-12-23 232442.749] INFO default: [conn.go:629] [main.(*RoomClient).playRemoteAudio] Starting audio playback from agent-AJ_BjGgBFTpGeTH (codec: audio/opus, trackID: TR_AFbdKZAn79gq6) [2025-12-23 232442.749] INFO default: [conn.go:634] [main.(*RoomClient).playRemoteAudio] Stopping previous audio playback for new track [2025-12-23 232442.749] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #0 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AFbdKZAn79gq6, size: 67 bytes) [2025-12-23 232442.749] INFO default: [conn.go:711] [main.(*RoomClient).playRemoteAudio] Continuing RTP stream: output seq=236, input seq=25460, time gap=3ms (960 samples) [2025-12-23 232442.767] INFO default: [conn.go:662] [main.(*RoomClient).playRemoteAudio] Audio playback stopped by new track (was playing from agent-AJ_BjGgBFTpGeTH) [2025-12-23 232442.771] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #1 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AFbdKZAn79gq6, size: 67 bytes) [2025-12-23 232442.792] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #2 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AFbdKZAn79gq6, size: 67 bytes) [2025-12-23 232442.805] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #3 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AFbdKZAn79gq6, size: 67 bytes) [2025-12-23 232442.826] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #4 from agent-AJ_BjGgBFTpGeTH (trackID[2025-12-23 232451.675] DEBUG default: [conn.go:471] [main.(*RoomClient).runFFmpegAudioPipeline.func2] Audio bridged 1000 samples [2025-12-23 232452.938] INFO default: [conn.go:296] [main.(*RoomClient).bridgeVideoTrack] Bridged 8000 RTP packets [2025-12-23 232503.751] DEBUG default: [conn.go:686] [main.(*RoomClient).playRemoteAudio] Speaker: received packet #100 from agent-AJ_BjGgBFTpGeTH (trackID: TR_AFbdKZAn79gq6, size: 13 bytes) [2025-12-23 232507.687] DEBUG default: [conn.go:471] [main.(*RoomClient).runFFmpegAudioPipeline.func2] Audio bridged 1500 samples 2025/12/23 232507 pion.ice: "level"=0 "msg"="Discard success response from (udp4 host 143.223.95.38:50007 (resolved: 143.223.95.38:50007)), unknown TransactionID 0x87d2cf7d5a3e0cf8e375e3ca" 2025/12/23 232507 pion.ice: "level"=0 "msg"="Discard success response from (udp4 host 143.223.95.38:50007 (resolved: 143.223.95.38:50007)), unknown TransactionID 0xe88e9e059664406bce184e91" 2025/12/23 232507 pion.ice: "level"=0 "msg"="Discard success response from (udp4 host 143.223.95.38:50010 (resolved: 143.223.95.38:50010)), unknown TransactionID 0x9b6ee8bc8a519e86812179d7" 2025/12/23 232509 pion.ice: "level"=0 "msg"="Discard success response from (udp4 host 143.223.95.38:50007 (resolved: 143.223.95.38:50007)), unknown TransactionID 0xc3cc1c4cf88dc22c0b192ddc" [2025-12-23 232509.550] INFO default: [conn.go:296] [main.(*RoomClient).bridgeVideoTrack] Bridged 9000 RTP packets

Thanks — I’m looking at the logs you pasted and responding to your goal: why the room-client mic is heard by other clients but not recognized by the Agent’s STT. What the logs show (relevant facts) • The speaker pipeline decodes an Opus stream at 48 kHz, stereo: “Audio: opus, 48000 Hz, stereo, fltp”. • Speaker playback packets from the Agent are arriving and being bridged; the Agent → client audio path is working. • Track subscription events occur: “track subscribed” for the Agent audio track. • FFmpeg bridged audio frames show audio is being bridged on the room-client side (500 / 1000 / 1500 samples logged). • ICE shows some TURN timeouts/IO timeouts but they don’t prevent media bridging in these logs. Most likely causes given those facts • Channel/format mismatch: the room-client audio is stereo Opus at 48 kHz. STT/VAD pipelines typically expect mono (and often a specific sample rate). If the Agent isn’t downmixing/remixing before STT, VAD may not detect speech. • Subscription/timing gap for Agent input: if the client published audio before the Agent’s RoomIO/STT pipeline was subscribed/started, early frames can be missed (preconnect buffer issue). • Agent STT pipeline not receiving the bridged frames: logs show bridging on the client side but not evidence of those frames entering the Agent’s STT (no

user_input_transcribed

events shown). Recommended immediate checks and actions (quick, actionable) 1. Confirm whether the Agent’s STT receives frames: ◦ Tail Agent logs for STT/VAD events while you speak from the room-client (look for

user_input_transcribed

, VAD start/stop, or “received frame” lines). 1. Force mono downmix on the Agent side (or in the room-client) and retest: ◦ Ensure audio fed to STT is mono 16-bit PCM at the expected sample rate. If you can’t change the client, add a remix/resample step in the Agent pipeline. 1. Start the Agent first, then join the room-client and speak: ◦ This verifies whether subscription/timing is the issue (if it works when Agent is already started, preconnect is likely the problem). 1. Confirm RoomIO/subscribe config: ◦ Share how you instantiate RoomIO (snippet showing

participant

or

AudioInputOptions

) and whether

pre_connect_audio

is enabled. 1. Capture a short raw audio sample that the Agent receives: ◦ If possible, log or dump a few seconds of the audio frames the Agent sees and attach them (mono vs stereo will be obvious). What to send me next so I can pinpoint it • Agent-side logs around the same timeframe showing STT/VAD events (search for

user_input_transcribed

, VAD start/stop, “silence” or STT errors). • The RoomIO or session startup code (how RoomIO is created,

AudioInputOptions

,

participant

used). • If available, a 5–10s audio dump of what the Agent receives (or ffmpeg input used by Agent). If you prefer, I can show the exact config snippet to downmix/resample in the Agent pipeline and a minimal RoomIO example to force subscription. Which would you like first?

근데 이제 녹음해봤을때 에이전트 목소리는 들린데 걔도 사실 내 목소리를 받아서 stt 까진 잘 진행했어 그러면 뭐가 문제일까

요약 확인합니다 — 녹음했을 때는 에이전트 목소리(Agent → client)도 들리고, 에이전트가 당신 목소리를 받아 STT까지는 정상적으로 처리한 상태입니다. 그럼에도 불구하고 "실시간 대화에서 인식/응답이 안 된다"는 문제만 남은 거죠? 몇 가지 가능성과 빠른 진단 질문 드립니다. 가능한 원인 (우선점 순) • 타이밍/버퍼: 녹음(후처리)에서는 STT 이벤트가 발생하지만, 실시간에서는 음성이 에이전트의 VAD가 활성화된 순간과 맞지 않아 일부 프레임이 무시될 수 있습니다. • 출력→입력 루프(에코) 필터링: 에이전트가 자기 출력을 VAD에서 제거하도록 에코/루프 제거가 동작하면, 방음 경로에서 들어오는 본인 음성(또는 덜 분리된 음성)을 차단할 수 있습니다. • 채널/포맷 불일치의 간헐적 영향: 녹음 시엔 ffmpeg가 재생/리샘플링을 통해 정상화하지만, 실시간 파이프라인에서 일부 프레임은 여전히 스테레오/샘플레이트 문제로 VAD/STT에 도달하지 않을 수 있습니다. • 우선순위/큐잉 지연: STT가 기록 전송/응답 처리 후에만 트리거되고, 실시간 루프에서 응답 타이밍이 어긋나면 짧거나 빈 응답이 발생합니다. 즉시 확인해주실 항목 (한두 줄로 가능한 것부터) 1) 실시간으로 말할 때 에이전트 로그에

user_input_transcribed

이벤트가 꾸준히 뜨나요, 아니면 녹음에서만 뜹니까? (로그 샘플 한두 줄 붙여주세요.) 2) 실시간으로 말했을 때 에이전트가 짧게 "다시 말해주세요" 같은 반응을 하나요, 아니면 아예 무응답인가요? 3) 녹음(성공한 케이스)과 실시간(문제 있는 케이스)의 오디오 소스/설정(스테레오 여부, 샘플레이트, ffmpeg 파라미터)이 동일한가요? 4) Agent가 에코 제거/스피커-미크로 필터를 적용 중인가요? 설정이 있으면 알려주세요. 다음 조치 제안 (하나만 선택해 시도해 주세요) • 로그 확인 우선: 실시간 말하기 중 Agent 로그에서 VAD 시작/종료와

user_input_transcribed

타임스탬프를 캡처해서 보내주세요. • 임시로 에코 제거/루프 필터 끄기(가능하면) — 그 상태에서 실시간 테스트해 보세요. • 실시간 파이프라인에서 mono로 강제 리샘플/리믹스 적용 후 재시험. 원하시면 제가 먼저 어떤 로그 라인을 보어야 할지 예시 포맷(붙여넣기용) 보내드릴게요. 어떤 걸 먼저 해볼까요?