Update engine

app/main.py

@@ -24,7 +24,6 @@ from core.transports import SocketTransport, WebRtcTransport, BaseTransport
 from core.session import Session
 from processors.tracks import Resampled16kTrack
 from core.events import get_event_bus, reset_event_bus
-from models.ws_v1 import ev
 
 # Check interval for heartbeat/timeout (seconds)
 _HEARTBEAT_CHECK_INTERVAL_SEC = 5
@@ -54,9 +53,7 @@ async def heartbeat_and_timeout_task(
             break
         if now - last_heartbeat_at[0] >= heartbeat_interval_sec:
             try:
-                await transport.send_event({
+                await session.send_heartbeat()
-                    **ev("heartbeat"),
-                })
                 last_heartbeat_at[0] = now
             except Exception as e:
                 logger.debug(f"Session {session_id}: heartbeat send failed: {e}")

core/duplex_pipeline.py

@@ -14,7 +14,8 @@ event-driven design.
 import asyncio
 import json
 import time
-from typing import Any, Dict, List, Optional, Tuple
+import uuid
+from typing import Any, Callable, Dict, List, Optional, Tuple
 
 import numpy as np
 from loguru import logger
@@ -59,6 +60,12 @@ class DuplexPipeline:
     _MIN_SPLIT_SPOKEN_CHARS = 6
     _TOOL_WAIT_TIMEOUT_SECONDS = 15.0
     _SERVER_TOOL_TIMEOUT_SECONDS = 15.0
+    TRACK_AUDIO_IN = "audio_in"
+    TRACK_AUDIO_OUT = "audio_out"
+    TRACK_CONTROL = "control"
+    _PCM_FRAME_BYTES = 640  # 16k mono pcm_s16le, 20ms
+    _ASR_DELTA_THROTTLE_MS = 300
+    _LLM_DELTA_THROTTLE_MS = 80
     _DEFAULT_TOOL_SCHEMAS: Dict[str, Dict[str, Any]] = {
         "current_time": {
             "name": "current_time",
@@ -96,6 +103,9 @@ class DuplexPipeline:
         self.transport = transport
         self.session_id = session_id
         self.event_bus = get_event_bus()
+        self.track_audio_in = self.TRACK_AUDIO_IN
+        self.track_audio_out = self.TRACK_AUDIO_OUT
+        self.track_control = self.TRACK_CONTROL
 
         # Initialize VAD
         self.vad_model = SileroVAD(
@@ -120,6 +130,8 @@ class DuplexPipeline:
 
         # Track last sent transcript to avoid duplicates
         self._last_sent_transcript = ""
+        self._pending_transcript_delta: str = ""
+        self._last_transcript_delta_emit_ms: float = 0.0
 
         # Conversation manager
         self.conversation = ConversationManager(
@@ -153,6 +165,7 @@ class DuplexPipeline:
         self._outbound_seq = 0
         self._outbound_task: Optional[asyncio.Task] = None
         self._drop_outbound_audio = False
+        self._audio_out_frame_buffer: bytes = b""
 
         # Interruption handling
         self._interrupt_event = asyncio.Event()
@@ -186,9 +199,28 @@ class DuplexPipeline:
         self._pending_tool_waiters: Dict[str, asyncio.Future] = {}
         self._early_tool_results: Dict[str, Dict[str, Any]] = {}
         self._completed_tool_call_ids: set[str] = set()
+        self._pending_client_tool_call_ids: set[str] = set()
+        self._next_seq: Optional[Callable[[], int]] = None
+        self._local_seq: int = 0
+
+        # Cross-service correlation IDs
+        self._turn_count: int = 0
+        self._response_count: int = 0
+        self._tts_count: int = 0
+        self._utterance_count: int = 0
+        self._current_turn_id: Optional[str] = None
+        self._current_utterance_id: Optional[str] = None
+        self._current_response_id: Optional[str] = None
+        self._current_tts_id: Optional[str] = None
+        self._pending_llm_delta: str = ""
+        self._last_llm_delta_emit_ms: float = 0.0
 
         logger.info(f"DuplexPipeline initialized for session {session_id}")
 
+    def set_event_sequence_provider(self, provider: Callable[[], int]) -> None:
+        """Use session-scoped monotonic sequence provider for envelope events."""
+        self._next_seq = provider
+
     def apply_runtime_overrides(self, metadata: Optional[Dict[str, Any]]) -> None:
         """
         Apply runtime overrides from WS session.start metadata.
@@ -276,6 +308,131 @@ class DuplexPipeline:
         if self.llm_service and hasattr(self.llm_service, "set_tool_schemas"):
             self.llm_service.set_tool_schemas(self._resolved_tool_schemas())
 
+    def resolved_runtime_config(self) -> Dict[str, Any]:
+        """Return current effective runtime configuration without secrets."""
+        llm_provider = str(self._runtime_llm.get("provider") or "openai").lower()
+        tts_provider = str(self._runtime_tts.get("provider") or settings.tts_provider).lower()
+        asr_provider = str(self._runtime_asr.get("provider") or settings.asr_provider).lower()
+        output_mode = str(self._runtime_output.get("mode") or "").strip().lower()
+        if not output_mode:
+            output_mode = "audio" if self._tts_output_enabled() else "text"
+
+        return {
+            "output": {"mode": output_mode},
+            "services": {
+                "llm": {
+                    "provider": llm_provider,
+                    "model": str(self._runtime_llm.get("model") or settings.llm_model),
+                    "baseUrl": self._runtime_llm.get("baseUrl") or settings.openai_api_url,
+                },
+                "asr": {
+                    "provider": asr_provider,
+                    "model": str(self._runtime_asr.get("model") or settings.siliconflow_asr_model),
+                    "interimIntervalMs": int(self._runtime_asr.get("interimIntervalMs") or settings.asr_interim_interval_ms),
+                    "minAudioMs": int(self._runtime_asr.get("minAudioMs") or settings.asr_min_audio_ms),
+                },
+                "tts": {
+                    "enabled": self._tts_output_enabled(),
+                    "provider": tts_provider,
+                    "model": str(self._runtime_tts.get("model") or settings.siliconflow_tts_model),
+                    "voice": str(self._runtime_tts.get("voice") or settings.tts_voice),
+                    "speed": float(self._runtime_tts.get("speed") or settings.tts_speed),
+                },
+            },
+            "tools": {
+                "allowlist": sorted(self._runtime_tool_executor.keys()),
+            },
+            "tracks": {
+                "audio_in": self.track_audio_in,
+                "audio_out": self.track_audio_out,
+                "control": self.track_control,
+            },
+        }
+
+    def _next_event_seq(self) -> int:
+        if self._next_seq:
+            return self._next_seq()
+        self._local_seq += 1
+        return self._local_seq
+
+    def _event_source(self, event_type: str) -> str:
+        if event_type.startswith("transcript.") or event_type.startswith("input.speech_"):
+            return "asr"
+        if event_type.startswith("assistant.response."):
+            return "llm"
+        if event_type.startswith("assistant.tool_"):
+            return "tool"
+        if event_type.startswith("output.audio.") or event_type == "metrics.ttfb":
+            return "tts"
+        return "system"
+
+    def _new_id(self, prefix: str, counter: int) -> str:
+        return f"{prefix}_{counter}_{uuid.uuid4().hex[:8]}"
+
+    def _start_turn(self) -> str:
+        self._turn_count += 1
+        self._current_turn_id = self._new_id("turn", self._turn_count)
+        self._current_utterance_id = None
+        self._current_response_id = None
+        self._current_tts_id = None
+        return self._current_turn_id
+
+    def _start_response(self) -> str:
+        self._response_count += 1
+        self._current_response_id = self._new_id("resp", self._response_count)
+        self._current_tts_id = None
+        return self._current_response_id
+
+    def _start_tts(self) -> str:
+        self._tts_count += 1
+        self._current_tts_id = self._new_id("tts", self._tts_count)
+        return self._current_tts_id
+
+    def _finalize_utterance(self) -> str:
+        if self._current_utterance_id:
+            return self._current_utterance_id
+        self._utterance_count += 1
+        self._current_utterance_id = self._new_id("utt", self._utterance_count)
+        if not self._current_turn_id:
+            self._start_turn()
+        return self._current_utterance_id
+
+    def _envelope_event(self, event: Dict[str, Any]) -> Dict[str, Any]:
+        event_type = str(event.get("type") or "")
+        source = str(event.get("source") or self._event_source(event_type))
+        track_id = event.get("trackId")
+        if not track_id:
+            if source == "asr":
+                track_id = self.track_audio_in
+            elif source in {"llm", "tts", "tool"}:
+                track_id = self.track_audio_out
+            else:
+                track_id = self.track_control
+
+        data = event.get("data")
+        if not isinstance(data, dict):
+            data = {}
+        if self._current_turn_id:
+            data.setdefault("turn_id", self._current_turn_id)
+        if self._current_utterance_id:
+            data.setdefault("utterance_id", self._current_utterance_id)
+        if self._current_response_id:
+            data.setdefault("response_id", self._current_response_id)
+        if self._current_tts_id:
+            data.setdefault("tts_id", self._current_tts_id)
+
+        for k, v in event.items():
+            if k in {"type", "timestamp", "sessionId", "seq", "source", "trackId", "data"}:
+                continue
+            data.setdefault(k, v)
+
+        event["sessionId"] = self.session_id
+        event["seq"] = self._next_event_seq()
+        event["source"] = source
+        event["trackId"] = track_id
+        event["data"] = data
+        return event
+
     @staticmethod
     def _coerce_bool(value: Any) -> Optional[bool]:
         if isinstance(value, bool):
@@ -472,11 +629,13 @@ class DuplexPipeline:
                         greeting_to_speak = generated_greeting
                         self.conversation.greeting = generated_greeting
                 if greeting_to_speak:
+                    self._start_turn()
+                    self._start_response()
                     await self._send_event(
                         ev(
                             "assistant.response.final",
                             text=greeting_to_speak,
-                            trackId=self.session_id,
+                            trackId=self.track_audio_out,
                         ),
                         priority=20,
                     )
@@ -494,10 +653,58 @@ class DuplexPipeline:
         await self._outbound_q.put((priority, self._outbound_seq, kind, payload))
 
     async def _send_event(self, event: Dict[str, Any], priority: int = 20) -> None:
-        await self._enqueue_outbound("event", event, priority)
+        await self._enqueue_outbound("event", self._envelope_event(event), priority)
 
     async def _send_audio(self, pcm_bytes: bytes, priority: int = 50) -> None:
-        await self._enqueue_outbound("audio", pcm_bytes, priority)
+        if not pcm_bytes:
+            return
+        self._audio_out_frame_buffer += pcm_bytes
+        while len(self._audio_out_frame_buffer) >= self._PCM_FRAME_BYTES:
+            frame = self._audio_out_frame_buffer[: self._PCM_FRAME_BYTES]
+            self._audio_out_frame_buffer = self._audio_out_frame_buffer[self._PCM_FRAME_BYTES :]
+            await self._enqueue_outbound("audio", frame, priority)
+
+    async def _flush_audio_out_frames(self, priority: int = 50) -> None:
+        """Flush remaining outbound audio as one padded 20ms PCM frame."""
+        if not self._audio_out_frame_buffer:
+            return
+        tail = self._audio_out_frame_buffer
+        self._audio_out_frame_buffer = b""
+        if len(tail) < self._PCM_FRAME_BYTES:
+            tail = tail + (b"\x00" * (self._PCM_FRAME_BYTES - len(tail)))
+        await self._enqueue_outbound("audio", tail, priority)
+
+    async def _emit_transcript_delta(self, text: str) -> None:
+        await self._send_event(
+            {
+                **ev(
+                    "transcript.delta",
+                    trackId=self.track_audio_in,
+                    text=text,
+                )
+            },
+            priority=30,
+        )
+
+    async def _emit_llm_delta(self, text: str) -> None:
+        await self._send_event(
+            {
+                **ev(
+                    "assistant.response.delta",
+                    trackId=self.track_audio_out,
+                    text=text,
+                )
+            },
+            priority=20,
+        )
+
+    async def _flush_pending_llm_delta(self) -> None:
+        if not self._pending_llm_delta:
+            return
+        chunk = self._pending_llm_delta
+        self._pending_llm_delta = ""
+        self._last_llm_delta_emit_ms = time.monotonic() * 1000.0
+        await self._emit_llm_delta(chunk)
 
     async def _outbound_loop(self) -> None:
         """Single sender loop that enforces priority for interrupt events."""
@@ -546,13 +753,13 @@ class DuplexPipeline:
 
                     # Emit VAD event
                     await self.event_bus.publish(event_type, {
-                        "trackId": self.session_id,
+                        "trackId": self.track_audio_in,
                         "probability": probability
                     })
                     await self._send_event(
                         ev(
                             "input.speech_started" if event_type == "speaking" else "input.speech_stopped",
-                            trackId=self.session_id,
+                            trackId=self.track_audio_in,
                             probability=probability,
                         ),
                         priority=30,
@@ -661,6 +868,9 @@ class DuplexPipeline:
         # Cancel any current speaking
         await self._stop_current_speech()
 
+        self._start_turn()
+        self._finalize_utterance()
+
         # Start new turn
         await self.conversation.end_user_turn(text)
         self._current_turn_task = asyncio.create_task(self._handle_turn(text))
@@ -683,24 +893,45 @@ class DuplexPipeline:
         if text == self._last_sent_transcript and not is_final:
             return
 
+        now_ms = time.monotonic() * 1000.0
         self._last_sent_transcript = text
 
-        # Send transcript event to client
+        if is_final:
-        await self._send_event({
+            self._pending_transcript_delta = ""
-            **ev(
+            self._last_transcript_delta_emit_ms = 0.0
-                "transcript.final" if is_final else "transcript.delta",
+            await self._send_event(
-                trackId=self.session_id,
+                {
-                text=text,
+                    **ev(
+                        "transcript.final",
+                        trackId=self.track_audio_in,
+                        text=text,
+                    )
+                },
+                priority=30,
             )
-        }, priority=30)
+            logger.debug(f"Sent transcript (final): {text[:50]}...")
+            return
+
+        self._pending_transcript_delta = text
+        should_emit = (
+            self._last_transcript_delta_emit_ms <= 0.0
+            or now_ms - self._last_transcript_delta_emit_ms >= self._ASR_DELTA_THROTTLE_MS
+        )
+        if should_emit and self._pending_transcript_delta:
+            delta = self._pending_transcript_delta
+            self._pending_transcript_delta = ""
+            self._last_transcript_delta_emit_ms = now_ms
+            await self._emit_transcript_delta(delta)
 
         if not is_final:
             logger.info(f"[ASR] ASR interim: {text[:100]}")
-        logger.debug(f"Sent transcript ({'final' if is_final else 'interim'}): {text[:50]}...")
+            logger.debug(f"Sent transcript (interim): {text[:50]}...")
 
     async def _on_speech_start(self) -> None:
         """Handle user starting to speak."""
         if self.conversation.state in (ConversationState.IDLE, ConversationState.INTERRUPTED):
+            self._start_turn()
+            self._finalize_utterance()
             await self.conversation.start_user_turn()
             self._audio_buffer = b""
             self._last_sent_transcript = ""
@@ -779,6 +1010,7 @@ class DuplexPipeline:
             return
 
         logger.info(f"[EOU] Detected - user said: {user_text[:100]}...")
+        self._finalize_utterance()
 
         # For ASR backends that already emitted final via callback,
         # avoid duplicating transcript.final on EOU.
@@ -786,7 +1018,7 @@ class DuplexPipeline:
             await self._send_event({
                 **ev(
                     "transcript.final",
-                    trackId=self.session_id,
+                    trackId=self.track_audio_in,
                     text=user_text,
                 )
             }, priority=25)
@@ -794,6 +1026,8 @@ class DuplexPipeline:
         # Clear buffers
         self._audio_buffer = b""
         self._last_sent_transcript = ""
+        self._pending_transcript_delta = ""
+        self._last_transcript_delta_emit_ms = 0.0
         self._asr_capture_active = False
         self._pending_speech_audio = b""
 
@@ -894,6 +1128,44 @@ class DuplexPipeline:
         # Default to server execution unless explicitly marked as client.
         return "server"
 
+    def _tool_arguments(self, tool_call: Dict[str, Any]) -> Dict[str, Any]:
+        fn = tool_call.get("function")
+        if not isinstance(fn, dict):
+            return {}
+        raw = fn.get("arguments")
+        if isinstance(raw, dict):
+            return raw
+        if isinstance(raw, str) and raw.strip():
+            try:
+                parsed = json.loads(raw)
+                return parsed if isinstance(parsed, dict) else {"raw": raw}
+            except Exception:
+                return {"raw": raw}
+        return {}
+
+    def _normalize_tool_result(self, result: Dict[str, Any]) -> Dict[str, Any]:
+        status = result.get("status") if isinstance(result.get("status"), dict) else {}
+        status_code = int(status.get("code") or 0) if status else 0
+        status_message = str(status.get("message") or "") if status else ""
+        tool_call_id = str(result.get("tool_call_id") or result.get("id") or "")
+        tool_name = str(result.get("name") or "unknown_tool")
+        ok = bool(200 <= status_code < 300)
+        retryable = status_code >= 500 or status_code in {429, 408}
+        error: Optional[Dict[str, Any]] = None
+        if not ok:
+            error = {
+                "code": status_code or 500,
+                "message": status_message or "tool_execution_failed",
+                "retryable": retryable,
+            }
+        return {
+            "tool_call_id": tool_call_id,
+            "tool_name": tool_name,
+            "ok": ok,
+            "error": error,
+            "status": {"code": status_code, "message": status_message},
+        }
+
     async def _emit_tool_result(self, result: Dict[str, Any], source: str) -> None:
         tool_name = str(result.get("name") or "unknown_tool")
         call_id = str(result.get("tool_call_id") or result.get("id") or "")
@@ -904,12 +1176,17 @@ class DuplexPipeline:
             f"[Tool] emit result source={source} name={tool_name} call_id={call_id} "
             f"status={status_code} {status_message}".strip()
         )
+        normalized = self._normalize_tool_result(result)
         await self._send_event(
             {
                 **ev(
                     "assistant.tool_result",
-                    trackId=self.session_id,
+                    trackId=self.track_audio_out,
                     source=source,
+                    tool_call_id=normalized["tool_call_id"],
+                    tool_name=normalized["tool_name"],
+                    ok=normalized["ok"],
+                    error=normalized["error"],
                     result=result,
                 )
             },
@@ -927,6 +1204,9 @@ class DuplexPipeline:
             call_id = str(item.get("tool_call_id") or item.get("id") or "").strip()
             if not call_id:
                 continue
+            if self._pending_client_tool_call_ids and call_id not in self._pending_client_tool_call_ids:
+                logger.warning(f"[Tool] ignore unsolicited client result call_id={call_id}")
+                continue
             if call_id in self._completed_tool_call_ids:
                 logger.debug(f"[Tool] ignore duplicate client result call_id={call_id}")
                 continue
@@ -972,6 +1252,7 @@ class DuplexPipeline:
             }
         finally:
             self._pending_tool_waiters.pop(call_id, None)
+            self._pending_client_tool_call_ids.discard(call_id)
 
     def _normalize_stream_event(self, item: Any) -> LLMStreamEvent:
         if isinstance(item, LLMStreamEvent):
@@ -998,6 +1279,11 @@ class DuplexPipeline:
             user_text: User's transcribed text
         """
         try:
+            if not self._current_turn_id:
+                self._start_turn()
+            if not self._current_utterance_id:
+                self._finalize_utterance()
+            self._start_response()
             # Start latency tracking
             self._turn_start_time = time.time()
             self._first_audio_sent = False
@@ -1012,6 +1298,8 @@ class DuplexPipeline:
             self._drop_outbound_audio = False
 
             first_audio_sent = False
+            self._pending_llm_delta = ""
+            self._last_llm_delta_emit_ms = 0.0
             for _ in range(max_rounds):
                 if self._interrupt_event.is_set():
                     break
@@ -1028,6 +1316,7 @@ class DuplexPipeline:
 
                     event = self._normalize_stream_event(raw_event)
                     if event.type == "tool_call":
+                        await self._flush_pending_llm_delta()
                         tool_call = event.tool_call if isinstance(event.tool_call, dict) else None
                         if not tool_call:
                             continue
@@ -1045,11 +1334,19 @@ class DuplexPipeline:
                             f"executor={executor} args={args_preview}"
                         )
                         tool_calls.append(enriched_tool_call)
+                        tool_arguments = self._tool_arguments(enriched_tool_call)
+                        if executor == "client" and call_id:
+                            self._pending_client_tool_call_ids.add(call_id)
                         await self._send_event(
                             {
                                 **ev(
                                     "assistant.tool_call",
-                                    trackId=self.session_id,
+                                    trackId=self.track_audio_out,
+                                    tool_call_id=call_id,
+                                    tool_name=tool_name,
+                                    arguments=tool_arguments,
+                                    executor=executor,
+                                    timeout_ms=int(self._TOOL_WAIT_TIMEOUT_SECONDS * 1000),
                                     tool_call=enriched_tool_call,
                                 )
                             },
@@ -1071,19 +1368,13 @@ class DuplexPipeline:
                     round_response += text_chunk
                     sentence_buffer += text_chunk
                     await self.conversation.update_assistant_text(text_chunk)
-
+                    self._pending_llm_delta += text_chunk
-                    await self._send_event(
+                    now_ms = time.monotonic() * 1000.0
-                        {
+                    if (
-                            **ev(
+                        self._last_llm_delta_emit_ms <= 0.0
-                                "assistant.response.delta",
+                        or now_ms - self._last_llm_delta_emit_ms >= self._LLM_DELTA_THROTTLE_MS
-                                trackId=self.session_id,
+                    ):
-                                text=text_chunk,
+                        await self._flush_pending_llm_delta()
-                            )
-                        },
-                        # Keep delta/final on the same event priority so FIFO seq
-                        # preserves stream order (avoid late-delta after final).
-                        priority=20,
-                    )
 
                     while True:
                         split_result = extract_tts_sentence(
@@ -1112,11 +1403,12 @@ class DuplexPipeline:
 
                         if self._tts_output_enabled() and not self._interrupt_event.is_set():
                             if not first_audio_sent:
+                                self._start_tts()
                                 await self._send_event(
                                     {
                                         **ev(
                                             "output.audio.start",
-                                            trackId=self.session_id,
+                                            trackId=self.track_audio_out,
                                         )
                                     },
                                     priority=10,
@@ -1130,6 +1422,7 @@ class DuplexPipeline:
                             )
 
                 remaining_text = f"{pending_punctuation}{sentence_buffer}".strip()
+                await self._flush_pending_llm_delta()
                 if (
                     self._tts_output_enabled()
                     and remaining_text
@@ -1137,11 +1430,12 @@ class DuplexPipeline:
                     and not self._interrupt_event.is_set()
                 ):
                     if not first_audio_sent:
+                        self._start_tts()
                         await self._send_event(
                             {
                                 **ev(
                                     "output.audio.start",
-                                    trackId=self.session_id,
+                                    trackId=self.track_audio_out,
                                 )
                             },
                             priority=10,
@@ -1204,11 +1498,12 @@ class DuplexPipeline:
                 ]
 
             if full_response and not self._interrupt_event.is_set():
+                await self._flush_pending_llm_delta()
                 await self._send_event(
                     {
                         **ev(
                             "assistant.response.final",
-                            trackId=self.session_id,
+                            trackId=self.track_audio_out,
                             text=full_response,
                         )
                     },
@@ -1217,10 +1512,11 @@ class DuplexPipeline:
 
             # Send track end
             if first_audio_sent:
+                await self._flush_audio_out_frames(priority=50)
                 await self._send_event({
                     **ev(
                         "output.audio.end",
-                        trackId=self.session_id,
+                        trackId=self.track_audio_out,
                     )
                 }, priority=10)
 
@@ -1241,6 +1537,8 @@ class DuplexPipeline:
             self._barge_in_speech_start_time = None
             self._barge_in_speech_frames = 0
             self._barge_in_silence_frames = 0
+            self._current_response_id = None
+            self._current_tts_id = None
 
     async def _speak_sentence(self, text: str, fade_in_ms: int = 0, fade_out_ms: int = 8) -> None:
         """
@@ -1277,7 +1575,7 @@ class DuplexPipeline:
                     await self._send_event({
                         **ev(
                             "metrics.ttfb",
-                            trackId=self.session_id,
+                            trackId=self.track_audio_out,
                             latencyMs=round(ttfb_ms),
                         )
                     }, priority=25)
@@ -1354,10 +1652,11 @@ class DuplexPipeline:
             first_audio_sent = False
 
             # Send track start event
+            self._start_tts()
             await self._send_event({
                 **ev(
                     "output.audio.start",
-                    trackId=self.session_id,
+                    trackId=self.track_audio_out,
                 )
             }, priority=10)
 
@@ -1379,7 +1678,7 @@ class DuplexPipeline:
                     await self._send_event({
                         **ev(
                             "metrics.ttfb",
-                            trackId=self.session_id,
+                            trackId=self.track_audio_out,
                             latencyMs=round(ttfb_ms),
                         )
                     }, priority=25)
@@ -1391,10 +1690,11 @@ class DuplexPipeline:
                 await asyncio.sleep(0.01)
 
             # Send track end event
+            await self._flush_audio_out_frames(priority=50)
             await self._send_event({
                 **ev(
                     "output.audio.end",
-                    trackId=self.session_id,
+                    trackId=self.track_audio_out,
                 )
             }, priority=10)
 
@@ -1422,13 +1722,14 @@ class DuplexPipeline:
         self._interrupt_event.set()
         self._is_bot_speaking = False
         self._drop_outbound_audio = True
+        self._audio_out_frame_buffer = b""
 
         # Send interrupt event to client IMMEDIATELY
         # This must happen BEFORE canceling services, so client knows to discard in-flight audio
         await self._send_event({
             **ev(
                 "response.interrupted",
-                trackId=self.session_id,
+                trackId=self.track_audio_out,
             )
         }, priority=0)
 
@@ -1455,6 +1756,7 @@ class DuplexPipeline:
     async def _stop_current_speech(self) -> None:
         """Stop any current speech task."""
         self._drop_outbound_audio = True
+        self._audio_out_frame_buffer = b""
         if self._current_turn_task and not self._current_turn_task.done():
             self._interrupt_event.set()
             self._current_turn_task.cancel()

core/session.py

@@ -1,15 +1,16 @@
 """Session management for active calls."""
 
 import asyncio
-import uuid
+import hashlib
 import json
-import time
 import re
+import time
 from enum import Enum
 from typing import Optional, Dict, Any, List
 from loguru import logger
 
 from app.backend_client import (
+    fetch_assistant_config,
     create_history_call_record,
     add_history_transcript,
     finalize_history_call_record,
@@ -49,6 +50,32 @@ class Session:
     Uses full duplex voice conversation pipeline.
     """
 
+    TRACK_AUDIO_IN = "audio_in"
+    TRACK_AUDIO_OUT = "audio_out"
+    TRACK_CONTROL = "control"
+    AUDIO_FRAME_BYTES = 640  # 16k mono pcm_s16le, 20ms
+    _CLIENT_METADATA_OVERRIDES = {
+        "firstTurnMode",
+        "greeting",
+        "generatedOpenerEnabled",
+        "systemPrompt",
+        "output",
+        "bargeIn",
+        "knowledge",
+        "knowledgeBaseId",
+        "history",
+        "userId",
+        "assistantId",
+        "source",
+    }
+    _CLIENT_METADATA_ID_KEYS = {
+        "appId",
+        "app_id",
+        "channel",
+        "configVersionId",
+        "config_version_id",
+    }
+
     def __init__(self, session_id: str, transport: BaseTransport, use_duplex: bool = None):
         """
         Initialize session.
@@ -78,7 +105,10 @@ class Session:
         self.authenticated: bool = False
 
         # Track IDs
-        self.current_track_id: Optional[str] = str(uuid.uuid4())
+        self.current_track_id: str = self.TRACK_CONTROL
+        self._event_seq: int = 0
+        self._audio_ingress_buffer: bytes = b""
+        self._audio_frame_error_reported: bool = False
         self._history_call_id: Optional[str] = None
         self._history_turn_index: int = 0
         self._history_call_started_mono: Optional[float] = None
@@ -89,6 +119,7 @@ class Session:
         self._workflow_last_user_text: str = ""
         self._workflow_initial_node: Optional[WorkflowNodeDef] = None
 
+        self.pipeline.set_event_sequence_provider(self._next_event_seq)
         self.pipeline.conversation.on_turn_complete(self._on_turn_complete)
 
         logger.info(f"Session {self.id} created (duplex={self.use_duplex})")
@@ -129,13 +160,52 @@ class Session:
                 "client",
                 "Audio received before session.start",
                 "protocol.order",
+                stage="protocol",
+                retryable=False,
             )
             return
 
         try:
-            await self.pipeline.process_audio(audio_bytes)
+            if not audio_bytes:
+                return
+            if len(audio_bytes) % 2 != 0:
+                await self._send_error(
+                    "client",
+                    "Invalid PCM payload: odd number of bytes",
+                    "audio.invalid_pcm",
+                    stage="audio",
+                    retryable=False,
+                )
+                return
+
+            frame_bytes = self.AUDIO_FRAME_BYTES
+            self._audio_ingress_buffer += audio_bytes
+
+            # Protocol v1 audio framing: 20ms PCM frame (640 bytes).
+            # Allow aggregated frames in one WS message (multiple of 640).
+            if len(audio_bytes) % frame_bytes != 0 and not self._audio_frame_error_reported:
+                self._audio_frame_error_reported = True
+                await self._send_error(
+                    "client",
+                    f"Audio frame size should be multiple of {frame_bytes} bytes (20ms PCM)",
+                    "audio.frame_size_mismatch",
+                    stage="audio",
+                    retryable=True,
+                )
+
+            while len(self._audio_ingress_buffer) >= frame_bytes:
+                frame = self._audio_ingress_buffer[:frame_bytes]
+                self._audio_ingress_buffer = self._audio_ingress_buffer[frame_bytes:]
+                await self.pipeline.process_audio(frame)
         except Exception as e:
             logger.error(f"Session {self.id} handle_audio error: {e}", exc_info=True)
+            await self._send_error(
+                "server",
+                f"Audio processing failed: {e}",
+                "audio.processing_failed",
+                stage="audio",
+                retryable=True,
+            )
 
     async def _handle_v1_message(self, message: Any) -> None:
         """Route validated WS v1 message to handlers."""
@@ -217,10 +287,9 @@ class Session:
         self.authenticated = True
         self.protocol_version = message.version
         self.ws_state = WsSessionState.WAIT_START
-        await self.transport.send_event(
+        await self._send_event(
             ev(
                 "hello.ack",
-                sessionId=self.id,
                 version=self.protocol_version,
             )
         )
@@ -231,8 +300,12 @@ class Session:
             await self._send_error("client", "Duplicate session.start", "protocol.order")
             return
 
-        metadata = message.metadata or {}
+        raw_metadata = message.metadata or {}
-        metadata = self._merge_runtime_metadata(metadata, self._bootstrap_workflow(metadata))
+        workflow_runtime = self._bootstrap_workflow(raw_metadata)
+        server_runtime = await self._load_server_runtime_metadata(raw_metadata, workflow_runtime)
+        client_runtime = self._sanitize_client_metadata(raw_metadata)
+        metadata = self._merge_runtime_metadata(server_runtime, self._sanitize_untrusted_runtime_metadata(workflow_runtime))
+        metadata = self._merge_runtime_metadata(metadata, client_runtime)
 
         # Create history call record early so later turn callbacks can append transcripts.
         await self._start_history_bridge(metadata)
@@ -248,28 +321,37 @@ class Session:
 
         self.state = "accepted"
         self.ws_state = WsSessionState.ACTIVE
-        await self.transport.send_event(
+        await self._send_event(
             ev(
                 "session.started",
-                sessionId=self.id,
                 trackId=self.current_track_id,
+                tracks={
+                    "audio_in": self.TRACK_AUDIO_IN,
+                    "audio_out": self.TRACK_AUDIO_OUT,
+                    "control": self.TRACK_CONTROL,
+                },
                 audio=message.audio or {},
             )
         )
+        await self._send_event(
+            ev(
+                "config.resolved",
+                trackId=self.TRACK_CONTROL,
+                config=self._build_config_resolved(metadata),
+            )
+        )
         if self.workflow_runner and self._workflow_initial_node:
-            await self.transport.send_event(
+            await self._send_event(
                 ev(
                     "workflow.started",
-                    sessionId=self.id,
                     workflowId=self.workflow_runner.workflow_id,
                     workflowName=self.workflow_runner.name,
                     nodeId=self._workflow_initial_node.id,
                 )
             )
-            await self.transport.send_event(
+            await self._send_event(
                 ev(
                     "workflow.node.entered",
-                    sessionId=self.id,
                     workflowId=self.workflow_runner.workflow_id,
                     nodeId=self._workflow_initial_node.id,
                     nodeName=self._workflow_initial_node.name,
@@ -285,17 +367,23 @@ class Session:
         stop_reason = reason or "client_requested"
         self.state = "hungup"
         self.ws_state = WsSessionState.STOPPED
-        await self.transport.send_event(
+        await self._send_event(
             ev(
                 "session.stopped",
-                sessionId=self.id,
                 reason=stop_reason,
             )
         )
         await self._finalize_history(status="connected")
         await self.transport.close()
 
-    async def _send_error(self, sender: str, error_message: str, code: str) -> None:
+    async def _send_error(
+        self,
+        sender: str,
+        error_message: str,
+        code: str,
+        stage: Optional[str] = None,
+        retryable: Optional[bool] = None,
+    ) -> None:
         """
         Send error event to client.
 
@@ -304,13 +392,25 @@ class Session:
             error_message: Error message
             code: Machine-readable error code
         """
-        await self.transport.send_event(
+        resolved_stage = stage or self._infer_error_stage(code)
+        resolved_retryable = retryable if retryable is not None else (resolved_stage in {"asr", "llm", "tts", "tool", "audio"})
+        await self._send_event(
             ev(
                 "error",
                 sender=sender,
                 code=code,
                 message=error_message,
+                stage=resolved_stage,
+                retryable=resolved_retryable,
                 trackId=self.current_track_id,
+                data={
+                    "error": {
+                        "stage": resolved_stage,
+                        "code": code,
+                        "message": error_message,
+                        "retryable": resolved_retryable,
+                    }
+                },
             )
         )
 
@@ -483,10 +583,9 @@ class Session:
         node = transition.node
         edge = transition.edge
 
-        await self.transport.send_event(
+        await self._send_event(
             ev(
                 "workflow.edge.taken",
-                sessionId=self.id,
                 workflowId=self.workflow_runner.workflow_id,
                 edgeId=edge.id,
                 fromNodeId=edge.from_node_id,
@@ -494,10 +593,9 @@ class Session:
                 reason=reason,
             )
         )
-        await self.transport.send_event(
+        await self._send_event(
             ev(
                 "workflow.node.entered",
-                sessionId=self.id,
                 workflowId=self.workflow_runner.workflow_id,
                 nodeId=node.id,
                 nodeName=node.name,
@@ -510,10 +608,9 @@ class Session:
             self.pipeline.apply_runtime_overrides(node_runtime)
 
         if node.node_type == "tool":
-            await self.transport.send_event(
+            await self._send_event(
                 ev(
                     "workflow.tool.requested",
-                    sessionId=self.id,
                     workflowId=self.workflow_runner.workflow_id,
                     nodeId=node.id,
                     tool=node.tool or {},
@@ -522,10 +619,9 @@ class Session:
             return
 
         if node.node_type == "human_transfer":
-            await self.transport.send_event(
+            await self._send_event(
                 ev(
                     "workflow.human_transfer",
-                    sessionId=self.id,
                     workflowId=self.workflow_runner.workflow_id,
                     nodeId=node.id,
                 )
@@ -534,16 +630,68 @@ class Session:
             return
 
         if node.node_type == "end":
-            await self.transport.send_event(
+            await self._send_event(
                 ev(
                     "workflow.ended",
-                    sessionId=self.id,
                     workflowId=self.workflow_runner.workflow_id,
                     nodeId=node.id,
                 )
             )
             await self._handle_session_stop("workflow_end")
 
+    def _next_event_seq(self) -> int:
+        self._event_seq += 1
+        return self._event_seq
+
+    def _event_source(self, event_type: str) -> str:
+        if event_type.startswith("workflow."):
+            return "system"
+        if event_type.startswith("session.") or event_type.startswith("hello.") or event_type == "heartbeat":
+            return "system"
+        if event_type == "error":
+            return "system"
+        return "system"
+
+    def _infer_error_stage(self, code: str) -> str:
+        normalized = str(code or "").strip().lower()
+        if normalized.startswith("audio."):
+            return "audio"
+        if normalized.startswith("tool."):
+            return "tool"
+        if normalized.startswith("asr."):
+            return "asr"
+        if normalized.startswith("llm."):
+            return "llm"
+        if normalized.startswith("tts."):
+            return "tts"
+        return "protocol"
+
+    def _envelope_event(self, event: Dict[str, Any]) -> Dict[str, Any]:
+        event_type = str(event.get("type") or "")
+        source = str(event.get("source") or self._event_source(event_type))
+        track_id = event.get("trackId") or self.TRACK_CONTROL
+
+        data = event.get("data")
+        if not isinstance(data, dict):
+            data = {}
+        for k, v in event.items():
+            if k in {"type", "timestamp", "sessionId", "seq", "source", "trackId", "data"}:
+                continue
+            data.setdefault(k, v)
+
+        event["sessionId"] = self.id
+        event["seq"] = self._next_event_seq()
+        event["source"] = source
+        event["trackId"] = track_id
+        event["data"] = data
+        return event
+
+    async def _send_event(self, event: Dict[str, Any]) -> None:
+        await self.transport.send_event(self._envelope_event(event))
+
+    async def send_heartbeat(self) -> None:
+        await self._send_event(ev("heartbeat", trackId=self.TRACK_CONTROL))
+
     async def _workflow_llm_route(
         self,
         node: WorkflowNodeDef,
@@ -629,6 +777,100 @@ class Session:
                 merged[key] = value
         return merged
 
+    async def _load_server_runtime_metadata(
+        self,
+        client_metadata: Dict[str, Any],
+        workflow_runtime: Dict[str, Any],
+    ) -> Dict[str, Any]:
+        """Load trusted runtime metadata from backend assistant config."""
+        assistant_id = (
+            workflow_runtime.get("assistantId")
+            or client_metadata.get("assistantId")
+            or client_metadata.get("appId")
+            or client_metadata.get("app_id")
+        )
+        if assistant_id is None:
+            return {}
+        if not settings.backend_url:
+            return {}
+
+        payload = await fetch_assistant_config(str(assistant_id).strip())
+        if not isinstance(payload, dict):
+            return {}
+
+        assistant_cfg = payload.get("assistant") if isinstance(payload.get("assistant"), dict) else payload
+        if not isinstance(assistant_cfg, dict):
+            return {}
+
+        runtime: Dict[str, Any] = {}
+        if assistant_cfg.get("systemPrompt") is not None:
+            runtime["systemPrompt"] = str(assistant_cfg.get("systemPrompt") or "")
+        elif assistant_cfg.get("prompt") is not None:
+            runtime["systemPrompt"] = str(assistant_cfg.get("prompt") or "")
+
+        if assistant_cfg.get("greeting") is not None:
+            runtime["greeting"] = assistant_cfg.get("greeting")
+        elif assistant_cfg.get("opener") is not None:
+            runtime["greeting"] = assistant_cfg.get("opener")
+
+        if isinstance(assistant_cfg.get("services"), dict):
+            runtime["services"] = assistant_cfg.get("services")
+        if isinstance(assistant_cfg.get("tools"), list):
+            runtime["tools"] = assistant_cfg.get("tools")
+
+        runtime["assistantId"] = str(assistant_id)
+        return runtime
+
+    def _sanitize_untrusted_runtime_metadata(self, metadata: Dict[str, Any]) -> Dict[str, Any]:
+        """
+        Sanitize untrusted metadata sources.
+
+        This keeps only a small override whitelist and stable config ID fields.
+        """
+        if not isinstance(metadata, dict):
+            return {}
+
+        sanitized: Dict[str, Any] = {}
+        for key in self._CLIENT_METADATA_ID_KEYS:
+            if key in metadata:
+                sanitized[key] = metadata[key]
+        for key in self._CLIENT_METADATA_OVERRIDES:
+            if key in metadata:
+                sanitized[key] = metadata[key]
+
+        return sanitized
+
+    def _sanitize_client_metadata(self, metadata: Dict[str, Any]) -> Dict[str, Any]:
+        """Apply client metadata whitelist and remove forbidden secrets."""
+        sanitized = self._sanitize_untrusted_runtime_metadata(metadata)
+        if isinstance(metadata.get("services"), dict):
+            logger.warning(
+                "Session {} provided metadata.services from client; client-side service config is ignored",
+                self.id,
+            )
+        return sanitized
+
+    def _build_config_resolved(self, metadata: Dict[str, Any]) -> Dict[str, Any]:
+        """Build public resolved config payload (secrets removed)."""
+        system_prompt = str(metadata.get("systemPrompt") or self.pipeline.conversation.system_prompt or "")
+        prompt_hash = hashlib.sha256(system_prompt.encode("utf-8")).hexdigest() if system_prompt else None
+        runtime = self.pipeline.resolved_runtime_config()
+
+        return {
+            "appId": metadata.get("appId") or metadata.get("app_id") or metadata.get("assistantId"),
+            "channel": metadata.get("channel"),
+            "configVersionId": metadata.get("configVersionId") or metadata.get("config_version_id"),
+            "prompt": {"sha256": prompt_hash},
+            "output": runtime.get("output", {}),
+            "services": runtime.get("services", {}),
+            "tools": runtime.get("tools", {}),
+            "tracks": {
+                "audio_in": self.TRACK_AUDIO_IN,
+                "audio_out": self.TRACK_AUDIO_OUT,
+                "control": self.TRACK_CONTROL,
+            },
+        }
+
     def _extract_json_obj(self, text: str) -> Optional[Dict[str, Any]]:
         """Best-effort extraction of a JSON object from freeform text."""
         try:

docs/ws_v1_schema.md

@@ -52,43 +52,26 @@ Rules:
     "channels": 1
   },
   "metadata": {
+    "appId": "assistant_123",
+    "channel": "web",
+    "configVersionId": "cfg_20260217_01",
     "client": "web-debug",
     "output": {
       "mode": "audio"
     },
     "systemPrompt": "You are concise.",
-    "greeting": "Hi, how can I help?",
+    "greeting": "Hi, how can I help?"
-    "services": {
-      "llm": {
-        "provider": "openai",
-        "model": "gpt-4o-mini",
-        "apiKey": "sk-...",
-        "baseUrl": "https://api.openai.com/v1"
-      },
-      "asr": {
-        "provider": "openai_compatible",
-        "model": "FunAudioLLM/SenseVoiceSmall",
-        "apiKey": "sf-...",
-        "interimIntervalMs": 500,
-        "minAudioMs": 300
-      },
-      "tts": {
-        "enabled": true,
-        "provider": "openai_compatible",
-        "model": "FunAudioLLM/CosyVoice2-0.5B",
-        "apiKey": "sf-...",
-        "voice": "anna",
-        "speed": 1.0
-      }
-    }
   }
 }
 ```
 
-`metadata.services` is optional. If omitted, server defaults to environment configuration.
+Rules:
+- Client-side `metadata.services` is ignored.
+- Service config (including secrets) is resolved server-side (env/backend).
+- Client should pass stable IDs (`appId`, `channel`, `configVersionId`) plus small runtime overrides (e.g. `output`, `bargeIn`, greeting/prompt style hints).
 
 Text-only mode:
-- Set `metadata.output.mode = "text"` OR `metadata.services.tts.enabled = false`.
+- Set `metadata.output.mode = "text"`.
 - In this mode server still sends `assistant.response.delta/final`, but will not emit audio frames or `output.audio.start/end`.
 
 ### `input.text`
@@ -121,6 +104,7 @@ Text-only mode:
 ### `tool_call.results`
 
 Client tool execution results returned to server.
+Only needed when `assistant.tool_call.executor == "client"` (default execution is server-side).
 
 ```json
 {
@@ -138,21 +122,35 @@ Client tool execution results returned to server.
 
 ## Server -> Client Events
 
-All server events include:
+All server events include an envelope:
 
 ```json
 {
   "type": "event.name",
-  "timestamp": 1730000000000
+  "timestamp": 1730000000000,
+  "sessionId": "sess_xxx",
+  "seq": 42,
+  "source": "asr",
+  "trackId": "audio_in",
+  "data": {}
 }
 ```
 
+Envelope notes:
+- `seq` is monotonically increasing within one session (for replay/resume).
+- `source` is one of: `asr | llm | tts | tool | system`.
+- `data` is structured payload; legacy top-level fields are kept for compatibility.
+
 Common events:
 
 - `hello.ack`
   - Fields: `sessionId`, `version`
 - `session.started`
-  - Fields: `sessionId`, `trackId`, `audio`
+  - Fields: `sessionId`, `trackId`, `tracks`, `audio`
+- `config.resolved`
+  - Fields: `sessionId`, `trackId`, `config`
+  - Sent immediately after `session.started`.
+  - Contains effective model/voice/output/tool allowlist/prompt hash, and never includes secrets.
 - `session.stopped`
   - Fields: `sessionId`, `reason`
 - `heartbeat`
@@ -169,9 +167,10 @@ Common events:
 - `assistant.response.final`
   - Fields: `trackId`, `text`
 - `assistant.tool_call`
-  - Fields: `trackId`, `tool_call` (`tool_call.executor` is `client` or `server`)
+  - Fields: `trackId`, `tool_call`, `tool_call_id`, `tool_name`, `arguments`, `executor`, `timeout_ms`
 - `assistant.tool_result`
-  - Fields: `trackId`, `source`, `result`
+  - Fields: `trackId`, `source`, `result`, `tool_call_id`, `tool_name`, `ok`, `error`
+  - `error`: `{ code, message, retryable }` when `ok=false`
 - `output.audio.start`
   - Fields: `trackId`
 - `output.audio.end`
@@ -183,15 +182,49 @@ Common events:
 - `error`
   - Fields: `sender`, `code`, `message`, `trackId`
 
+Track IDs (MVP fixed values):
+- `audio_in`: ASR/VAD input-side events (`input.*`, `transcript.*`)
+- `audio_out`: assistant output-side events (`assistant.*`, `output.audio.*`, `response.interrupted`, `metrics.ttfb`)
+- `control`: session/control events (`session.*`, `hello.*`, `error`, `config.resolved`)
+
+Correlation IDs (`event.data`):
+- `turn_id`: one user-assistant interaction turn.
+- `utterance_id`: one ASR final utterance.
+- `response_id`: one assistant response generation.
+- `tool_call_id`: one tool invocation.
+- `tts_id`: one TTS playback segment.
+
 ## Binary Audio Frames
 
 After `session.started`, client may send binary PCM chunks continuously.
 
-Recommended format:
+MVP fixed format:
-- 16-bit signed little-endian PCM.
+- 16-bit signed little-endian PCM (`pcm_s16le`)
-- 1 channel.
+- mono (1 channel)
-- 16000 Hz.
+- 16000 Hz
-- 20ms frames (640 bytes) preferred.
+- 20ms frame = 640 bytes
+
+Framing rules:
+- Binary audio frame unit is 640 bytes.
+- A WS binary message may carry one or multiple complete 640-byte frames.
+- Non-640-multiple payloads are treated as `audio.frame_size_mismatch` protocol errors.
+
+TTS boundary events:
+- `output.audio.start` and `output.audio.end` mark assistant playback boundaries.
+
+## Event Throttling
+
+To keep client rendering and server load stable, v1 applies/recommends:
+- `transcript.delta`: merge to ~200-500ms cadence (server default: 300ms).
+- `assistant.response.delta`: merge to ~50-100ms cadence (server default: 80ms).
+- Metrics streams (if enabled beyond `metrics.ttfb`): emit every ~500-1000ms.
+
+## Error Structure
+
+`error` keeps legacy top-level fields (`code`, `message`) and adds structured info:
+- `stage`: `protocol | asr | llm | tts | tool | audio`
+- `retryable`: boolean
+- `data.error`: `{ stage, code, message, retryable }`
 
 ## Compatibility
 

examples/mic_client.py

@@ -59,8 +59,12 @@ class MicrophoneClient:
         url: str,
         sample_rate: int = 16000,
         chunk_duration_ms: int = 20,
+        app_id: str = "assistant_demo",
+        channel: str = "mic_client",
+        config_version_id: str = "local-dev",
         input_device: int = None,
-        output_device: int = None
+        output_device: int = None,
+        track_debug: bool = False,
     ):
         """
         Initialize microphone client.
@@ -76,8 +80,12 @@ class MicrophoneClient:
         self.sample_rate = sample_rate
         self.chunk_duration_ms = chunk_duration_ms
         self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
+        self.app_id = app_id
+        self.channel = channel
+        self.config_version_id = config_version_id
         self.input_device = input_device
         self.output_device = output_device
+        self.track_debug = track_debug
         
         # WebSocket connection
         self.ws = None
@@ -106,6 +114,17 @@ class MicrophoneClient:
         
         # Verbose mode for streaming LLM responses
         self.verbose = False
+
+    @staticmethod
+    def _event_ids_suffix(event: dict) -> str:
+        data = event.get("data") if isinstance(event.get("data"), dict) else {}
+        keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
+        parts = []
+        for key in keys:
+            value = data.get(key, event.get(key))
+            if value:
+                parts.append(f"{key}={value}")
+        return f" [{' '.join(parts)}]" if parts else ""
     
     async def connect(self) -> None:
         """Connect to WebSocket server."""
@@ -114,20 +133,30 @@ class MicrophoneClient:
         self.running = True
         print("Connected!")
         
-        # Send invite command
+        # WS v1 handshake: hello -> session.start
         await self.send_command({
-            "command": "invite",
+            "type": "hello",
-            "option": {
+            "version": "v1",
-                "codec": "pcm",
+        })
-                "sampleRate": self.sample_rate
+        await self.send_command({
-            }
+            "type": "session.start",
+            "audio": {
+                "encoding": "pcm_s16le",
+                "sample_rate_hz": self.sample_rate,
+                "channels": 1,
+            },
+            "metadata": {
+                "appId": self.app_id,
+                "channel": self.channel,
+                "configVersionId": self.config_version_id,
+            },
         })
     
     async def send_command(self, cmd: dict) -> None:
         """Send JSON command to server."""
         if self.ws:
             await self.ws.send(json.dumps(cmd))
-            print(f"→ Command: {cmd.get('command', 'unknown')}")
+            print(f"→ Command: {cmd.get('type', 'unknown')}")
     
     async def send_chat(self, text: str) -> None:
         """Send chat message (text input)."""
@@ -136,7 +165,7 @@ class MicrophoneClient:
         self.first_audio_received = False
         
         await self.send_command({
-            "command": "chat",
+            "type": "input.text",
             "text": text
         })
         print(f"→ Chat: {text}")
@@ -144,13 +173,14 @@ class MicrophoneClient:
     async def send_interrupt(self) -> None:
         """Send interrupt command."""
         await self.send_command({
-            "command": "interrupt"
+            "type": "response.cancel",
+            "graceful": False,
         })
     
     async def send_hangup(self, reason: str = "User quit") -> None:
         """Send hangup command."""
         await self.send_command({
-            "command": "hangup",
+            "type": "session.stop",
             "reason": reason
         })
     
@@ -295,43 +325,48 @@ class MicrophoneClient:
     
     async def _handle_event(self, event: dict) -> None:
         """Handle incoming event."""
-        event_type = event.get("event", "unknown")
+        event_type = event.get("type", event.get("event", "unknown"))
+        ids = self._event_ids_suffix(event)
+        if self.track_debug:
+            print(f"[track-debug] event={event_type} trackId={event.get('trackId')}{ids}")
         
-        if event_type == "answer":
+        if event_type in {"hello.ack", "session.started"}:
-            print("← Session ready!")
+            print(f"← Session ready!{ids}")
-        elif event_type == "speaking":
+        elif event_type == "config.resolved":
-            print("← User speech detected")
+            print(f"← Config resolved: {event.get('config', {}).get('output', {})}{ids}")
-        elif event_type == "silence":
+        elif event_type == "input.speech_started":
-            print("← User silence detected")
+            print(f"← User speech detected{ids}")
-        elif event_type == "transcript":
+        elif event_type == "input.speech_stopped":
+            print(f"← User silence detected{ids}")
+        elif event_type in {"transcript", "transcript.delta", "transcript.final"}:
             # Display user speech transcription
             text = event.get("text", "")
-            is_final = event.get("isFinal", False)
+            is_final = event_type == "transcript.final" or bool(event.get("isFinal"))
             if is_final:
                 # Clear the interim line and print final
                 print(" " * 80, end="\r")  # Clear previous interim text
-                print(f"→ You: {text}")
+                print(f"→ You: {text}{ids}")
             else:
                 # Interim result - show with indicator (overwrite same line)
                 display_text = text[:60] + "..." if len(text) > 60 else text
                 print(f"  [listening] {display_text}".ljust(80), end="\r")
-        elif event_type == "ttfb":
+        elif event_type in {"ttfb", "metrics.ttfb"}:
             # Server-side TTFB event
             latency_ms = event.get("latencyMs", 0)
             print(f"← [TTFB] Server reported latency: {latency_ms}ms")
-        elif event_type == "llmResponse":
+        elif event_type in {"llmResponse", "assistant.response.delta", "assistant.response.final"}:
             # LLM text response
             text = event.get("text", "")
-            is_final = event.get("isFinal", False)
+            is_final = event_type == "assistant.response.final" or bool(event.get("isFinal"))
             if is_final:
                 # Print final LLM response
                 print(f"← AI: {text}")
             elif self.verbose:
                 # Show streaming chunks only in verbose mode
                 display_text = text[:60] + "..." if len(text) > 60 else text
-                print(f"  [streaming] {display_text}")
+                print(f"  [streaming] {display_text}{ids}")
-        elif event_type == "trackStart":
+        elif event_type in {"trackStart", "output.audio.start"}:
-            print("← Bot started speaking")
+            print(f"← Bot started speaking{ids}")
             # IMPORTANT: Accept audio again after trackStart
             self._discard_audio = False
             self._audio_sequence += 1
@@ -342,13 +377,13 @@ class MicrophoneClient:
             # Clear any old audio in buffer
             with self.audio_output_lock:
                 self.audio_output_buffer = b""
-        elif event_type == "trackEnd":
+        elif event_type in {"trackEnd", "output.audio.end"}:
-            print("← Bot finished speaking")
+            print(f"← Bot finished speaking{ids}")
             # Reset TTFB tracking after response completes
             self.request_start_time = None
             self.first_audio_received = False
-        elif event_type == "interrupt":
+        elif event_type in {"interrupt", "response.interrupted"}:
-            print("← Bot interrupted!")
+            print(f"← Bot interrupted!{ids}")
             # IMPORTANT: Discard all audio until next trackStart
             self._discard_audio = True
             # Clear audio buffer immediately
@@ -357,12 +392,12 @@ class MicrophoneClient:
                 self.audio_output_buffer = b""
                 print(f"   (cleared {buffer_ms:.0f}ms, discarding audio until new track)")
         elif event_type == "error":
-            print(f"← Error: {event.get('error')}")
+            print(f"← Error: {event.get('error')}{ids}")
-        elif event_type == "hangup":
+        elif event_type in {"hangup", "session.stopped"}:
-            print(f"← Hangup: {event.get('reason')}")
+            print(f"← Hangup: {event.get('reason')}{ids}")
             self.running = False
         else:
-            print(f"← Event: {event_type}")
+            print(f"← Event: {event_type}{ids}")
     
     async def interactive_mode(self) -> None:
         """Run interactive mode for text chat."""
@@ -573,6 +608,26 @@ async def main():
         action="store_true",
         help="Show streaming LLM response chunks"
     )
+    parser.add_argument(
+        "--app-id",
+        default="assistant_demo",
+        help="Stable app/assistant identifier for server-side config lookup"
+    )
+    parser.add_argument(
+        "--channel",
+        default="mic_client",
+        help="Client channel name"
+    )
+    parser.add_argument(
+        "--config-version-id",
+        default="local-dev",
+        help="Optional config version identifier"
+    )
+    parser.add_argument(
+        "--track-debug",
+        action="store_true",
+        help="Print event trackId for protocol debugging"
+    )
     
     args = parser.parse_args()
     
@@ -583,8 +638,12 @@ async def main():
     client = MicrophoneClient(
         url=args.url,
         sample_rate=args.sample_rate,
+        app_id=args.app_id,
+        channel=args.channel,
+        config_version_id=args.config_version_id,
         input_device=args.input_device,
-        output_device=args.output_device
+        output_device=args.output_device,
+        track_debug=args.track_debug,
     )
     client.verbose = args.verbose
     

examples/simple_client.py

@@ -52,9 +52,21 @@ if not PYAUDIO_AVAILABLE and not SD_AVAILABLE:
 class SimpleVoiceClient:
     """Simple voice client with reliable audio playback."""
     
-    def __init__(self, url: str, sample_rate: int = 16000):
+    def __init__(
+        self,
+        url: str,
+        sample_rate: int = 16000,
+        app_id: str = "assistant_demo",
+        channel: str = "simple_client",
+        config_version_id: str = "local-dev",
+        track_debug: bool = False,
+    ):
         self.url = url
         self.sample_rate = sample_rate
+        self.app_id = app_id
+        self.channel = channel
+        self.config_version_id = config_version_id
+        self.track_debug = track_debug
         self.ws = None
         self.running = False
         
@@ -75,6 +87,17 @@ class SimpleVoiceClient:
         
         # Interrupt handling - discard audio until next trackStart
         self._discard_audio = False
+
+    @staticmethod
+    def _event_ids_suffix(event: dict) -> str:
+        data = event.get("data") if isinstance(event.get("data"), dict) else {}
+        keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
+        parts = []
+        for key in keys:
+            value = data.get(key, event.get(key))
+            if value:
+                parts.append(f"{key}={value}")
+        return f" [{' '.join(parts)}]" if parts else ""
     
     async def connect(self):
         """Connect to server."""
@@ -83,12 +106,25 @@ class SimpleVoiceClient:
         self.running = True
         print("Connected!")
         
-        # Send invite
+        # WS v1 handshake: hello -> session.start
         await self.ws.send(json.dumps({
-            "command": "invite",
+            "type": "hello",
-            "option": {"codec": "pcm", "sampleRate": self.sample_rate}
+            "version": "v1",
         }))
-        print("-> invite")
+        await self.ws.send(json.dumps({
+            "type": "session.start",
+            "audio": {
+                "encoding": "pcm_s16le",
+                "sample_rate_hz": self.sample_rate,
+                "channels": 1,
+            },
+            "metadata": {
+                "appId": self.app_id,
+                "channel": self.channel,
+                "configVersionId": self.config_version_id,
+            },
+        }))
+        print("-> hello/session.start")
     
     async def send_chat(self, text: str):
         """Send chat message."""
@@ -96,8 +132,8 @@ class SimpleVoiceClient:
         self.request_start_time = time.time()
         self.first_audio_received = False
         
-        await self.ws.send(json.dumps({"command": "chat", "text": text}))
+        await self.ws.send(json.dumps({"type": "input.text", "text": text}))
-        print(f"-> chat: {text}")
+        print(f"-> input.text: {text}")
     
     def play_audio(self, audio_data: bytes):
         """Play audio data immediately."""
@@ -152,34 +188,39 @@ class SimpleVoiceClient:
                 else:
                     # JSON event
                     event = json.loads(msg)
-                    etype = event.get("event", "?")
+                    etype = event.get("type", event.get("event", "?"))
+                    ids = self._event_ids_suffix(event)
+                    if self.track_debug:
+                        print(f"[track-debug] event={etype} trackId={event.get('trackId')}{ids}")
                     
-                    if etype == "transcript":
+                    if etype in {"transcript", "transcript.delta", "transcript.final"}:
                         # User speech transcription
                         text = event.get("text", "")
-                        is_final = event.get("isFinal", False)
+                        is_final = etype == "transcript.final" or bool(event.get("isFinal"))
                         if is_final:
-                            print(f"<- You said: {text}")
+                            print(f"<- You said: {text}{ids}")
                         else:
                             print(f"<- [listening] {text}", end="\r")
-                    elif etype == "ttfb":
+                    elif etype in {"ttfb", "metrics.ttfb"}:
                         # Server-side TTFB event
                         latency_ms = event.get("latencyMs", 0)
                         print(f"<- [TTFB] Server reported latency: {latency_ms}ms")
-                    elif etype == "trackStart":
+                    elif etype in {"trackStart", "output.audio.start"}:
                         # New track starting - accept audio again
                         self._discard_audio = False
-                        print(f"<- {etype}")
+                        print(f"<- {etype}{ids}")
-                    elif etype == "interrupt":
+                    elif etype in {"interrupt", "response.interrupted"}:
                         # Interrupt - discard audio until next trackStart
                         self._discard_audio = True
-                        print(f"<- {etype} (discarding audio until new track)")
+                        print(f"<- {etype}{ids} (discarding audio until new track)")
-                    elif etype == "hangup":
+                    elif etype in {"hangup", "session.stopped"}:
-                        print(f"<- {etype}")
+                        print(f"<- {etype}{ids}")
                         self.running = False
                         break
+                    elif etype == "config.resolved":
+                        print(f"<- config.resolved {event.get('config', {}).get('output', {})}{ids}")
                     else:
-                        print(f"<- {etype}")
+                        print(f"<- {etype}{ids}")
                         
             except asyncio.TimeoutError:
                 continue
@@ -270,6 +311,10 @@ async def main():
     parser.add_argument("--text", help="Send text and play response")
     parser.add_argument("--list-devices", action="store_true")
     parser.add_argument("--sample-rate", type=int, default=16000)
+    parser.add_argument("--app-id", default="assistant_demo")
+    parser.add_argument("--channel", default="simple_client")
+    parser.add_argument("--config-version-id", default="local-dev")
+    parser.add_argument("--track-debug", action="store_true")
     
     args = parser.parse_args()
     
@@ -277,7 +322,14 @@ async def main():
         list_audio_devices()
         return
     
-    client = SimpleVoiceClient(args.url, args.sample_rate)
+    client = SimpleVoiceClient(
+        args.url,
+        args.sample_rate,
+        app_id=args.app_id,
+        channel=args.channel,
+        config_version_id=args.config_version_id,
+        track_debug=args.track_debug,
+    )
     await client.run(args.text)
 
 

examples/test_websocket.py

@@ -36,8 +36,18 @@ def generate_sine_wave(duration_ms=1000):
     return audio_data
 
 
-async def receive_loop(ws, ready_event: asyncio.Event):
+async def receive_loop(ws, ready_event: asyncio.Event, track_debug: bool = False):
     """Listen for incoming messages from the server."""
+    def event_ids_suffix(data):
+        payload = data.get("data") if isinstance(data.get("data"), dict) else {}
+        keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
+        parts = []
+        for key in keys:
+            value = payload.get(key, data.get(key))
+            if value:
+                parts.append(f"{key}={value}")
+        return f" [{' '.join(parts)}]" if parts else ""
+
     print("👂 Listening for server responses...")
     async for msg in ws:
         timestamp = datetime.now().strftime("%H:%M:%S")
@@ -46,7 +56,10 @@ async def receive_loop(ws, ready_event: asyncio.Event):
             try:
                 data = json.loads(msg.data)
                 event_type = data.get('type', 'Unknown')
-                print(f"[{timestamp}] 📨 Event: {event_type} | {msg.data[:150]}...")
+                ids = event_ids_suffix(data)
+                print(f"[{timestamp}] 📨 Event: {event_type}{ids} | {msg.data[:150]}...")
+                if track_debug:
+                    print(f"[{timestamp}] [track-debug] event={event_type} trackId={data.get('trackId')}{ids}")
                 if event_type == "session.started":
                     ready_event.set()
             except json.JSONDecodeError:
@@ -113,7 +126,7 @@ async def send_sine_loop(ws):
     print("\n✅ Finished streaming test audio.")
 
 
-async def run_client(url, file_path=None, use_sine=False):
+async def run_client(url, file_path=None, use_sine=False, track_debug: bool = False):
     """Run the WebSocket test client."""
     session = aiohttp.ClientSession()
     try:
@@ -121,7 +134,7 @@ async def run_client(url, file_path=None, use_sine=False):
         async with session.ws_connect(url) as ws:
             print("✅ Connected!")
             session_ready = asyncio.Event()
-            recv_task = asyncio.create_task(receive_loop(ws, session_ready))
+            recv_task = asyncio.create_task(receive_loop(ws, session_ready, track_debug=track_debug))
 
             # Send v1 hello + session.start handshake
             await ws.send_json({"type": "hello", "version": "v1"})
@@ -131,7 +144,12 @@ async def run_client(url, file_path=None, use_sine=False):
                     "encoding": "pcm_s16le",
                     "sample_rate_hz": SAMPLE_RATE,
                     "channels": 1
-                }
+                },
+                "metadata": {
+                    "appId": "assistant_demo",
+                    "channel": "test_websocket",
+                    "configVersionId": "local-dev",
+                },
             })
             print("📤 Sent v1 hello/session.start")
             await asyncio.wait_for(session_ready.wait(), timeout=8)
@@ -168,9 +186,10 @@ if __name__ == "__main__":
     parser.add_argument("--url", default=SERVER_URL, help="WebSocket endpoint URL")
     parser.add_argument("--file", help="Path to PCM/WAV file to stream")
     parser.add_argument("--sine", action="store_true", help="Use sine wave generation (default)")
+    parser.add_argument("--track-debug", action="store_true", help="Print event trackId for protocol debugging")
     args = parser.parse_args()
 
     try:
-        asyncio.run(run_client(args.url, args.file, args.sine))
+        asyncio.run(run_client(args.url, args.file, args.sine, args.track_debug))
     except KeyboardInterrupt:
         print("\n👋 Client stopped.")

examples/wav_client.py

@@ -57,10 +57,15 @@ class WavFileClient:
         url: str,
         input_file: str,
         output_file: str,
+        app_id: str = "assistant_demo",
+        channel: str = "wav_client",
+        config_version_id: str = "local-dev",
         sample_rate: int = 16000,
         chunk_duration_ms: int = 20,
         wait_time: float = 15.0,
-        verbose: bool = False
+        verbose: bool = False,
+        track_debug: bool = False,
+        tail_silence_ms: int = 800,
     ):
         """
         Initialize WAV file client.
@@ -77,11 +82,17 @@ class WavFileClient:
         self.url = url
         self.input_file = Path(input_file)
         self.output_file = Path(output_file)
+        self.app_id = app_id
+        self.channel = channel
+        self.config_version_id = config_version_id
         self.sample_rate = sample_rate
         self.chunk_duration_ms = chunk_duration_ms
         self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
         self.wait_time = wait_time
         self.verbose = verbose
+        self.track_debug = track_debug
+        self.tail_silence_ms = max(0, int(tail_silence_ms))
+        self.frame_bytes = 640  # 16k mono pcm_s16le, 20ms
         
         # WebSocket connection
         self.ws = None
@@ -125,6 +136,17 @@ class WavFileClient:
             # Replace problematic characters for console output
             safe_message = message.encode('ascii', errors='replace').decode('ascii')
             print(f"{direction} {safe_message}")
+
+    @staticmethod
+    def _event_ids_suffix(event: dict) -> str:
+        data = event.get("data") if isinstance(event.get("data"), dict) else {}
+        keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
+        parts = []
+        for key in keys:
+            value = data.get(key, event.get(key))
+            if value:
+                parts.append(f"{key}={value}")
+        return f" [{' '.join(parts)}]" if parts else ""
     
     async def connect(self) -> None:
         """Connect to WebSocket server."""
@@ -144,7 +166,12 @@ class WavFileClient:
                 "encoding": "pcm_s16le",
                 "sample_rate_hz": self.sample_rate,
                 "channels": 1
-            }
+            },
+            "metadata": {
+                "appId": self.app_id,
+                "channel": self.channel,
+                "configVersionId": self.config_version_id,
+            },
         })
     
     async def send_command(self, cmd: dict) -> None:
@@ -216,6 +243,10 @@ class WavFileClient:
             end_sample = min(sent_samples + chunk_size, total_samples)
             chunk = audio_data[sent_samples:end_sample]
             chunk_bytes = chunk.tobytes()
+            if len(chunk_bytes) % self.frame_bytes != 0:
+                # v1 audio framing requires 640-byte (20ms) PCM units.
+                pad = self.frame_bytes - (len(chunk_bytes) % self.frame_bytes)
+                chunk_bytes += b"\x00" * pad
             
             # Send to server
             if self.ws:
@@ -232,6 +263,16 @@ class WavFileClient:
             # Delay to simulate real-time streaming
             # Server expects audio at real-time pace for VAD/ASR to work properly
             await asyncio.sleep(self.chunk_duration_ms / 1000)
+
+        # Add a short silence tail to help VAD/EOU close the final utterance.
+        if self.tail_silence_ms > 0 and self.ws:
+            tail_frames = max(1, self.tail_silence_ms // 20)
+            silence = b"\x00" * self.frame_bytes
+            for _ in range(tail_frames):
+                await self.ws.send(silence)
+                self.bytes_sent += len(silence)
+                await asyncio.sleep(0.02)
+            self.log_event("→", f"Sent trailing silence: {self.tail_silence_ms}ms")
         
         self.send_completed = True
         elapsed = time.time() - self.send_start_time
@@ -284,16 +325,22 @@ class WavFileClient:
     async def _handle_event(self, event: dict) -> None:
         """Handle incoming event."""
         event_type = event.get("type", "unknown")
+        ids = self._event_ids_suffix(event)
+        if self.track_debug:
+            print(f"[track-debug] event={event_type} trackId={event.get('trackId')}{ids}")
 
         if event_type == "hello.ack":
-            self.log_event("←", "Handshake acknowledged")
+            self.log_event("←", f"Handshake acknowledged{ids}")
         elif event_type == "session.started":
             self.session_ready = True
-            self.log_event("←", "Session ready!")
+            self.log_event("←", f"Session ready!{ids}")
+        elif event_type == "config.resolved":
+            config = event.get("config", {})
+            self.log_event("←", f"Config resolved (output={config.get('output', {})}){ids}")
         elif event_type == "input.speech_started":
-            self.log_event("←", "Speech detected")
+            self.log_event("←", f"Speech detected{ids}")
         elif event_type == "input.speech_stopped":
-            self.log_event("←", "Silence detected")
+            self.log_event("←", f"Silence detected{ids}")
         elif event_type == "transcript.delta":
             text = event.get("text", "")
             display_text = text[:60] + "..." if len(text) > 60 else text
@@ -301,35 +348,35 @@ class WavFileClient:
         elif event_type == "transcript.final":
             text = event.get("text", "")
             print(" " * 80, end="\r")
-            self.log_event("←", f"→ You: {text}")
+            self.log_event("←", f"→ You: {text}{ids}")
         elif event_type == "metrics.ttfb":
             latency_ms = event.get("latencyMs", 0)
             self.log_event("←", f"[TTFB] Server latency: {latency_ms}ms")
         elif event_type == "assistant.response.delta":
             text = event.get("text", "")
             if self.verbose and text:
-                self.log_event("←", f"LLM: {text}")
+                self.log_event("←", f"LLM: {text}{ids}")
         elif event_type == "assistant.response.final":
             text = event.get("text", "")
             if text:
-                self.log_event("←", f"LLM Response (final): {text[:100]}{'...' if len(text) > 100 else ''}")
+                self.log_event("←", f"LLM Response (final): {text[:100]}{'...' if len(text) > 100 else ''}{ids}")
         elif event_type == "output.audio.start":
             self.track_started = True
             self.response_start_time = time.time()
             self.waiting_for_first_audio = True
-            self.log_event("←", "Bot started speaking")
+            self.log_event("←", f"Bot started speaking{ids}")
         elif event_type == "output.audio.end":
             self.track_ended = True
-            self.log_event("←", "Bot finished speaking")
+            self.log_event("←", f"Bot finished speaking{ids}")
         elif event_type == "response.interrupted":
-            self.log_event("←", "Bot interrupted!")
+            self.log_event("←", f"Bot interrupted!{ids}")
         elif event_type == "error":
-            self.log_event("!", f"Error: {event.get('message')}")
+            self.log_event("!", f"Error: {event.get('message')}{ids}")
         elif event_type == "session.stopped":
-            self.log_event("←", f"Session stopped: {event.get('reason')}")
+            self.log_event("←", f"Session stopped: {event.get('reason')}{ids}")
             self.running = False
         else:
-            self.log_event("←", f"Event: {event_type}")
+            self.log_event("←", f"Event: {event_type}{ids}")
     
     def save_output_wav(self) -> None:
         """Save received audio to output WAV file."""
@@ -473,6 +520,21 @@ async def main():
         default=16000,
         help="Target sample rate for audio (default: 16000)"
     )
+    parser.add_argument(
+        "--app-id",
+        default="assistant_demo",
+        help="Stable app/assistant identifier for server-side config lookup"
+    )
+    parser.add_argument(
+        "--channel",
+        default="wav_client",
+        help="Client channel name"
+    )
+    parser.add_argument(
+        "--config-version-id",
+        default="local-dev",
+        help="Optional config version identifier"
+    )
     parser.add_argument(
         "--chunk-duration",
         type=int,
@@ -490,6 +552,17 @@ async def main():
         action="store_true",
         help="Enable verbose output"
     )
+    parser.add_argument(
+        "--track-debug",
+        action="store_true",
+        help="Print event trackId for protocol debugging"
+    )
+    parser.add_argument(
+        "--tail-silence-ms",
+        type=int,
+        default=800,
+        help="Trailing silence to send after WAV playback for EOU detection (default: 800)"
+    )
     
     args = parser.parse_args()
     
@@ -497,10 +570,15 @@ async def main():
         url=args.url,
         input_file=args.input,
         output_file=args.output,
+        app_id=args.app_id,
+        channel=args.channel,
+        config_version_id=args.config_version_id,
         sample_rate=args.sample_rate,
         chunk_duration_ms=args.chunk_duration,
         wait_time=args.wait_time,
-        verbose=args.verbose
+        verbose=args.verbose,
+        track_debug=args.track_debug,
+        tail_silence_ms=args.tail_silence_ms,
     )
     
     await client.run()

examples/web_client.html

@@ -401,6 +401,9 @@
 
       const targetSampleRate = 16000;
       const playbackStopRampSec = 0.008;
+      const appId = "assistant_demo";
+      const channel = "web_client";
+      const configVersionId = "local-dev";
 
       function logLine(type, text, data) {
         const time = new Date().toLocaleTimeString();
@@ -604,15 +607,35 @@
         logLine("sys", `→ ${cmd.type}`, cmd);
       }
 
+      function eventIdsSuffix(event) {
+        const data = event && typeof event.data === "object" && event.data ? event.data : {};
+        const keys = ["turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id"];
+        const parts = [];
+        for (const key of keys) {
+          const value = data[key] || event[key];
+          if (value) parts.push(`${key}=${value}`);
+        }
+        return parts.length ? ` [${parts.join(" ")}]` : "";
+      }
+
       function handleEvent(event) {
         const type = event.type || "unknown";
-        logLine("event", type, event);
+        const ids = eventIdsSuffix(event);
+        logLine("event", `${type}${ids}`, event);
         if (type === "hello.ack") {
           sendCommand({
             type: "session.start",
             audio: { encoding: "pcm_s16le", sample_rate_hz: targetSampleRate, channels: 1 },
+            metadata: {
+              appId,
+              channel,
+              configVersionId,
+            },
           });
         }
+        if (type === "config.resolved") {
+          logLine("sys", "config.resolved", event.config || {});
+        }
         if (type === "transcript.final") {
           if (event.text) {
             setInterim("You", "");