wip introduce a performance monitor in-app

The Kyutai open-source model turned out not to be production-ready:
it caused disruptions in the production environment, especially on
long-running meeting sessions.

Switch to the Voxtral realtime model, which looks like a
credible competitor and behaves much better in our setup.

For now, the code handling the Voxtral realtime API lives directly
in the project. It could be extracted into an open-source package
later.

See PR #1277 for the full details of the implementation, proposed
by @cameldev.

CHANGELOG.md

@@ -8,6 +8,10 @@ and this project adheres to
 
 ## [Unreleased]
 
+### Added
+
+- ✨(agent) support Voxtral realtime as inference engine
+
 ### Changed
 
 - 🔥(backend) remove the S3 storage-event webhook for recordings

compose.yml

@@ -213,6 +213,8 @@ services:
       - "7880:7880"
       - "7881:7881"
       - "7882:7882/udp"
+      - "3478:3478/udp"
+      - "30000-30100:30000-30100/udp"
     volumes:
       - ./docker/livekit/config/livekit-server.yaml:/config.yaml
     depends_on:
@@ -251,6 +253,7 @@ services:
     build:
       context: ./src/agents
       target: development
+    command: ["python", "multi_user_transcriber.py", "dev"]
     env_file:
       - env.d/development/multi_user_transcriber
     volumes:

docker/livekit/config/livekit-server.yaml

@@ -8,3 +8,16 @@ webhook:
   api_key: devkey
   urls:
       - http://app-dev:8000/api/v1.0/rooms/webhooks-livekit/
+
+turn:
+  enabled: true
+  domain: turn.192.168.1.50.nip.io
+  udp_port: 3478
+  tls_port: 0
+  external_tls: false
+  relay_range_start: 30000
+  relay_range_end: 30100
+  allow_restricted_peer_cidrs:
+    - 192.168.0.0/16
+    - 172.16.0.0/12
+

env.d/development/multi_user_transcriber.dist

@@ -1,14 +1,23 @@
+AWS_S3_ENDPOINT_URL=minio:9000
+AWS_S3_ACCESS_KEY_ID=meet
+AWS_S3_SECRET_ACCESS_KEY=password
+
 LIVEKIT_URL=ws://livekit:7880
 LIVEKIT_API_KEY=devkey
 LIVEKIT_API_SECRET=secret
 
-STT_PROVIDER=kyutai # kyutai, deepgram
+STT_PROVIDER=voxtral-vllm # voxtral-vllm, kyutai, deepgram
 ENABLE_SILERO_VAD=False
 
-DEEPGRAM_API_KEY=
+DEEPGRAM_API_KEY=your-deepgram-api-key
 
-KYUTAI_STT_BASE_URL=
+KYUTAI_STT_BASE_URL=url
-KYUTAI_API_KEY=
+KYUTAI_API_KEY=your-kyutai-api-key
+
+VOXTRAL_VLLM_BASE_URL=wss://<host>/v1/realtime
+VOXTRAL_VLLM_MODEL=voxtral-mini-4b-realtime-2602
+VOXTRAL_VLLM_API_KEY=your-vllm-api-key
+VOXTRAL_VLLM_TARGET_STREAMING_DELAY_MS=480
 
 SENTRY_DSN=
 SENTRY_ENVIRONMENT=

src/agents/multi_user_transcriber.py

@@ -1,6 +1,7 @@
 """Multi user transcription agent."""
 
 import asyncio
+import contextlib
 import logging
 import os
 
@@ -25,6 +26,7 @@ from livekit.agents import (
 )
 from livekit.plugins import deepgram, silero
 
+import voxtral_vllm_stt
 from observability import configure_sentry, set_job_context
 from tasks import done_callback
 
@@ -36,9 +38,18 @@ TRANSCRIBER_AGENT_NAME = os.getenv("TRANSCRIBER_AGENT_NAME", "multi-user-transcr
 STT_PROVIDER = os.getenv("STT_PROVIDER", "deepgram")
 ENABLE_SILERO_VAD = os.getenv("ENABLE_SILERO_VAD", "true").lower() == "true"
 
+SESSION_DRAIN_TIMEOUT_S = 15.0
 
-def create_stt_provider():
+
-    """Create STT provider based on environment configuration."""
+def create_stt_provider(vad: silero.VAD | None = None):
+    """Create STT provider based on environment configuration.
+
+    Args:
+        vad: Shared, prewarmed VAD instance. Required in practice for
+            voxtral-vllm (no server-side endpointing): if omitted, the plugin
+            loads its own Silero model synchronously on the event loop, once
+            per participant, freezing all active sessions for the duration.
+    """
     if STT_PROVIDER == "deepgram":
         # Note: Not all Deepgram API parameters are supported by the LiveKit plugin
         # detect_language is NOT supported for real-time streaming
@@ -49,6 +60,9 @@ def create_stt_provider():
         )
     elif STT_PROVIDER == "kyutai":
         _stt_instance = kyutai.STT(base_url=os.getenv("KYUTAI_STT_BASE_URL"))
+    elif STT_PROVIDER == "voxtral-vllm":
+        # The plugin resolves base_url / model / api_key from the environment.
+        _stt_instance = voxtral_vllm_stt.STT(vad=vad)
     else:
         raise ValueError(f"Unknown STT_PROVIDER: {STT_PROVIDER}")
 
@@ -58,9 +72,9 @@ def create_stt_provider():
 class Transcriber(Agent):
     """Create a transcription agent for a specific participant."""
 
-    def __init__(self, *, participant_identity: str):
+    def __init__(self, *, participant_identity: str, vad: silero.VAD | None = None):
         """Init transcription agent."""
-        stt = create_stt_provider()
+        stt = create_stt_provider(vad=vad)
 
         super().__init__(
             instructions="not-needed",
@@ -76,6 +90,7 @@ class MultiUserTranscriber:
         """Init multi user transcription agent."""
         self.ctx = ctx
         self._sessions: dict[str, AgentSession] = {}
+        self._starting: dict[str, asyncio.Task] = {}
         self._tasks: set[asyncio.Task] = set()
 
     def start(self):
@@ -96,22 +111,30 @@ class MultiUserTranscriber:
 
     def on_participant_connected(self, participant: rtc.RemoteParticipant):
         """Handle new participant connection by starting transcription session."""
-        if participant.identity in self._sessions:
+        identity = participant.identity
+        if identity in self._sessions or identity in self._starting:
             return
 
-        logger.info(f"starting session for {participant.identity}")
+        logger.info(f"starting session for {identity}")
         task = asyncio.create_task(self._start_session(participant))
+        self._starting[identity] = task
         self._tasks.add(task)
+        task.add_done_callback(lambda t, i=identity: self._starting.pop(i, None))
         task.add_done_callback(
             done_callback(
                 logger,
                 self._tasks,
-                f"start transcription session for {participant.identity}",
+                f"start transcription session for {identity}",
             )
         )
 
     def on_participant_disconnected(self, participant: rtc.RemoteParticipant):
         """Handle participant disconnection by closing transcription session."""
+        if (start_task := self._starting.pop(participant.identity, None)) is not None:
+            logger.info(f"cancelling pending session start for {participant.identity}")
+            start_task.cancel()
+            return
+
         if (session := self._sessions.pop(participant.identity, None)) is None:
             return
 
@@ -127,10 +150,12 @@ class MultiUserTranscriber:
         )
 
     async def _start_session(self, participant: rtc.RemoteParticipant) -> AgentSession:
-        """Create and start transcription session for participant."""
+        """Create and start transcription session for participant.
-        if participant.identity in self._sessions:
-            return self._sessions[participant.identity]
 
+        Deduplication happens synchronously in on_participant_connected via
+        self._starting; by the time this coroutine runs, the identity is
+        already reserved.
+        """
         vad = self.ctx.proc.userdata.get("vad", None)
         session = AgentSession(vad=vad)
         room_io = RoomIO(
@@ -141,18 +166,30 @@ class MultiUserTranscriber:
                 text_input=False, audio_output=False, text_output=True
             ),
         )
-        await room_io.start()
+        try:
-        await session.start(
+            await room_io.start()
-            agent=Transcriber(
+            await session.start(
-                participant_identity=participant.identity,
+                agent=Transcriber(
+                    participant_identity=participant.identity,
+                    vad=vad,
+                )
             )
-        )
+        except BaseException:
+            with contextlib.suppress(Exception):
+                await session.aclose()
+            raise
         self._sessions[participant.identity] = session
         return session
 
     async def _close_session(self, sess: AgentSession) -> None:
         """Close and cleanup transcription session."""
-        await sess.drain()
+        try:
+            await asyncio.wait_for(sess.drain(), timeout=SESSION_DRAIN_TIMEOUT_S)
+        except (TimeoutError, asyncio.TimeoutError):
+            logger.warning(
+                "session drain timed out after %.0fs; forcing close",
+                SESSION_DRAIN_TIMEOUT_S,
+            )
         await sess.aclose()
 
 

src/agents/pyproject.toml

@@ -12,6 +12,8 @@ dependencies = [
     "protobuf==6.33.6",
     "minio==7.2.20",
     "sentry-sdk==2.66.1",
+    "websockets==17.1",
+    "httpx==0.28.1",
 ]
 
 [project.optional-dependencies]

src/agents/voxtral_vllm_stt.py

@@ -0,0 +1,476 @@
+"""LiveKit STT plugin for Voxtral Realtime served via vLLM (/v1/realtime).
+
+vLLM exposes Voxtral Realtime over a WebSocket that follows the OpenAI Realtime
+API protocol (not Mistral's proprietary realtime protocol).
+"""
+
+from __future__ import annotations
+
+import asyncio
+import base64
+import json
+import logging
+import os
+import weakref
+from collections import deque
+from dataclasses import dataclass, field
+
+import websockets
+from livekit.agents import (
+    DEFAULT_API_CONNECT_OPTIONS,
+    APIConnectionError,
+    APIConnectOptions,
+    APIStatusError,
+    stt,
+    utils,
+)
+from livekit.agents import (
+    vad as vad_module,
+)
+from livekit.agents.types import NOT_GIVEN, NotGivenOr
+from livekit.agents.utils import is_given
+
+logger = logging.getLogger("voxtral-vllm-stt")
+
+SAMPLE_RATE = 16000
+NUM_CHANNELS = 1
+CHUNK_SAMPLES = 1600  # 100 ms @ 16 kHz mono
+PREROLL_CHUNKS = 5  # keep 500 ms of audio before start of speech as detected by VAD
+
+# Reconnect policy: exponential backoff capped at MAX, give up after MAX_ATTEMPTS
+# consecutive failures (a successful handshake resets the counter).
+RECONNECT_BACKOFF_BASE_S = 0.5
+RECONNECT_BACKOFF_MAX_S = 8.0
+RECONNECT_MAX_ATTEMPTS = 5
+
+
+@dataclass
+class _STTOptions:
+    base_url: str
+    model: str
+    api_key: str | None
+    target_streaming_delay_ms: int | None
+
+
+@dataclass
+class _PendingUtterance:
+    """An utterance in flight on the shared websocket used for reconnect.
+
+    `sent_chunks` holds every chunk we have already enqueued for send on this
+    or a prior connection; on reconnect we replay them before resuming reads
+    from `queue`. vLLM concatenates `input_audio_buffer.append` events into a
+    single audio buffer per generation, so duplicates from a partial prior send
+    are harmless.
+    """
+
+    queue: asyncio.Queue[bytes | None]
+    sent_chunks: list[bytes] = field(default_factory=list)
+    ended: bool = False
+
+
+class STT(stt.STT):
+    """LiveKit STT speaking the OpenAI Realtime protocol served by vLLM."""
+
+    def __init__(
+        self,
+        *,
+        base_url: NotGivenOr[str] = NOT_GIVEN,
+        model: NotGivenOr[str] = NOT_GIVEN,
+        api_key: NotGivenOr[str] = NOT_GIVEN,
+        target_streaming_delay_ms: NotGivenOr[int] = NOT_GIVEN,
+        vad: vad_module.VAD | None = None,
+    ) -> None:
+        """Build the STT.
+
+        Args:
+            base_url: WebSocket URL of the vLLM realtime endpoint, e.g.
+                ws://example:8000/v1/realtime. Falls back to $VOXTRAL_VLLM_BASE_URL.
+            model: Model name exposed by vLLM, default
+                mistralai/Voxtral-Mini-4B-Realtime-2602.
+            api_key: Optional bearer token. Falls back to $VOXTRAL_VLLM_API_KEY.
+            target_streaming_delay_ms: Target streaming delay in ms forwarded to
+                vLLM via session.update. Falls back to
+                $VOXTRAL_VLLM_TARGET_STREAMING_DELAY_MS, else server default.
+            vad: Voice Activity Detector. If omitted, Silero VAD is loaded.
+        """
+        super().__init__(
+            capabilities=stt.STTCapabilities(streaming=True, interim_results=True)
+        )
+
+        resolved_url = (
+            base_url
+            if is_given(base_url)
+            else os.environ.get(
+                "VOXTRAL_VLLM_BASE_URL", "ws://127.0.0.1:8000/v1/realtime"
+            )
+        )
+        resolved_model = (
+            model
+            if is_given(model)
+            else os.environ.get(
+                "VOXTRAL_VLLM_MODEL", "mistralai/Voxtral-Mini-4B-Realtime-2602"
+            )
+        )
+        resolved_key = (
+            api_key if is_given(api_key) else os.environ.get("VOXTRAL_VLLM_API_KEY")
+        )
+        resolved_delay = (
+            target_streaming_delay_ms
+            if is_given(target_streaming_delay_ms)
+            else (
+                int(os.environ["VOXTRAL_VLLM_TARGET_STREAMING_DELAY_MS"])
+                if os.environ.get("VOXTRAL_VLLM_TARGET_STREAMING_DELAY_MS")
+                else None
+            )
+        )
+
+        if vad is None:
+            try:
+                from livekit.plugins.silero import VAD as SileroVAD  # noqa: PLC0415
+            except ImportError as exc:
+                raise ImportError(
+                    "livekit-plugins-silero is required for vLLM Voxtral realtime "
+                    "(no server-side endpointing)."
+                ) from exc
+            vad = SileroVAD.load()
+        self._vad = vad
+
+        self._opts = _STTOptions(
+            base_url=resolved_url,
+            model=resolved_model,
+            api_key=resolved_key,
+            target_streaming_delay_ms=resolved_delay,
+        )
+        self._streams: weakref.WeakSet[SpeechStream] = weakref.WeakSet()
+
+    @property
+    def model(self) -> str:
+        """Return the configured vLLM model name."""
+        return self._opts.model
+
+    @property
+    def provider(self) -> str:
+        """Return the provider identifier."""
+        return "vllm-voxtral-realtime"
+
+    async def _recognize_impl(self, *_args, **_kwargs) -> stt.SpeechEvent:
+        raise NotImplementedError(
+            "vLLM Voxtral Realtime STT only supports streaming recognition."
+        )
+
+    def stream(
+        self,
+        *,
+        conn_options: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,
+    ) -> SpeechStream:
+        """Open a new streaming recognition stream."""
+        s = SpeechStream(
+            stt=self,
+            opts=self._opts,
+            vad_instance=self._vad,
+            conn_options=conn_options,
+        )
+        self._streams.add(s)
+        return s
+
+
+class SpeechStream(stt.RecognizeStream):
+    """Voxtral realtime handler."""
+
+    def __init__(
+        self,
+        *,
+        stt: STT,
+        opts: _STTOptions,
+        vad_instance: vad_module.VAD,
+        conn_options: APIConnectOptions,
+    ) -> None:
+        """Init the speech stream."""
+        super().__init__(stt=stt, conn_options=conn_options, sample_rate=SAMPLE_RATE)
+        self._opts = opts
+        self._vad = vad_instance
+        self._utterance_q: asyncio.Queue[bytes | None] | None = None
+        self._speaking = False
+        self._preroll: deque[bytes] = deque(maxlen=PREROLL_CHUNKS)
+        # Voxtral realtime is strictly sequential: only one generation runs at a
+        # time, and a new `commit` is ignored while the previous one is still
+        # producing. We queue per-utterance audio buffers here and let the
+        # pipeline process them one by one on the shared websocket.
+        self._utterance_chan: asyncio.Queue[asyncio.Queue[bytes | None] | None] = (
+            asyncio.Queue()
+        )
+
+    @utils.log_exceptions(logger=logger)
+    async def _run(self) -> None:
+        vad_stream = self._vad.stream()
+
+        bstream = utils.audio.AudioByteStream(
+            sample_rate=SAMPLE_RATE,
+            num_channels=NUM_CHANNELS,
+            samples_per_channel=CHUNK_SAMPLES,
+        )
+
+        async def input_task() -> None:
+            async for data in self._input_ch:
+                if isinstance(data, self._FlushSentinel):
+                    for frame in bstream.flush():
+                        self._handle_chunk(frame.data.tobytes())
+                    continue
+
+                vad_stream.push_frame(data)
+                for frame in bstream.write(data.data.tobytes()):
+                    self._handle_chunk(frame.data.tobytes())
+
+            vad_stream.end_input()
+
+        async def vad_task() -> None:
+            async for ev in vad_stream:
+                if ev.type == vad_module.VADEventType.START_OF_SPEECH:
+                    self._on_start_of_speech()
+                elif ev.type == vad_module.VADEventType.END_OF_SPEECH:
+                    self._on_end_of_speech()
+
+        pipeline_t = asyncio.create_task(self._utterance_pipeline())
+        try:
+            await asyncio.gather(input_task(), vad_task())
+            # signal end-of-stream; pipeline finishes pending utterances first
+            self._utterance_chan.put_nowait(None)
+            await pipeline_t
+        except (APIStatusError, APIConnectionError, asyncio.CancelledError):
+            raise
+        except Exception as exc:
+            logger.exception("vLLM realtime stream failed")
+            raise APIConnectionError() from exc
+        finally:
+            if not pipeline_t.done():
+                pipeline_t.cancel()
+            try:
+                await pipeline_t
+            except asyncio.CancelledError:
+                # CancelledError is the expected flow on cancel()
+                pass
+            except Exception:
+                logger.exception("utterance pipeline failed during finalize")
+            await vad_stream.aclose()
+
+    def _handle_chunk(self, chunk: bytes) -> None:
+        self._preroll.append(chunk)
+        if self._speaking and self._utterance_q is not None:
+            self._utterance_q.put_nowait(chunk)
+
+    def _on_start_of_speech(self) -> None:
+        if self._speaking:
+            return
+        self._speaking = True
+        q: asyncio.Queue[bytes | None] = asyncio.Queue()
+        for chunk in self._preroll:
+            q.put_nowait(chunk)
+        self._utterance_q = q
+        self._utterance_chan.put_nowait(q)
+        self._event_ch.send_nowait(
+            stt.SpeechEvent(type=stt.SpeechEventType.START_OF_SPEECH)
+        )
+
+    def _on_end_of_speech(self) -> None:
+        if not self._speaking:
+            return
+        self._speaking = False
+        if self._utterance_q is not None:
+            self._utterance_q.put_nowait(None)
+        self._utterance_q = None
+        self._event_ch.send_nowait(
+            stt.SpeechEvent(type=stt.SpeechEventType.END_OF_SPEECH)
+        )
+
+    async def _handshake(self, ws: websockets.ClientConnection) -> str:
+        created = json.loads(await ws.recv())
+        if created.get("type") != "session.created":
+            raise APIStatusError(
+                f"expected session.created, got {created}",
+                status_code=500,
+                body=created,
+            )
+        session_update: dict = {"type": "session.update", "model": self._opts.model}
+        if self._opts.target_streaming_delay_ms is not None:
+            session_update["target_streaming_delay_ms"] = (
+                self._opts.target_streaming_delay_ms
+            )
+        await ws.send(json.dumps(session_update))
+        return created.get("id", "")
+
+    def _auth_headers(self) -> dict[str, str]:
+        if self._opts.api_key:
+            return {"Authorization": f"Bearer {self._opts.api_key}"}
+        return {}
+
+    async def _utterance_pipeline(self) -> None:
+        # Owns the websocket lifecycle. On drop, reopens and resumes the
+        # in-flight utterance (if any) by replaying its already-sent chunks.
+        pending: _PendingUtterance | None = None
+        attempt = 0
+        while True:
+            try:
+                async with websockets.connect(
+                    self._opts.base_url,
+                    additional_headers=self._auth_headers(),
+                    open_timeout=self._conn_options.timeout,
+                ) as ws:
+                    request_id = await self._handshake(ws)
+                    attempt = 0
+                    while True:
+                        if pending is None:
+                            q = await self._utterance_chan.get()
+                            if q is None:
+                                return
+                            pending = _PendingUtterance(queue=q)
+                        await self._process_utterance(ws, pending, request_id)
+                        pending = None
+            except (websockets.WebSocketException, OSError, TimeoutError) as exc:
+                attempt += 1
+                if attempt > RECONNECT_MAX_ATTEMPTS:
+                    logger.exception(
+                        "vLLM realtime: giving up after %d reconnect attempts",
+                        RECONNECT_MAX_ATTEMPTS,
+                    )
+                    raise APIConnectionError() from exc
+                backoff = min(
+                    RECONNECT_BACKOFF_BASE_S * (2 ** (attempt - 1)),
+                    RECONNECT_BACKOFF_MAX_S,
+                )
+                if pending is None:
+                    logger.warning(
+                        "vLLM WS connection lost between utterances "
+                        "(attempt %d/%d): %s; retrying in %.1fs",
+                        attempt,
+                        RECONNECT_MAX_ATTEMPTS,
+                        exc,
+                        backoff,
+                    )
+                else:
+                    logger.warning(
+                        "vLLM WS dropped mid-utterance (%d chunks buffered, "
+                        "ended=%s, attempt %d/%d): %s; retrying in %.1fs",
+                        len(pending.sent_chunks),
+                        pending.ended,
+                        attempt,
+                        RECONNECT_MAX_ATTEMPTS,
+                        exc,
+                        backoff,
+                    )
+                await asyncio.sleep(backoff)
+
+    async def _process_utterance(
+        self,
+        ws: websockets.ClientConnection,
+        pending: _PendingUtterance,
+        request_id: str,
+    ) -> None:
+        # Start a fresh generation. Safe to send here: the previous utterance's
+        # transcription.done has already been received (we await it below), so
+        # the server-side generation_task is done and won't ignore this commit.
+        await ws.send(json.dumps({"type": "input_audio_buffer.commit"}))
+        send_t = asyncio.create_task(self._send_audio(ws, pending))
+        try:
+            await self._receive_one_transcription(ws, request_id)
+        finally:
+            if not send_t.done():
+                send_t.cancel()
+            try:
+                await send_t
+            except (asyncio.CancelledError, websockets.WebSocketException):
+                pass
+            except Exception:
+                logger.exception("send-audio task failed during finalize")
+
+    @staticmethod
+    async def _send_audio(
+        ws: websockets.ClientConnection, pending: _PendingUtterance
+    ) -> None:
+        # Replay anything already sent on a previous (now-dead) connection.
+        # sent_chunks is appended before send, so a chunk that failed to send
+        # last time is still present and gets retried here.
+        for chunk in pending.sent_chunks:
+            await ws.send(
+                json.dumps(
+                    {
+                        "type": "input_audio_buffer.append",
+                        "audio": base64.b64encode(chunk).decode("ascii"),
+                    }
+                )
+            )
+        if pending.ended:
+            await ws.send(
+                json.dumps({"type": "input_audio_buffer.commit", "final": True})
+            )
+            return
+        while True:
+            chunk = await pending.queue.get()
+            if chunk is None:
+                pending.ended = True
+                await ws.send(
+                    json.dumps({"type": "input_audio_buffer.commit", "final": True})
+                )
+                return
+            pending.sent_chunks.append(chunk)
+            await ws.send(
+                json.dumps(
+                    {
+                        "type": "input_audio_buffer.append",
+                        "audio": base64.b64encode(chunk).decode("ascii"),
+                    }
+                )
+            )
+
+    async def _receive_one_transcription(
+        self, ws: websockets.ClientConnection, request_id: str
+    ) -> None:
+        # Use recv() rather than `async for`: the latter swallows
+        # ConnectionClosed on close-mid-iteration, which would let a dropped
+        # WS look like a clean "no transcription" return.
+        current_text = ""
+        while True:
+            raw = await ws.recv()
+            data = json.loads(raw)
+            event_type = data.get("type")
+
+            if event_type == "transcription.delta":
+                delta = data.get("delta", "")
+                if not delta:
+                    continue
+                current_text += delta
+                self._event_ch.send_nowait(
+                    stt.SpeechEvent(
+                        type=stt.SpeechEventType.INTERIM_TRANSCRIPT,
+                        request_id=request_id,
+                        alternatives=[stt.SpeechData(text=current_text, language="")],
+                    )
+                )
+            elif event_type == "transcription.done":
+                final_text = data.get("text") or current_text
+                self._event_ch.send_nowait(
+                    stt.SpeechEvent(
+                        type=stt.SpeechEventType.FINAL_TRANSCRIPT,
+                        request_id=request_id,
+                        alternatives=[stt.SpeechData(text=final_text, language="")],
+                    )
+                )
+                usage = data.get("usage") or {}
+                self._event_ch.send_nowait(
+                    stt.SpeechEvent(
+                        type=stt.SpeechEventType.RECOGNITION_USAGE,
+                        request_id=request_id,
+                        recognition_usage=stt.RecognitionUsage(
+                            audio_duration=float(
+                                usage.get("audio_seconds")
+                                or usage.get("prompt_audio_seconds")
+                                or 0
+                            ),
+                            input_tokens=int(usage.get("prompt_tokens") or 0),
+                            output_tokens=int(usage.get("completion_tokens") or 0),
+                        ),
+                    )
+                )
+                return
+            elif event_type == "error":
+                err = data.get("error")
+                raise APIStatusError(str(err), status_code=500, body=data)

src/frontend/src/features/devtools/MeetDevtools.tsx

@@ -0,0 +1,608 @@
+import { CSSProperties, useState } from 'react'
+import { useConnectionState, useRoomContext } from '@livekit/components-react'
+import { ConnectionState } from 'livekit-client'
+import { StatsSnapshot, TrackRow, useWebRTCStats } from './useWebRTCStats'
+import { readRoomConfig } from './roomConfig'
+import {
+  forceTransport,
+  releaseForcedTransport,
+  Scenario,
+  SCENARIOS,
+  setDownlinkCap,
+  setUplinkCap,
+  stepBitrate,
+  TransportMode,
+  transportModeFromRoute,
+} from './simulation'
+
+const SANS =
+  "-apple-system, BlinkMacSystemFont, 'Segoe UI', Roboto, Helvetica, Arial, sans-serif"
+const MONO = 'ui-monospace, SFMono-Regular, Menlo, Consolas, monospace'
+
+const COLOR = {
+  text: '#e8e8e8',
+  muted: '#9a9a9a',
+  faint: '#6f6f6f',
+  hairline: '#2c2d31',
+  border: '#3d3e44',
+  surface: '#1b1c1e',
+  chartBg: '#141517',
+  down: '#f6821f',
+  up: '#5a9cf8',
+  ok: '#10b981',
+  bad: '#f05a4a',
+  busy: '#e5a13c',
+}
+
+const stateColor = (state: ConnectionState): string => {
+  switch (state) {
+    case ConnectionState.Connected:
+      return COLOR.ok
+    case ConnectionState.Reconnecting:
+    case ConnectionState.SignalReconnecting:
+      return COLOR.busy
+    case ConnectionState.Connecting:
+      return COLOR.up
+    case ConnectionState.Disconnected:
+      return COLOR.bad
+  }
+}
+
+const styles: Record<string, CSSProperties> = {
+  toggle: {
+    position: 'fixed',
+    bottom: 12,
+    right: 12,
+    zIndex: 9999,
+    fontFamily: SANS,
+    fontSize: 12,
+    lineHeight: 1,
+    padding: '8px 12px',
+    borderRadius: 999,
+    borderWidth: 1,
+    borderStyle: 'solid',
+    borderColor: COLOR.border,
+    background: COLOR.surface,
+    color: COLOR.text,
+    cursor: 'pointer',
+    boxShadow: '0 2px 10px #0006',
+  },
+  panel: {
+    position: 'fixed',
+    bottom: 12,
+    right: 12,
+    zIndex: 9999,
+    width: 460,
+    maxWidth: 'calc(100vw - 24px)',
+    maxHeight: 'calc(100vh - 24px)',
+    overflowY: 'auto',
+    fontFamily: SANS,
+    fontSize: 11,
+    lineHeight: 1.5,
+    color: COLOR.text,
+    background: COLOR.surface,
+    borderWidth: 1,
+    borderStyle: 'solid',
+    borderColor: COLOR.border,
+    borderRadius: 10,
+    padding: 14,
+    boxShadow: '0 10px 34px #00000080',
+  },
+  sectionTitle: {
+    margin: '14px 0 5px',
+    color: COLOR.faint,
+    textTransform: 'uppercase',
+    letterSpacing: 1.2,
+    fontSize: 9,
+    fontWeight: 600,
+    display: 'flex',
+    justifyContent: 'space-between',
+  },
+  metricsLine: {
+    display: 'flex',
+    alignItems: 'center',
+    gap: 16,
+    margin: '0 0 10px',
+    paddingBottom: 10,
+    borderBottom: `1px solid ${COLOR.hairline}`,
+    fontVariantNumeric: 'tabular-nums',
+  },
+  legend: {
+    display: 'flex',
+    gap: 12,
+    marginTop: 3,
+    color: COLOR.faint,
+    fontSize: 10,
+  },
+  list: { display: 'flex', flexDirection: 'column', maxHeight: '200px', overflowY: 'auto' },
+  listRow: { display: 'flex', gap: 8, padding: '1px 0' },
+  listLabel: { flex: '0 0 34%', color: COLOR.muted },
+  trackRow: {
+    display: 'flex',
+    gap: 8,
+    padding: '3px 0',
+    borderBottom: `1px solid ${COLOR.hairline}`,
+    alignItems: 'baseline',
+  },
+  trackLabel: {
+    flex: '0 0 34%',
+    whiteSpace: 'nowrap',
+    overflow: 'hidden',
+    textOverflow: 'ellipsis',
+  },
+  trackCodec: {
+    flex: '0 0 11%',
+    fontFamily: MONO,
+    fontSize: 10,
+    color: COLOR.muted,
+  },
+  trackKbps: {
+    flex: '0 0 9%',
+    textAlign: 'right',
+    fontVariantNumeric: 'tabular-nums',
+  },
+  trackDetail: { flex: 1, minWidth: 0 },
+  mono: { fontFamily: MONO, fontSize: 10, color: COLOR.muted },
+  button: {
+    fontFamily: SANS,
+    fontSize: 11,
+    padding: '3px 9px',
+    borderRadius: 5,
+    borderWidth: 1,
+    borderStyle: 'solid',
+    borderColor: COLOR.border,
+    background: '#232428',
+    color: COLOR.text,
+    cursor: 'pointer',
+  },
+  buttonActive: { borderColor: COLOR.down, color: COLOR.down },
+  buttonDisabled: { color: COLOR.faint, cursor: 'not-allowed' },
+  row: { display: 'flex', alignItems: 'center', gap: 6, flexWrap: 'wrap' },
+  stepperValue: {
+    minWidth: 92,
+    textAlign: 'center',
+    borderWidth: 1,
+    borderStyle: 'solid',
+    borderColor: COLOR.hairline,
+    borderRadius: 4,
+    padding: '2px 6px',
+    background: COLOR.chartBg,
+    fontVariantNumeric: 'tabular-nums',
+  },
+  close: {
+    background: 'none',
+    border: 'none',
+    color: COLOR.faint,
+    cursor: 'pointer',
+    fontSize: 14,
+  },
+}
+
+const Sparkline = ({ history }: { history: StatsSnapshot[] }) => {
+  const width = 430
+  const height = 54
+  const max = Math.max(...history.map((s) => Math.max(s.upKbps, s.downKbps)), 1)
+  const points = (pick: (s: StatsSnapshot) => number) =>
+    history
+      .map(
+        (s, i) =>
+          `${((i / Math.max(history.length - 1, 1)) * width).toFixed(1)},${(
+            height -
+            (pick(s) / max) * (height - 6) -
+            3
+          ).toFixed(1)}`
+      )
+      .join(' ')
+  return (
+    <>
+      <svg
+        width={width}
+        height={height}
+        role="img"
+        aria-label="Bandwidth over the last minute"
+        style={{
+          display: 'block',
+          background: COLOR.chartBg,
+          borderRadius: 6,
+          border: `1px solid ${COLOR.hairline}`,
+        }}
+      >
+        {history.length >= 2 && (
+          <>
+            <polyline
+              points={points((s) => s.downKbps)}
+              fill="none"
+              stroke={COLOR.down}
+              strokeWidth="1.5"
+            />
+            <polyline
+              points={points((s) => s.upKbps)}
+              fill="none"
+              stroke={COLOR.up}
+              strokeWidth="1.5"
+            />
+          </>
+        )}
+      </svg>
+      {/* Legend outside the plot: text over the lines was unreadable. */}
+      <div style={styles.legend}>
+        <span>
+          <span style={{ color: COLOR.down }}>●</span> down
+        </span>
+        <span>
+          <span style={{ color: COLOR.up }}>●</span> up
+        </span>
+        <span>last 60s · max {Math.round(max)} kbps</span>
+      </div>
+    </>
+  )
+}
+
+const TrackDetail = ({ track }: { track: TrackRow }) => {
+  const parts: Array<{ text: string; tone?: 'warn' | 'muted' }> = []
+  if (track.res) parts.push({ text: track.res })
+  if (track.fps !== undefined) {
+    parts.push({ text: `${Math.round(track.fps)}fps` })
+  }
+  if (track.lossPct !== undefined && track.lossPct > 0) {
+    parts.push({ text: `loss ${track.lossPct.toFixed(1)}%`, tone: 'warn' })
+  }
+  if (track.froze) parts.push({ text: 'freeze', tone: 'warn' })
+  if (track.limitation) {
+    parts.push({ text: `lim:${track.limitation}`, tone: 'warn' })
+  }
+  const layer = track.layer
+  if (layer) {
+    if (layer.availRes && layer.availRes !== track.res) {
+      parts.push({
+        text: `≤${layer.availRes}${layer.layerCount ? `(${layer.layerCount}L)` : ''}`,
+        tone: 'muted',
+      })
+    }
+    if (layer.reason === 'adaptive' && layer.elementRes) {
+      parts.push({ text: `fit:${layer.elementRes}`, tone: 'muted' })
+    } else if (layer.reason === 'bandwidth') {
+      parts.push({ text: 'bw-limited', tone: 'warn' })
+    } else if (layer.reason === 'paused') {
+      parts.push({ text: 'paused:SFU', tone: 'warn' })
+    } else if (layer.reason === 'off-screen') {
+      parts.push({ text: 'off-screen', tone: 'muted' })
+    }
+  }
+  return (
+    <>
+      {parts.map((part, i) => (
+        <span
+          key={part.text}
+          style={
+            part.tone === 'warn'
+              ? { color: COLOR.bad }
+              : part.tone === 'muted'
+                ? { color: COLOR.faint }
+                : {}
+          }
+        >
+          {i > 0 && ' · '}
+          {part.text}
+        </span>
+      ))}
+    </>
+  )
+}
+
+const CapStepper = ({
+  label,
+  hint,
+  valueKbps,
+  onChange,
+}: {
+  label: string
+  hint: string
+  valueKbps: number | null
+  onChange: (kbps: number | null) => void
+}) => (
+  <div style={styles.row}>
+    <span style={{ flex: '0 0 70px' }}>{label}</span>
+    <button
+      type="button"
+      style={styles.button}
+      aria-label={`Decrease ${label} bandwidth`}
+      onClick={() => onChange(stepBitrate(valueKbps, 'decrease'))}
+    >
+      −
+    </button>
+    <span
+      style={{
+        ...styles.stepperValue,
+        ...(valueKbps !== null ? { color: COLOR.down } : {}),
+      }}
+    >
+      {valueKbps === null ? 'unlimited' : `≤ ${valueKbps} kbps`}
+    </span>
+    <button
+      type="button"
+      style={styles.button}
+      aria-label={`Increase ${label} bandwidth`}
+      onClick={() => onChange(stepBitrate(valueKbps, 'increase'))}
+    >
+      +
+    </button>
+    <span style={styles.mono}>{hint}</span>
+  </div>
+)
+
+const TrackList = ({ title, rows }: { title: string; rows: TrackRow[] }) => (
+  <>
+    <div style={styles.sectionTitle}>
+      <span>
+        {title} ({rows.length})
+      </span>
+      <span>
+        {Math.round(rows.reduce((sum, t) => sum + t.kbps, 0))} kbps media
+      </span>
+    </div>
+    <div style={styles.list}>
+      {rows.map((track) => (
+        <div key={track.key} style={styles.trackRow}>
+          <span style={styles.trackLabel}>{track.label}</span>
+          <span style={styles.trackCodec}>{track.codec ?? '–'}</span>
+          <span style={styles.trackKbps}>{Math.round(track.kbps)}</span>
+          <span style={styles.trackDetail}>
+            <TrackDetail track={track} />
+          </span>
+        </div>
+      ))}
+      {rows.length === 0 && <span style={{ color: COLOR.faint }}>none</span>}
+    </div>
+  </>
+)
+
+const MeetDevtools = () => {
+  const room = useRoomContext()
+  const connState = useConnectionState(room)
+  const [open, setOpen] = useState(false)
+  const [firedScenario, setFiredScenario] = useState<string>()
+  const [uplinkCap, setUplinkCapState] = useState<number | null>(null)
+  const [downlinkCap, setDownlinkCapState] = useState<number | null>(null)
+  const { snapshot, history } = useWebRTCStats(room, open)
+
+  const transportMode: TransportMode = transportModeFromRoute(snapshot?.route)
+
+  const fireScenario = (scenario: Scenario) => {
+    setFiredScenario(scenario.id)
+    scenario.run(room).catch((e) => {
+      console.warn('[MeetDevtools] simulateScenario failed', e)
+    })
+    window.setTimeout(
+      () =>
+        setFiredScenario((current) =>
+          current === scenario.id ? undefined : current
+        ),
+      1200
+    )
+  }
+
+  if (!open) {
+    return (
+      <button
+        type="button"
+        style={styles.toggle}
+        onClick={() => setOpen(true)}
+        aria-label="Open WebRTC devtools"
+      >
+        <span style={{ color: stateColor(connState) }}>●</span> rtc
+      </button>
+    )
+  }
+
+  const config = readRoomConfig(room)
+  const published = snapshot?.tracks.filter((t) => t.dir === 'up') ?? []
+  const subscribed = snapshot?.tracks.filter((t) => t.dir === 'down') ?? []
+  const turnProtocols = snapshot?.turnProtocols ?? []
+  const transports: Array<{
+    mode: TransportMode
+    label: string
+    requires?: string
+    title: string
+  }> = [
+    {
+      mode: 'auto',
+      label: 'auto (udp)',
+      title:
+        'clears the server-cached transport preference and relay-only policy, then full reconnect',
+    },
+    {
+      mode: 'tcp',
+      label: 'tcp',
+      title: 'force-tcp — server prefers TCP candidates (ICE/TCP)',
+    },
+    {
+      mode: 'turn-udp',
+      label: 'turn:udp',
+      requires: 'udp',
+      title:
+        'client-side: iceTransportPolicy relay + full reconnect (no server hook exists)',
+    },
+    {
+      mode: 'turn-tcp',
+      label: 'turn:tcp',
+      requires: 'tcp',
+      title: 'force-tcp — lands on TURN/TCP when it is the TCP path',
+    },
+    {
+      mode: 'turn-tls',
+      label: 'turn:tls',
+      requires: 'tls',
+      title: 'force-tls — server switches you to TURN over TLS',
+    },
+  ]
+
+  return (
+    <section style={styles.panel} aria-label="WebRTC devtools">
+      <div style={styles.metricsLine}>
+        <span>
+          <span style={{ color: COLOR.down }}>↓</span> {snapshot?.downKbps ?? 0}{' '}
+          kbps
+        </span>
+        <span>
+          <span style={{ color: COLOR.up }}>↑</span> {snapshot?.upKbps ?? 0}{' '}
+          kbps
+        </span>
+        <span style={{ color: COLOR.muted }}>
+          rtt{' '}
+          <span style={{ color: COLOR.text }}>{snapshot?.rttMs ?? '–'}</span> ms
+        </span>
+        <span style={{ color: COLOR.muted }}>
+          jitter{' '}
+          <span style={{ color: COLOR.text }}>{snapshot?.jitterMs ?? '–'}</span>{' '}
+          ms
+        </span>
+        <span
+          style={{
+            color: stateColor(connState),
+            fontSize: 10,
+            marginLeft: 'auto',
+          }}
+        >
+          ● {connState}
+        </span>
+        <button
+          type="button"
+          style={styles.close}
+          onClick={() => setOpen(false)}
+          aria-label="Close WebRTC devtools"
+        >
+          ✕
+        </button>
+      </div>
+
+      <Sparkline history={history} />
+
+      <div style={styles.sectionTitle}>room configuration (live)</div>
+      <div style={styles.list}>
+        {config.map((entry) => (
+          <div key={entry.label} style={styles.listRow}>
+            <span
+              style={{
+                color:
+                  entry.on === undefined
+                    ? COLOR.faint
+                    : entry.on
+                      ? COLOR.ok
+                      : COLOR.bad,
+              }}
+            >
+              ●
+            </span>
+            <span style={styles.listLabel}>{entry.label}</span>
+            <span>{entry.value}</span>
+          </div>
+        ))}
+      </div>
+
+      <TrackList title="published ↑" rows={published} />
+      <TrackList title="subscribed ↓" rows={subscribed} />
+
+      <div style={styles.sectionTitle}>bandwidth</div>
+      <div style={{ ...styles.list, gap: 4 }}>
+        <CapStepper
+          label="uplink"
+          hint="encoder cap (setParameters)"
+          valueKbps={uplinkCap}
+          onChange={(kbps) => {
+            setUplinkCapState(kbps)
+            void setUplinkCap(room, kbps)
+          }}
+        />
+        <CapStepper
+          label="downlink"
+          hint="SFU limit (subscriber-bandwidth)"
+          valueKbps={downlinkCap}
+          onChange={(kbps) => {
+            setDownlinkCapState(kbps)
+            void setDownlinkCap(room, kbps).catch((e) =>
+              console.warn('[MeetDevtools] downlink cap failed', e)
+            )
+          }}
+        />
+      </div>
+
+      <div style={styles.sectionTitle}>transport</div>
+      <div style={styles.row}>
+        {transports.map(({ mode, label, requires, title }) => {
+          const available = !requires || turnProtocols.includes(requires)
+          const active = transportMode === mode
+          const clickable = available && !active
+          return (
+            <button
+              key={mode}
+              type="button"
+              aria-pressed={active}
+              disabled={!clickable}
+              title={available ? title : `${title} — not configured`}
+              style={{
+                ...styles.button,
+                ...(clickable || active ? {} : styles.buttonDisabled),
+                ...(active ? styles.buttonActive : {}),
+              }}
+              onClick={() => {
+                if (!clickable) return
+                void (
+                  mode === 'auto'
+                    ? releaseForcedTransport(room)
+                    : forceTransport(
+                        room,
+                        mode as 'tcp' | 'turn-udp' | 'turn-tcp' | 'turn-tls'
+                      )
+                ).catch((e) =>
+                  console.warn('[MeetDevtools] transport change failed', e)
+                )
+              }}
+            >
+              {requires && (
+                <span style={{ color: available ? COLOR.ok : COLOR.faint }}>
+                  ●{' '}
+                </span>
+              )}
+              {label}
+            </button>
+          )
+        })}
+        <span style={styles.mono}>
+          route:{' '}
+          {snapshot?.route
+            ? [
+                snapshot.route.protocol,
+                snapshot.route.type,
+                snapshot.route.relayProtocol &&
+                  `relay:${snapshot.route.relayProtocol}`,
+              ]
+                .filter(Boolean)
+                .join('·')
+            : '–'}
+        </span>
+      </div>
+
+      <div style={styles.sectionTitle}>connection scenarios</div>
+      <div style={styles.row}>
+        {SCENARIOS.map((scenario) => (
+          <button
+            key={scenario.id}
+            type="button"
+            title={`${scenario.side}-side simulation — momentary, watch the state dot`}
+            style={{
+              ...styles.button,
+              ...(firedScenario === scenario.id ? styles.buttonActive : {}),
+            }}
+            onClick={() => fireScenario(scenario)}
+          >
+            {scenario.side === 'server' ? '☁ ' : ''}
+            {scenario.label}
+          </button>
+        ))}
+      </div>
+    </section>
+  )
+}
+
+export default MeetDevtools

src/frontend/src/features/devtools/index.tsx

@@ -0,0 +1,15 @@
+import { lazy, Suspense } from 'react'
+
+
+const LazyPanel = import.meta.env.DEV
+  ? lazy(() => import('./MeetDevtools'))
+  : null
+
+export const MeetDevtools = () => {
+  if (!LazyPanel) return null
+  return (
+    <Suspense fallback={null}>
+      <LazyPanel />
+    </Suspense>
+  )
+}

src/frontend/src/features/devtools/roomConfig.ts

@@ -0,0 +1,89 @@
+import { Room } from 'livekit-client'
+
+
+export type ConfigEntry = {
+  label: string
+  value: string
+  /** true = feature actively on, false = off, undefined = informational */
+  on?: boolean
+}
+
+const formatBackupCodec = (backup: unknown): string => {
+  if (backup === undefined || backup === true) return 'auto'
+  if (backup === false) return 'off'
+  if (typeof backup === 'object' && backup !== null && 'codec' in backup) {
+    return String((backup as { codec: unknown }).codec)
+  }
+  return String(backup)
+}
+
+export const readRoomConfig = (room: Room): ConfigEntry[] => {
+  const options = room.options
+  const publish = options.publishDefaults
+  const adaptive = options.adaptiveStream
+
+  const entries: ConfigEntry[] = [
+    {
+      label: 'adaptiveStream',
+      value:
+        typeof adaptive === 'object'
+          ? JSON.stringify(adaptive)
+          : String(!!adaptive),
+      on: !!adaptive,
+    },
+    {
+      label: 'dynacast',
+      value: String(!!options.dynacast),
+      on: !!options.dynacast,
+    },
+    {
+      label: 'e2ee',
+      value: String(room.isE2EEEnabled),
+      on: room.isE2EEEnabled,
+    },
+    {
+      label: 'videoCodec',
+      value: publish?.videoCodec ?? 'default',
+    },
+    {
+      label: 'backupCodec',
+      value: formatBackupCodec(publish?.backupCodec),
+    },
+    {
+      label: 'simulcast',
+      value: String(publish?.simulcast ?? true),
+      on: publish?.simulcast ?? true,
+    },
+    {
+      label: 'audio dtx',
+      value: String(publish?.dtx ?? true),
+      on: publish?.dtx ?? true,
+    },
+    {
+      label: 'audio red',
+      value: String(publish?.red ?? true),
+      on: publish?.red ?? true,
+    },
+    {
+      label: 'quality (local)',
+      value: room.localParticipant.connectionQuality,
+    },
+  ]
+
+  const server = room.serverInfo
+  if (server) {
+    entries.push({
+      label: 'server',
+      value: [
+        server.version && `v${server.version}`,
+        server.region,
+        server.protocol !== undefined && `proto ${server.protocol}`,
+        server.edition !== undefined && `edition ${server.edition}`,
+      ]
+        .filter(Boolean)
+        .join(' · '),
+    })
+  }
+
+  return entries
+}

src/frontend/src/features/devtools/simulation.ts

@@ -0,0 +1,193 @@
+import { Room, Track } from 'livekit-client'
+// Transitive dependency of livekit-client (pinned by it); used only to
+// build the one signal request room.simulateScenario cannot express.
+import { SimulateScenario } from '@livekit/protocol'
+
+/** Shared bandwidth ladder for the −/+ steppers; null = unlimited. */
+export const BITRATE_LADDER_KBPS: Array<number | null> = [
+  null,
+  2000,
+  1000,
+  600,
+  300,
+  150,
+]
+
+export const stepBitrate = (
+  current: number | null,
+  direction: 'decrease' | 'increase'
+): number | null => {
+  const index = BITRATE_LADDER_KBPS.indexOf(current)
+  const safeIndex = index === -1 ? 0 : index
+  const next =
+    direction === 'decrease'
+      ? Math.min(safeIndex + 1, BITRATE_LADDER_KBPS.length - 1)
+      : Math.max(safeIndex - 1, 0)
+  return BITRATE_LADDER_KBPS[next]
+}
+
+const savedEncodings = new WeakMap<RTCRtpSender, Array<number | undefined>>()
+
+export const setUplinkCap = async (
+  room: Room,
+  kbps: number | null
+): Promise<void> => {
+  const senders: RTCRtpSender[] = []
+  room.localParticipant.trackPublications.forEach((pub) => {
+    const track = pub.track
+    if (track?.kind === Track.Kind.Video && track.sender) {
+      senders.push(track.sender)
+    }
+  })
+
+  for (const sender of senders) {
+    const params = sender.getParameters()
+    if (!params.encodings || params.encodings.length === 0) continue
+
+    if (kbps === null) {
+      const original = savedEncodings.get(sender)
+      params.encodings.forEach((encoding, i) => {
+        encoding.maxBitrate = original?.[i]
+      })
+      savedEncodings.delete(sender)
+    } else {
+      if (!savedEncodings.has(sender)) {
+        savedEncodings.set(
+          sender,
+          params.encodings.map((encoding) => encoding.maxBitrate)
+        )
+      }
+      const activeCount =
+        params.encodings.filter((encoding) => encoding.active !== false)
+          .length || 1
+      // Split the budget across active simulcast layers / SVC encoding.
+      const perEncoding = Math.max(
+        30_000,
+        Math.floor((kbps * 1000) / activeCount)
+      )
+      params.encodings.forEach((encoding) => {
+        encoding.maxBitrate = perEncoding
+      })
+    }
+
+    try {
+      await sender.setParameters(params)
+    } catch (e) {
+      console.warn('[MeetDevtools] setParameters failed', e)
+    }
+  }
+}
+
+
+export const setDownlinkCap = (
+  room: Room,
+  kbps: number | null
+): Promise<void> =>
+  room.simulateScenario('subscriber-bandwidth', kbps === null ? 0 : kbps * 1000)
+
+export type Scenario = {
+  id: string
+  label: string
+  /** where the simulation happens */
+  side: 'client' | 'server'
+  run: (room: Room) => Promise<void>
+}
+
+export const SCENARIOS: Scenario[] = [
+  {
+    id: 'resume',
+    label: 'reconnect (resume)',
+    side: 'client',
+    // Replays a signaling WebSocket loss; media keeps flowing, client resumes.
+    run: (room) => room.simulateScenario('signal-reconnect'),
+  },
+  {
+    id: 'resume-fail',
+    label: 'reconnect (resume fails)',
+    side: 'client',
+    // Same, but the next resume attempt fails → exercises the retry ladder.
+    run: (room) => room.simulateScenario('resume-reconnect'),
+  },
+  {
+    id: 'full-reconnect',
+    label: 'full reconnect',
+    side: 'client',
+    // Complete rejoin with brand-new peer connections.
+    run: (room) => room.simulateScenario('full-reconnect'),
+  },
+  {
+    id: 'migration',
+    label: 'server migration',
+    side: 'server',
+    run: (room) => room.simulateScenario('migration'),
+  },
+  {
+    id: 'node-failure',
+    label: 'SFU node failure',
+    side: 'server',
+    run: (room) => room.simulateScenario('node-failure'),
+  },
+  {
+    id: 'server-leave',
+    label: 'server disconnect',
+    side: 'server',
+    // Server-initiated leave: the closest thing to "you got kicked".
+    run: (room) => room.simulateScenario('server-leave'),
+  },
+]
+
+export type TransportMode =
+  | 'auto'
+  | 'tcp'
+  | 'turn-udp'
+  | 'turn-tcp'
+  | 'turn-tls'
+
+const clearServerTransportPreference = (room: Room): Promise<void> =>
+  room.engine.client.sendSimulateScenario(
+    new SimulateScenario({
+      scenario: { case: 'switchCandidateProtocol', value: 0 },
+    })
+  )
+
+const setRelayOnly = (room: Room, relay: boolean) => {
+  room.engine.rtcConfig = {
+    ...room.engine.rtcConfig,
+    iceTransportPolicy: relay ? 'relay' : 'all',
+  }
+}
+
+const settle = () => new Promise((resolve) => setTimeout(resolve, 300))
+
+export const forceTransport = async (
+  room: Room,
+  mode: 'tcp' | 'turn-udp' | 'turn-tcp' | 'turn-tls'
+): Promise<void> => {
+  if (mode === 'turn-udp') {
+    await clearServerTransportPreference(room)
+    setRelayOnly(room, true)
+    await settle()
+    return room.simulateScenario('full-reconnect')
+  }
+  setRelayOnly(room, false)
+  return room.simulateScenario(mode === 'turn-tls' ? 'force-tls' : 'force-tcp')
+}
+
+export const releaseForcedTransport = async (room: Room): Promise<void> => {
+  setRelayOnly(room, false)
+  await clearServerTransportPreference(room)
+  await settle()
+  await room.simulateScenario('full-reconnect')
+}
+
+export const transportModeFromRoute = (route?: {
+  protocol?: string
+  relayProtocol?: string
+}): TransportMode => {
+  // relayProtocol is the client→TURN leg; protocol alone means no relay.
+  if (route?.relayProtocol === 'tls') return 'turn-tls'
+  if (route?.relayProtocol === 'tcp') return 'turn-tcp'
+  if (route?.relayProtocol === 'udp') return 'turn-udp'
+  if (route?.protocol === 'tcp') return 'tcp'
+  return 'auto'
+}

src/frontend/src/features/devtools/useWebRTCStats.ts

@@ -0,0 +1,357 @@
+import { useEffect, useRef, useState } from 'react'
+import { Room, Track } from 'livekit-client'
+
+
+export type LayerInfo = {
+  /** top published layer, e.g. '1280x720' */
+  availRes?: string
+  /** number of published spatial layers (simulcast/SVC), when known */
+  layerCount?: number
+  /** largest attached element size in device px, e.g. '480x270' */
+  elementRes?: string
+  /** why the forwarded layer is below the top one */
+  reason?: 'adaptive' | 'bandwidth' | 'paused' | 'off-screen'
+}
+
+export type TrackRow = {
+  key: string
+  dir: 'up' | 'down'
+  kind: string
+  label: string
+  codec?: string
+  kbps: number
+  fps?: number
+  res?: string
+  lossPct?: number
+  limitation?: string
+  /** subscribed video only: SFU layer forwarding context */
+  layer?: LayerInfo
+  /** this track froze during the last tick */
+  froze?: boolean
+}
+
+export type StatsSnapshot = {
+  ts: number
+  /** wire totals from transport stats (includes headers, RTCP, FEC) */
+  upKbps: number
+  downKbps: number
+  rttMs?: number
+  /** worst inbound RTP jitter across subscribed tracks */
+  jitterMs?: number
+  availableOutKbps?: number
+  /** selected ICE route of the publisher transport (measured, not assumed) */
+  route?: { protocol?: string; type?: string; relayProtocol?: string }
+  /**
+   * relayProtocol values of gathered relay local candidates — i.e. which
+   * client→TURN transports are actually configured (udp/tcp/tls). Empty
+   * when no TURN server is configured.
+   */
+  turnProtocols: string[]
+  tracks: TrackRow[]
+}
+
+type StatDict = Record<string, unknown>
+type Counters = Record<string, number>
+
+const asNumber = (v: unknown): number | undefined =>
+  typeof v === 'number' && Number.isFinite(v) ? v : undefined
+
+const asString = (v: unknown): string | undefined =>
+  typeof v === 'string' ? v : undefined
+
+const shortCodec = (mimeType?: string) =>
+  mimeType ? mimeType.replace(/^(audio|video)\//, '') : undefined
+
+type TrackContext = { label: string; layer?: LayerInfo; publishedH?: number }
+
+/**
+ * msTrackId -> label + subscribed-layer context, rebuilt on every tick.
+ * The "why is this tile blurry" answer needs SDK state (publication +
+ * element size) joined with getStats (forwarded resolution); this is the
+ * SDK-state half.
+ */
+const buildTrackContext = (room: Room) => {
+  const contexts = new Map<string, TrackContext>()
+  room.localParticipant.trackPublications.forEach((pub) => {
+    const id = pub.track?.mediaStreamTrack?.id
+    if (id) contexts.set(id, { label: `local ${pub.source}` })
+  })
+  const dpr = window.devicePixelRatio || 1
+  room.remoteParticipants.forEach((participant) => {
+    // Keep rows scannable: participant names capped at 10 chars.
+    const rawName = participant.name || participant.identity
+    const name = rawName.length > 15 ? `${rawName.slice(0, 15)}.` : rawName
+    participant.trackPublications.forEach((pub) => {
+      const track = pub.track
+      const id = track?.mediaStreamTrack?.id
+      if (!track || !id) return
+      const label = `${name} ${pub.source}`
+      if (track.kind !== Track.Kind.Video) {
+        contexts.set(id, { label })
+        return
+      }
+      let elementRes: string | undefined
+      let elementH: number | undefined
+      for (const element of track.attachedElements) {
+        const w = Math.round(element.clientWidth * dpr)
+        const h = Math.round(element.clientHeight * dpr)
+        if (elementH === undefined || h > elementH) {
+          elementH = h
+          elementRes = `${w}x${h}`
+        }
+      }
+      const dims = pub.dimensions
+      const layers = pub.trackInfo?.layers?.length
+      contexts.set(id, {
+        label,
+        publishedH: dims?.height,
+        layer: {
+          availRes: dims ? `${dims.width}x${dims.height}` : undefined,
+          layerCount: layers && layers > 1 ? layers : undefined,
+          elementRes,
+          reason: !pub.isEnabled
+            ? 'off-screen'
+            : track.streamState === Track.StreamState.Paused
+              ? 'paused'
+              : undefined,
+        },
+      })
+    })
+  })
+  return contexts
+}
+
+/** Decide why a forwarded layer is below the published top layer. */
+const resolveLayerReason = (
+  context: TrackContext,
+  forwardedH: number | undefined
+): LayerInfo | undefined => {
+  const layer = context.layer
+  if (!layer) return undefined
+  if (layer.reason) return layer // off-screen / paused already decided
+  const publishedH = context.publishedH
+  if (!publishedH || !forwardedH || forwardedH >= publishedH * 0.9) {
+    return { ...layer, reason: undefined } // full quality, nothing to explain
+  }
+  // Below top layer: if the element only needs about what we get, it's
+  // adaptiveStream fitting the element; otherwise the SFU is holding back
+  // a layer the element could use — congestion.
+  const elementH = layer.elementRes
+    ? Number(layer.elementRes.split('x')[1])
+    : undefined
+  const adaptive = elementH !== undefined && forwardedH >= elementH * 0.7
+  return { ...layer, reason: adaptive ? 'adaptive' : 'bandwidth' }
+}
+
+export const useWebRTCStats = (
+  room: Room,
+  enabled: boolean,
+  intervalMs = 1000
+) => {
+  // Single source of truth: the snapshot is just the last history entry.
+  const [history, setHistory] = useState<StatsSnapshot[]>([])
+  const prevRef = useRef(new Map<string, Counters>())
+
+  useEffect(() => {
+    if (!enabled) return
+
+    let cancelled = false
+    const prev = prevRef.current
+
+    /** Per-stat counter deltas; returns 0 on the first sighting. */
+    const deltas = (key: string, now: Counters): Counters => {
+      const before = prev.get(key)
+      prev.set(key, now)
+      const out: Counters = {}
+      for (const [name, value] of Object.entries(now)) {
+        out[name] = before?.[name] !== undefined ? value - before[name] : 0
+      }
+      return out
+    }
+
+    const collect = async () => {
+      const reports: Array<{ pc: 'pub' | 'sub'; report: RTCStatsReport }> = []
+      try {
+        // Not public API — see file header.
+        const manager = room.engine?.pcManager
+        const pub = await manager?.publisher?.getStats()
+        if (pub) reports.push({ pc: 'pub', report: pub })
+        const sub = await manager?.subscriber?.getStats()
+        if (sub) reports.push({ pc: 'sub', report: sub })
+      } catch {
+        // Engine not ready or SDK internals changed; panel shows nothing.
+      }
+      if (cancelled || reports.length === 0) return
+
+      const contexts = buildTrackContext(room)
+      const tracks: TrackRow[] = []
+      let upKbps = 0
+      let downKbps = 0
+      let rttMs: number | undefined
+      let jitterMs: number | undefined
+      let availableOutKbps: number | undefined
+      let route: StatsSnapshot['route']
+      const turnProtocols = new Set<string>()
+
+      for (const { pc, report } of reports) {
+        const byId = new Map<string, StatDict>()
+        report.forEach((stat) => byId.set(stat.id as string, stat as StatDict))
+
+        report.forEach((raw) => {
+          const stat = raw as StatDict
+          const type = asString(stat.type)
+          const ts = asNumber(stat.timestamp) ?? Date.now()
+          const key = `${pc}:${asString(stat.id) ?? ''}`
+
+          if (
+            type === 'local-candidate' &&
+            asString(stat.candidateType) === 'relay'
+          ) {
+            // Relay candidates are only gathered when a TURN server is
+            // configured and reachable; relayProtocol says how the client
+            // reaches it (udp/tcp/tls) — a turn:…?transport=udp server
+            // must NOT be presented as a TURN/TLS capability.
+            turnProtocols.add(
+              asString(stat.relayProtocol) ?? asString(stat.protocol) ?? 'udp'
+            )
+          }
+
+          if (type === 'transport') {
+            const d = deltas(key, {
+              sent: asNumber(stat.bytesSent) ?? 0,
+              received: asNumber(stat.bytesReceived) ?? 0,
+              ts,
+            })
+            if (d.ts > 0) {
+              upKbps += Math.max(0, (d.sent * 8) / d.ts)
+              downKbps += Math.max(0, (d.received * 8) / d.ts)
+            }
+          }
+
+          if (type === 'candidate-pair' && stat.nominated === true) {
+            const rtt = asNumber(stat.currentRoundTripTime)
+            if (rtt !== undefined) rttMs = Math.round(rtt * 1000)
+            const available = asNumber(stat.availableOutgoingBitrate)
+            if (available !== undefined && pc === 'pub') {
+              availableOutKbps = Math.round(available / 1000)
+            }
+            if (pc === 'pub') {
+              const local = byId.get(asString(stat.localCandidateId) ?? '')
+              route = {
+                protocol: asString(local?.protocol),
+                type: asString(local?.candidateType),
+                relayProtocol: asString(local?.relayProtocol),
+              }
+            }
+          }
+
+          if (type === 'outbound-rtp' || type === 'inbound-rtp') {
+            const isUp = type === 'outbound-rtp'
+            const kind = asString(stat.kind)
+
+            if (!isUp) {
+              const jitter = asNumber(stat.jitter)
+              if (jitter !== undefined) {
+                const ms = Math.round(jitter * 1000)
+                if (jitterMs === undefined || ms > jitterMs) jitterMs = ms
+              }
+            }
+
+            // Packet loss: reported directly on inbound-rtp; for outbound it
+            // lives on the matching remote-inbound-rtp (what the SFU got).
+            let packetsLost = asNumber(stat.packetsLost) ?? 0
+            if (isUp) {
+              const remote = byId.get(asString(stat.remoteId) ?? '')
+              packetsLost = asNumber(remote?.packetsLost) ?? 0
+            }
+
+            const d = deltas(key, {
+              bytes: asNumber(isUp ? stat.bytesSent : stat.bytesReceived) ?? 0,
+              packets:
+                asNumber(isUp ? stat.packetsSent : stat.packetsReceived) ?? 0,
+              packetsLost,
+              ts,
+              freezeCount: asNumber(stat.freezeCount) ?? 0,
+            })
+
+            const kbps = d.ts > 0 ? Math.max(0, (d.bytes * 8) / d.ts) : 0
+            const lostTotal = d.packetsLost + d.packets
+            const lossPct =
+              lostTotal > 0
+                ? Math.max(0, Math.min(100, (d.packetsLost / lostTotal) * 100))
+                : 0
+
+            // Resolve codec + source track.
+            const codecStat = byId.get(asString(stat.codecId) ?? '')
+            let msTrackId = asString(stat.trackIdentifier)
+            if (!msTrackId && isUp) {
+              const mediaSource = byId.get(asString(stat.mediaSourceId) ?? '')
+              msTrackId = asString(mediaSource?.trackIdentifier)
+            }
+            const rid = asString(stat.rid)
+            const context = msTrackId ? contexts.get(msTrackId) : undefined
+            const baseLabel =
+              context?.label ??
+              `${kind ?? 'media'} ssrc ${asNumber(stat.ssrc) ?? '?'}`
+
+            const width = asNumber(stat.frameWidth)
+            const height = asNumber(stat.frameHeight)
+
+            tracks.push({
+              key,
+              dir: isUp ? 'up' : 'down',
+              kind: kind ?? 'unknown',
+              label: rid ? `${baseLabel} [${rid}]` : baseLabel,
+              codec: shortCodec(asString(codecStat?.mimeType)),
+              kbps,
+              fps: asNumber(stat.framesPerSecond),
+              res: width && height ? `${width}x${height}` : undefined,
+              lossPct,
+              limitation:
+                asString(stat.qualityLimitationReason) === 'none'
+                  ? undefined
+                  : asString(stat.qualityLimitationReason),
+              layer:
+                !isUp && kind === 'video' && context
+                  ? resolveLayerReason(context, height)
+                  : undefined,
+              froze: !isUp && kind === 'video' && d.freezeCount > 0,
+            })
+          }
+        })
+      }
+
+      // Stable order (direction, then label): sorting by bitrate would
+      // reshuffle rows on every tick as kbps fluctuates.
+      tracks.sort((a, b) =>
+        a.dir === b.dir
+          ? a.label.localeCompare(b.label)
+          : a.dir === 'up'
+            ? -1
+            : 1
+      )
+
+      const next: StatsSnapshot = {
+        ts: Date.now(),
+        upKbps: Math.round(upKbps),
+        downKbps: Math.round(downKbps),
+        rttMs,
+        jitterMs,
+        availableOutKbps,
+        route,
+        turnProtocols: Array.from(turnProtocols).sort(),
+        tracks,
+      }
+      setHistory((h) => [...h.slice(-59), next])
+    }
+
+    void collect()
+    const id = window.setInterval(() => void collect(), intervalMs)
+    return () => {
+      cancelled = true
+      window.clearInterval(id)
+    }
+  }, [room, enabled, intervalMs])
+
+  return { snapshot: history[history.length - 1], history }
+}

src/frontend/src/features/rooms/components/Conference.tsx

@@ -43,6 +43,7 @@ import { useSnapshot } from 'valtio'
 import { userPreferencesStore } from '@/stores/userPreferences'
 import { userStore } from '@/stores/user'
 import { WatchMediaDeviceErrors } from './WatchMediaDeviceErrors'
+import { MeetDevtools } from '@/features/devtools'
 import { VOICE_AUDIO_CONSTRAINTS } from '@/features/rooms/livekit/utils/constants'
 
 export const Conference = ({
@@ -298,6 +299,7 @@ export const Conference = ({
           <VideoConference />
           {!isMobile && <InviteDialog mode={mode} />}
           <PictureInPictureConference />
+          <MeetDevtools />
         </LiveKitRoom>
       </Screen>
     </QueryAware>

src/frontend/src/features/subtitle/component/Subtitles.tsx

@@ -75,15 +75,15 @@ const useTranscriptionState = () => {
     const segment = segments[0]
 
     setTranscriptionSegments((prevSegments) => {
-      const existingSegmentIds = new Set(prevSegments.map((s) => s.id))
+      const existingIndex = prevSegments.findIndex(
-      if (existingSegmentIds.has(segment.id)) return prevSegments
+        (s: TranscriptionSegmentWithParticipant) => s.id === segment.id
-      return [
+      )
-        ...prevSegments,
+      if (existingIndex === -1) {
-        {
+        return [...prevSegments, { participant, ...segment }]
-          participant: participant,
+      }
-          ...segment,
+      const next = prevSegments.slice()
-        },
+      next[existingIndex] = { ...next[existingIndex], ...segment }
-      ]
+      return next
     })
   }