backend/lm-livekit-helper.git

			@@ -1,7 +1,10 @@
			mod asr_realtime;
			mod audio;
			mod service;

			use std::{
			borrow::Cow,
			collections::HashSet,
			env, fs,
			path::{Path, PathBuf},
			sync::{
			@@ -12,7 +15,10 @@
			};

			use anyhow::{Context, Result, anyhow};
			use audio::{AudioDiagnostics, PcmFrame, load_pre_recorded_frames};
			use asr_realtime::{
			AudioIngressMetadata, RealtimeAsrConfig, RealtimeAsrOutcome, RealtimeAsrUpload,
			};
			use audio::{AudioDiagnostics, load_pre_recorded_frames};
			use base64::{Engine as _, engine::general_purpose};
			use futures_util::StreamExt;
			use libwebrtc::{
			@@ -24,19 +30,35 @@
			options::TrackPublishOptions,
			prelude::{
			DataPacket, LocalAudioTrack, LocalTrack, ParticipantIdentity, RemoteAudioTrack,
			RemoteTrack, Room, RoomEvent, RoomOptions,
			RemoteParticipant, RemoteTrack, Room, RoomEvent, RoomOptions,
			},
			};
			use reqwest::Client;
			use serde::{Deserialize, Serialize};
			use serde_json::json;
			use tokio::time::{sleep, sleep_until};
			use tokio::{sync::mpsc::UnboundedReceiver, task::JoinHandle};
			use sha2::{Digest, Sha256};
			use tokio::time::{sleep, sleep_until, timeout};
			use tokio::{
			sync::{mpsc, mpsc::UnboundedReceiver, watch},
			task::JoinHandle,
			};
			use tracing::{info, warn};

			const TARGET_SAMPLE_RATE_HZ: u32 = 48_000;
			const TARGET_NUM_CHANNELS: u16 = 1;
			const USER_AUDIO_SAMPLE_RATE_HZ: u32 = 48_000;
			const USER_AUDIO_NUM_CHANNELS: u16 = 1;
			const INBOUND_AUDIO_QUEUE_CAPACITY: usize = 100;
			const INBOUND_AUDIO_DROP_LOG_INTERVAL: u64 = 100;
			const AUDIO_DRAIN_STOP_GRACE: Duration = Duration::from_secs(2);
			const DEFAULT_BOT_AUDIO_PROFILE: &str = "pcm-16k";
			const LIVEKIT_48K_SAMPLE_RATE_HZ: u32 = 48_000;
			const PCM_16K_SAMPLE_RATE_HZ: u32 = 16_000;
			const BOT_NUM_CHANNELS: u16 = 1;
			const TRACK_NAME: &str = "bot-main-audio";
			const STREAM_TIMING_VERSION: u32 = 1;
			const STREAM_TIMING_FIRST_SEGMENT: u64 = 1;
			const STREAM_TIMING_FIRST_CHUNK: u64 = 1;
			const STREAM_TIMING_MAX_ELAPSED_MS: u64 = 5_000;
			const STREAM_TIMING_SOURCE: &str = "stream_anchor_monotonic";

			#[tokio::main(flavor = "multi_thread")]
			async fn main() -> Result<()> {
			@@ -58,8 +80,8 @@
			&http,
			config.greeting_audio_file.as_deref(),
			config.greeting_audio_url.as_deref(),
			TARGET_SAMPLE_RATE_HZ,
			TARGET_NUM_CHANNELS,
			config.bot_audio_profile.sample_rate_hz,
			config.bot_audio_profile.num_channels,
			config.audio_debug_dump_dir.as_deref(),
			&config.call_id,
			"greeting",
			@@ -81,6 +103,9 @@
			room_alias = %redact(&config.room_id),
			participant_alias = %redact(&config.bot_participant_identity),
			greeting_source = %config.greeting_source,
			bot_audio_profile = %config.bot_audio_profile.profile,
			bot_sample_rate_hz = config.bot_audio_profile.sample_rate_hz,
			bot_num_channels = config.bot_audio_profile.num_channels,
			"combrabo voice runtime helper connected"
			);
			emit_activity(
			@@ -129,8 +154,9 @@
			&config.call_id,
			&config.trace_id,
			TRACK_NAME,
			TARGET_SAMPLE_RATE_HZ,
			u32::from(TARGET_NUM_CHANNELS),
			config.bot_audio_profile.profile.clone(),
			config.bot_audio_profile.sample_rate_hz,
			u32::from(config.bot_audio_profile.num_channels),
			config.user_participant_identity.clone(),
			)
			.await?;
			@@ -262,6 +288,11 @@
			runtime_turn_bridge_url: Option<String>,
			runtime_turn_bridge_token: Option<String>,
			runtime_turn_bridge_mode: String,
			runtime_asr_stream_enabled: bool,
			runtime_asr_stream_url: Option<String>,
			runtime_asr_realtime_enabled: bool,
			runtime_asr_realtime_url: Option<String>,
			runtime_asr_realtime_chunk_duration_ms: u64,
			runtime_turn_artifact_dir: Option<String>,
			runtime_session_nonce: Option<String>,
			user_audio_observer_enabled: bool,
			@@ -269,6 +300,54 @@
			simple_vad_gate_until_greeting_done: bool,
			simple_vad_post_greeting_delay_ms: u64,
			simple_vad_config: SimpleVadConfig,
			bot_audio_profile: BotAudioProfile,
			}

			#[derive(Clone)]
			struct BotAudioProfile {
			profile: String,
			sample_rate_hz: u32,
			num_channels: u16,
			}

			impl BotAudioProfile {
			fn from_env() -> Result<Self> {
			let profile = env::var("CV_BOT_AUDIO_PROFILE")
			.unwrap_or_else(\|_\| DEFAULT_BOT_AUDIO_PROFILE.to_string())
			.trim()
			.to_ascii_lowercase();
			match profile.as_str() {
			"livekit-48k" \| "48k" => Ok(Self {
			profile: "livekit-48k".to_string(),
			sample_rate_hz: LIVEKIT_48K_SAMPLE_RATE_HZ,
			num_channels: BOT_NUM_CHANNELS,
			}),
			"pcm-16k" \| "16k" => Ok(Self {
			profile: "pcm-16k".to_string(),
			sample_rate_hz: PCM_16K_SAMPLE_RATE_HZ,
			num_channels: BOT_NUM_CHANNELS,
			}),
			"custom" => {
			let sample_rate_hz = u32_env("CV_BOT_SAMPLE_RATE_HZ", LIVEKIT_48K_SAMPLE_RATE_HZ);
			let num_channels = u16_env("CV_BOT_NUM_CHANNELS", BOT_NUM_CHANNELS);
			if sample_rate_hz == 0 {
			return Err(anyhow!("CV_BOT_SAMPLE_RATE_HZ must be positive"));
			}
			if num_channels == 0 {
			return Err(anyhow!("CV_BOT_NUM_CHANNELS must be positive"));
			}
			Ok(Self {
			profile,
			sample_rate_hz,
			num_channels,
			})
			}
			_ => Err(anyhow!(
			"unsupported CV_BOT_AUDIO_PROFILE {}; expected livekit-48k, pcm-16k or custom",
			profile
			)),
			}
			}
			}

			#[derive(Clone)]
			@@ -276,6 +355,11 @@
			bridge_url: Option<String>,
			bridge_token: Option<String>,
			bridge_mode: String,
			asr_stream_enabled: bool,
			asr_stream_url: Option<String>,
			asr_realtime_enabled: bool,
			asr_realtime_url: Option<String>,
			asr_realtime_chunk_duration_ms: u64,
			artifact_dir: Option<String>,
			runtime_session_nonce: Option<String>,
			audio_debug_dump_dir: Option<String>,
			@@ -287,6 +371,11 @@
			bridge_url: config.runtime_turn_bridge_url.clone(),
			bridge_token: config.runtime_turn_bridge_token.clone(),
			bridge_mode: config.runtime_turn_bridge_mode.clone(),
			asr_stream_enabled: config.runtime_asr_stream_enabled,
			asr_stream_url: config.runtime_asr_stream_url.clone(),
			asr_realtime_enabled: config.runtime_asr_realtime_enabled,
			asr_realtime_url: config.runtime_asr_realtime_url.clone(),
			asr_realtime_chunk_duration_ms: config.runtime_asr_realtime_chunk_duration_ms,
			artifact_dir: config.runtime_turn_artifact_dir.clone(),
			runtime_session_nonce: config.runtime_session_nonce.clone(),
			audio_debug_dump_dir: config.audio_debug_dump_dir.clone(),
			@@ -318,6 +407,32 @@
			.as_deref()
			.is_some_and(\|value\| value.trim_end_matches('/').ends_with("/stream"))
			}

			fn is_asr_stream_ready(&self) -> bool {
			self.asr_stream_enabled
			&& self
			.asr_stream_url
			.as_ref()
			.is_some_and(\|value\| !value.is_empty())
			&& self
			.bridge_token
			.as_ref()
			.is_some_and(\|value\| !value.is_empty())
			&& self
			.runtime_session_nonce
			.as_ref()
			.is_some_and(\|value\| !value.is_empty())
			}

			fn realtime_asr_config(&self) -> RealtimeAsrConfig {
			RealtimeAsrConfig {
			enabled: self.asr_realtime_enabled,
			url: self.asr_realtime_url.clone(),
			runtime_token: self.bridge_token.clone(),
			runtime_session_nonce: self.runtime_session_nonce.clone(),
			chunk_duration_ms: self.asr_realtime_chunk_duration_ms,
			}
			}
			}

			impl Config {
			@@ -339,6 +454,14 @@
			runtime_turn_bridge_token: optional_env("CV_RUNTIME_TURN_BRIDGE_TOKEN"),
			runtime_turn_bridge_mode: env::var("CV_RUNTIME_TURN_BRIDGE_MODE")
			.unwrap_or_else(\|_\| "json".to_string()),
			runtime_asr_stream_enabled: bool_env("CV_RUNTIME_ASR_STREAM_ENABLED", false),
			runtime_asr_stream_url: optional_env("CV_RUNTIME_ASR_STREAM_URL"),
			runtime_asr_realtime_enabled: bool_env("CV_RUNTIME_ASR_REALTIME_ENABLED", false),
			runtime_asr_realtime_url: optional_env("CV_RUNTIME_ASR_REALTIME_URL"),
			runtime_asr_realtime_chunk_duration_ms: u64_env(
			"CV_RUNTIME_ASR_REALTIME_CHUNK_DURATION_MS",
			200,
			),
			runtime_turn_artifact_dir: optional_env("CV_RUNTIME_TURN_ARTIFACT_DIR"),
			runtime_session_nonce: optional_env("CV_RUNTIME_SESSION_NONCE"),
			user_audio_observer_enabled: bool_env("CV_ENABLE_USER_AUDIO_OBSERVER", true),
			@@ -346,6 +469,7 @@
			simple_vad_gate_until_greeting_done: bool_env("CV_VAD_GATE_UNTIL_GREETING_DONE", true),
			simple_vad_post_greeting_delay_ms: u64_env("CV_VAD_POST_GREETING_DELAY_MS", 800),
			simple_vad_config: SimpleVadConfig::from_env(),
			bot_audio_profile: BotAudioProfile::from_env()?,
			})
			}
			}
			@@ -492,6 +616,13 @@
			.unwrap_or(default_value)
			}

			fn u16_env(key: &str, default_value: u16) -> u16 {
			env::var(key)
			.ok()
			.and_then(\|value\| value.trim().parse::<u16>().ok())
			.unwrap_or(default_value)
			}

			fn redact(value: &str) -> String {
			if value.len() <= 8 {
			return "redacted".to_string();
			@@ -532,10 +663,36 @@
			"traceId": trace_id,
			"turnId": turn_id,
			"eventName": event_name,
			"eventWallTimeMs": current_time_millis(),
			"result": result,
			"reasonCode": reason_code,
			"retryable": retryable,
			"extension": extension,
			});
			println!("{payload}");
			}

			fn emit_anchored_activity(
			call_id: &str,
			trace_id: &str,
			turn_id: &str,
			event_name: &str,
			marker: &RuntimeTurnStreamTimingMarker,
			extension: serde_json::Value,
			) {
			let payload = json!({
			"type": "cv_activity",
			"callId": call_id,
			"traceId": trace_id,
			"turnId": turn_id,
			"eventName": event_name,
			"eventWallTimeMs": current_time_millis(),
			"serverDeltaMs": marker.server_delta_ms,
			"serverDeltaSource": STREAM_TIMING_SOURCE,
			"result": "ok",
			"reasonCode": null,
			"retryable": null,
			"extension": marker.extension_with(extension),
			});
			println!("{payload}");
			}
			@@ -552,6 +709,7 @@
			let enabled = config.user_audio_observer_enabled;
			let simple_vad_enabled = config.simple_vad_enabled;
			let simple_vad_config = config.simple_vad_config.clone();
			let user_participant_identity = config.user_participant_identity.clone();
			let turn_bridge_config = TurnBridgeConfig::from_config(config);

			tokio::spawn(async move {
			@@ -573,9 +731,14 @@
			turn_bridge_config,
			http,
			sink,
			user_participant_identity,
			)
			.await;
			})
			}

			fn is_bound_user_participant(identity: &str, expected: Option<&str>) -> bool {
			expected.is_none_or(\|value\| identity == value)
			}

			async fn observe_user_audio_events(
			@@ -588,6 +751,7 @@
			turn_bridge_config: TurnBridgeConfig,
			http: Client,
			sink: Arc<BotAudioOutputSink>,
			user_participant_identity: Option<String>,
			) {
			info!(
			call_id = %call_id,
			@@ -611,6 +775,15 @@
			publication: _,
			participant,
			} => {
			if !is_bound_user_participant(
			&participant.identity().to_string(),
			user_participant_identity.as_deref(),
			) {
			warn!(call_id = %call_id, trace_id = %trace_id,
			metadata_status = "wrong_participant",
			"runtime helper ignored non-user audio participant");
			continue;
			}
			let participant_alias = redact(&participant.identity().to_string());
			let track_sid_alias = redact(&track.sid().to_string());
			let track_name = track.name();
			@@ -636,6 +809,7 @@
			turn_bridge_config.clone(),
			http.clone(),
			sink.clone(),
			participant,
			);
			}
			RoomEvent::TrackSubscribed {
			@@ -686,6 +860,7 @@
			call_id: &str,
			trace_id: &str,
			turn: FinishedSpeechTurn,
			realtime_asr_result_ref: Option<String>,
			) {
			let turn_pipeline_started_at = Instant::now();
			if !bridge_config.is_ready() {
			@@ -742,9 +917,13 @@
			"turnArtifactBytes": byte_size,
			"frameCount": turn.frame_count,
			"sampleCount": turn.sample_count,
			"endReason": turn.end_reason,
			"endReason": turn.end_reason.as_str(),
			}),
			);
			let asr_result_ref = match realtime_asr_result_ref {
			Some(value) => Some(value),
			None => request_asr_result_ref(http, bridge_config, call_id, trace_id, &turn).await,
			};
			if bridge_config.is_stream_mode() {
			match request_turn_bridge_stream(
			http,
			@@ -755,6 +934,7 @@
			&turn,
			&path_ref,
			byte_size,
			asr_result_ref.as_deref(),
			turn_pipeline_started_at,
			)
			.await
			@@ -818,12 +998,17 @@
			&turn,
			&path_ref,
			byte_size,
			asr_result_ref.as_deref(),
			turn_pipeline_started_at,
			)
			.await
			{
			Ok(outcome) => {
			let mut published_device_outputs = HashSet::new();
			for output in &outcome.device_outputs {
			if !should_publish_device_output(&mut published_device_outputs, output) {
			continue;
			}
			if let Err(error) = sink
			.publish_device_output(call_id, trace_id, &turn.turn_id, output)
			.await
			@@ -930,13 +1115,258 @@
			audio::write_pcm_wav(
			&output_path,
			&turn.samples,
			TARGET_SAMPLE_RATE_HZ,
			TARGET_NUM_CHANNELS,
			USER_AUDIO_SAMPLE_RATE_HZ,
			USER_AUDIO_NUM_CHANNELS,
			)?;
			let byte_size = fs::metadata(&output_path)
			.context("failed to stat turn artifact")?
			.len();
			Ok((path_ref, byte_size))
			}

			async fn request_asr_result_ref(
			http: &Client,
			bridge_config: &TurnBridgeConfig,
			call_id: &str,
			trace_id: &str,
			turn: &FinishedSpeechTurn,
			) -> Option<String> {
			if !bridge_config.is_asr_stream_ready() {
			return None;
			}
			match request_asr_stream(http, bridge_config, call_id, trace_id, turn).await {
			Ok(Some(asr_result_ref)) => {
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"asr_stream_ref_ready",
			"ok",
			None,
			None,
			json!({
			"asrResultRefPresent": true,
			"format": "pcm_s16le",
			"sampleRate": 16000,
			"channels": 1,
			}),
			);
			Some(asr_result_ref)
			}
			Ok(None) => None,
			Err(error) => {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			error = %safe_error(&error.to_string()),
			"runtime helper asr_stream_failed_fallback"
			);
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"asr_stream_fallback",
			"skipped",
			Some("ASR_STREAM_INTERRUPTED"),
			Some(true),
			json!({
			"fallbackReason": "asr_stream_request_failed",
			"fallbackStage": "asr_stream",
			}),
			);
			None
			}
			}
			}

			async fn request_asr_stream(
			http: &Client,
			bridge_config: &TurnBridgeConfig,
			call_id: &str,
			trace_id: &str,
			turn: &FinishedSpeechTurn,
			) -> Result<Option<String>> {
			let started_at = Instant::now();
			let ndjson = build_asr_stream_ndjson(call_id, trace_id, turn, bridge_config)?;
			let response = http
			.post(bridge_config.asr_stream_url.as_deref().unwrap_or_default())
			.header("Content-Type", "application/x-ndjson")
			.header(
			"X-CV-Runtime-Token",
			bridge_config.bridge_token.as_deref().unwrap_or_default(),
			)
			.header("X-CV-Call-Id", call_id)
			.header("X-CV-Trace-Id", trace_id)
			.header(
			"X-CV-Runtime-Session-Nonce",
			bridge_config
			.runtime_session_nonce
			.as_deref()
			.unwrap_or_default(),
			)
			.body(ndjson)
			.send()
			.await
			.context("failed to post asr stream")?;
			let status = response.status();
			if !status.is_success() {
			let body_len = response.text().await.map(\|body\| body.len()).unwrap_or(0);
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			http_status = status.as_u16(),
			body_len,
			"runtime helper asr_stream_http_failed"
			);
			return Ok(None);
			}
			let body: RuntimeTurnCommonResult<RuntimeAsrStreamResp> = response
			.json()
			.await
			.context("failed to decode asr stream response")?;
			if body.code != 0 {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			code = body.code,
			msg_len = body.msg.as_deref().unwrap_or_default().len(),
			"runtime helper asr_stream_common_result_failed"
			);
			return Ok(None);
			}
			let Some(data) = body.data else {
			return Ok(None);
			};
			if data.status.as_deref() == Some("final") {
			info!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			chunk_count = data.chunk_count.unwrap_or_default(),
			audio_bytes = data.audio_bytes.unwrap_or_default(),
			asr_duration_ms = data.asr_duration_ms.unwrap_or_default(),
			wall_ms = started_at.elapsed().as_millis() as u64,
			provider = %data.provider_alias.as_deref().unwrap_or("unknown"),
			text_len = data.text_len.unwrap_or_default(),
			"runtime helper asr_stream_final"
			);
			return Ok(data.asr_result_ref);
			}
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"asr_stream_fallback",
			"skipped",
			None,
			Some(true),
			json!({
			"fallbackReason": data.fallback_reason,
			"fallbackStage": data.fallback_stage,
			"status": data.status,
			}),
			);
			Ok(None)
			}

			fn build_asr_stream_ndjson(
			call_id: &str,
			trace_id: &str,
			turn: &FinishedSpeechTurn,
			bridge_config: &TurnBridgeConfig,
			) -> Result<String> {
			let chunks = asr_pcm_16k_chunks(turn)?;
			let mut seq = 1u64;
			let mut lines = Vec::with_capacity(chunks.len() + 3);
			lines.push(serde_json::to_string(&json!({
			"event": "asr_stream_started",
			"seq": seq,
			"callId": call_id,
			"traceId": trace_id,
			"turnId": turn.turn_id.as_str(),
			"tsMs": current_time_millis(),
			"payload": {
			"format": "pcm_s16le",
			"sampleRate": 16000,
			"channels": 1,
			"runtimeSessionNonce": bridge_config.runtime_session_nonce.as_deref().unwrap_or_default(),
			"providerHint": "volcengine",
			}
			}))?);
			for (index, samples) in chunks.iter().enumerate() {
			seq += 1;
			let bytes = pcm_i16_to_le_bytes(samples);
			let duration_ms = ((samples.len() as u64) * 1000 / 16_000).max(1);
			lines.push(serde_json::to_string(&json!({
			"event": "asr_audio_chunk",
			"seq": seq,
			"callId": call_id,
			"traceId": trace_id,
			"turnId": turn.turn_id.as_str(),
			"tsMs": current_time_millis(),
			"payload": {
			"chunkSeq": index + 1,
			"format": "pcm_s16le",
			"sampleRate": 16000,
			"channels": 1,
			"durationMs": duration_ms,
			"payloadBase64": general_purpose::STANDARD.encode(bytes),
			}
			}))?);
			}
			seq += 1;
			lines.push(serde_json::to_string(&json!({
			"event": "vad_speech_end",
			"seq": seq,
			"callId": call_id,
			"traceId": trace_id,
			"turnId": turn.turn_id.as_str(),
			"tsMs": current_time_millis(),
			"payload": {
			"endReason": turn.end_reason.as_str(),
			"speechDurationMs": turn.duration_ms,
			}
			}))?);
			seq += 1;
			lines.push(serde_json::to_string(&json!({
			"event": "asr_stream_finish",
			"seq": seq,
			"callId": call_id,
			"traceId": trace_id,
			"turnId": turn.turn_id.as_str(),
			"tsMs": current_time_millis(),
			"payload": {
			"finalChunkSeq": chunks.len(),
			"audioDurationMs": turn.duration_ms,
			}
			}))?);
			Ok(lines.join("\n") + "\n")
			}

			fn asr_pcm_16k_chunks(turn: &FinishedSpeechTurn) -> Result<Vec<Vec<i16>>> {
			if turn.samples.is_empty() {
			return Err(anyhow!("empty turn samples"));
			}
			let samples_16k: Vec<i16> = turn.samples.iter().step_by(3).copied().collect();
			if samples_16k.is_empty() {
			return Err(anyhow!("empty 16k asr samples"));
			}
			let samples_per_chunk = 320usize;
			Ok(samples_16k
			.chunks(samples_per_chunk)
			.map(\|chunk\| chunk.to_vec())
			.collect())
			}

			fn pcm_i16_to_le_bytes(samples: &[i16]) -> Vec<u8> {
			let mut bytes = Vec::with_capacity(samples.len() * 2);
			for sample in samples {
			bytes.extend_from_slice(&sample.to_le_bytes());
			}
			bytes
			}

			async fn request_turn_bridge_stream(
			@@ -948,6 +1378,7 @@
			turn: &FinishedSpeechTurn,
			path_ref: &str,
			byte_size: u64,
			asr_result_ref: Option<&str>,
			turn_pipeline_started_at: Instant,
			) -> Result<RuntimeTurnStreamOutcome> {
			let bridge_started_at = Instant::now();
			@@ -955,15 +1386,16 @@
			call_id: call_id.to_string(),
			trace_id: trace_id.to_string(),
			turn_id: turn.turn_id.clone(),
			audio_artifact: RuntimeTurnAudioArtifact {
			audio_artifact: Some(RuntimeTurnAudioArtifact {
			artifact_type: "local_file".to_string(),
			path_ref: path_ref.to_string(),
			format: "wav".to_string(),
			sample_rate: TARGET_SAMPLE_RATE_HZ,
			channels: u32::from(TARGET_NUM_CHANNELS),
			sample_rate: USER_AUDIO_SAMPLE_RATE_HZ,
			channels: u32::from(USER_AUDIO_NUM_CHANNELS),
			duration_ms: turn.duration_ms,
			byte_size,
			},
			}),
			asr_result_ref: asr_result_ref.map(str::to_string),
			};
			let response = http
			.post(bridge_config.bridge_url.as_deref().unwrap_or_default())
			@@ -1042,6 +1474,7 @@
			.await?;
			}
			}
			state.close_timing();
			if !state.completed {
			warn!(
			call_id = %call_id,
			@@ -1171,6 +1604,7 @@
			call_id,
			trace_id,
			turn,
			&event,
			audio_chunk,
			state,
			turn_pipeline_started_at,
			@@ -1182,6 +1616,9 @@
			}
			Some("device_output") => {
			if let Some(output) = event.device_output.as_ref() {
			if !should_publish_device_output(&mut state.published_device_output_ids, output) {
			return Ok(());
			}
			sink.publish_device_output(call_id, trace_id, &turn.turn_id, output)
			.await?;
			state.device_output_count = state.device_output_count.saturating_add(1);
			@@ -1189,6 +1626,7 @@
			}
			Some("turn_completed") => {
			state.completed = true;
			state.close_timing();
			info!(
			call_id = %call_id,
			trace_id = %trace_id,
			@@ -1199,6 +1637,7 @@
			);
			}
			Some("turn_failed") => {
			state.close_timing();
			let error = event.error.as_ref();
			let reason_code = error
			.and_then(\|value\| value.reason_code.as_deref())
			@@ -1233,6 +1672,7 @@
			}
			Some("turn_cancelled") => {
			state.completed = true;
			state.close_timing();
			info!(
			call_id = %call_id,
			trace_id = %trace_id,
			@@ -1265,6 +1705,19 @@
			}
			Some("activity") => {
			if let Some(activity) = event.activity.as_ref() {
			if activity.event_type.as_deref() == Some("tts_first_audio_chunk_ready") {
			if let Some(runtime_session_nonce) =
			bridge_config.runtime_session_nonce.as_deref()
			{
			let _ = state.arm_timing_anchor(
			call_id,
			trace_id,
			&turn.turn_id,
			runtime_session_nonce,
			&event,
			);
			}
			}
			info!(
			call_id = %call_id,
			trace_id = %trace_id,
			@@ -1323,6 +1776,7 @@
			call_id: &str,
			trace_id: &str,
			turn: &FinishedSpeechTurn,
			event: &RuntimeTurnStreamEvent,
			audio_chunk: &RuntimeTurnStreamAudioChunk,
			state: &mut RuntimeTurnStreamState,
			turn_pipeline_started_at: Instant,
			@@ -1342,21 +1796,112 @@
			.unwrap_or("pcm_s16le")
			.trim()
			.to_ascii_lowercase();
			if !matches!(format.as_str(), "pcm_s16le" \| "mp3" \| "mpeg" \| "wav") {
			return Err(anyhow!("unsupported reply_audio_chunk format {format}"));
			}
			match state.reply_chunk_markers.observe(audio_chunk.segment_seq) {
			ReplyChunkMarker::FirstReply => {
			let extension = json!({
			"segmentSeq": audio_chunk.segment_seq,
			"chunkSeq": audio_chunk.chunk_seq,
			"format": format.as_str(),
			"bytes": payload.len(),
			});
			if let Some(marker) =
			state.record_m6(call_id, trace_id, &turn.turn_id, event, audio_chunk)
			{
			emit_anchored_activity(
			call_id,
			trace_id,
			&turn.turn_id,
			"helper_first_reply_audio_chunk_received",
			&marker,
			extension,
			);
			} else {
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"helper_first_reply_audio_chunk_received",
			"ok",
			None,
			None,
			extension,
			);
			}
			}
			ReplyChunkMarker::SegmentFirst => emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"helper_segment_first_audio_chunk_received",
			"ok",
			None,
			None,
			json!({
			"segmentSeq": audio_chunk.segment_seq,
			"chunkSeq": audio_chunk.chunk_seq,
			"format": format.as_str(),
			"bytes": payload.len(),
			}),
			),
			ReplyChunkMarker::None => {}
			}
			let frames = if format == "pcm_s16le" {
			pcm_s16le_payload_to_frames(
			&payload,
			audio_chunk.sample_rate.unwrap_or(TARGET_SAMPLE_RATE_HZ),
			audio_chunk
			.channels
			.unwrap_or(u32::from(TARGET_NUM_CHANNELS)),
			)?
			let sample_rate = audio_chunk.sample_rate.unwrap_or(sink.sample_rate_hz);
			let channels = audio_chunk.channels.unwrap_or(u32::from(sink.num_channels));
			if state.pcm_stream_decoder.is_none() {
			state.pcm_stream_decoder = Some(audio::PcmS16leStreamDecoder::new(
			sample_rate,
			channels,
			sink.sample_rate_hz,
			sink.num_channels,
			)?);
			}
			state.pcm_stream_network_chunk_count =
			state.pcm_stream_network_chunk_count.saturating_add(1);
			let stream_result = state
			.pcm_stream_decoder
			.as_mut()
			.expect("pcm stream decoder initialized")
			.push_bytes(
			&payload,
			sample_rate,
			channels,
			audio_chunk.last.unwrap_or(false),
			)?;
			if stream_result.dropped_tail_bytes > 0 {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			chunk_seq = audio_chunk.chunk_seq.unwrap_or_default(),
			dropped_tail_bytes = stream_result.dropped_tail_bytes,
			"runtime helper stream_audio_pcm_unaligned_tail_dropped"
			);
			}
			if stream_result.frames.is_empty() {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			chunk_seq = audio_chunk.chunk_seq.unwrap_or_default(),
			buffered_source_bytes = stream_result.buffered_source_bytes,
			buffered_source_samples = stream_result.buffered_source_samples,
			network_chunk_count = state.pcm_stream_network_chunk_count,
			"runtime helper stream_audio_pcm_waiting_for_20ms_frame"
			);
			return Ok(0);
			}
			stream_result.frames
			} else if matches!(format.as_str(), "mp3" \| "mpeg" \| "wav") {
			state.encoded_audio_buffer.extend_from_slice(&payload);
			match audio::decode_audio_bytes_to_frames(
			&state.encoded_audio_buffer,
			"stream_chunk",
			TARGET_SAMPLE_RATE_HZ,
			TARGET_NUM_CHANNELS,
			sink.sample_rate_hz,
			sink.num_channels,
			bridge_config.audio_debug_dump_dir.as_deref(),
			call_id,
			&format!("stream-reply-{}", turn.turn_id),
			@@ -1394,7 +1939,7 @@
			Err(error) => return Err(error).context("failed to decode final stream audio chunk"),
			}
			} else {
			return Err(anyhow!("unsupported reply_audio_chunk format {format}"));
			unreachable!("supported encoded format checked above")
			};
			if frames.is_empty() {
			return Ok(0);
			@@ -1422,25 +1967,65 @@
			sink.write_pcm_frame(frame).await?;
			if !state.first_audio_frame_written {
			state.first_audio_frame_written = true;
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"bot_reply_first_audio_frame_written",
			"ok",
			None,
			None,
			json!({
			"replyPlaybackMode": state.reply_playback_mode.as_str(),
			"format": format.as_str(),
			"chunkSeq": audio_chunk.chunk_seq,
			"replyTotalAfterVadEndMs": turn_pipeline_started_at.elapsed().as_millis() as u64,
			}),
			);
			let extension = json!({
			"replyPlaybackMode": state.reply_playback_mode.as_str(),
			"format": format.as_str(),
			"chunkSeq": audio_chunk.chunk_seq,
			"replyTotalAfterVadEndMs": turn_pipeline_started_at.elapsed().as_millis() as u64,
			});
			if let Some(marker) = state.record_m7() {
			emit_anchored_activity(
			call_id,
			trace_id,
			&turn.turn_id,
			"bot_reply_first_audio_frame_written",
			&marker,
			extension,
			);
			} else {
			emit_activity(
			call_id,
			trace_id,
			Some(&turn.turn_id),
			"bot_reply_first_audio_frame_written",
			"ok",
			None,
			None,
			extension,
			);
			}
			}
			sleep_until(pacing_started_at + Duration::from_millis(((index + 1) as u64) * 20)).await;
			}
			if audio_chunk.last.unwrap_or(false) {
			let mut debug_source_path = None;
			let mut debug_pcm_wav_path = None;
			let mut debug_pcm_wav_size_bytes = None;
			if format == "pcm_s16le" {
			if let (Some(debug_dump_dir), Some(decoder)) = (
			bridge_config.audio_debug_dump_dir.as_deref(),
			state.pcm_stream_decoder.as_ref(),
			) {
			match decoder.write_debug_dump(
			debug_dump_dir,
			call_id,
			&format!("stream-reply-{}", turn.turn_id),
			) {
			Ok(debug_dump) => {
			debug_source_path = debug_dump.debug_source_path;
			debug_pcm_wav_path = debug_dump.debug_pcm_wav_path;
			debug_pcm_wav_size_bytes = debug_dump.debug_pcm_wav_size_bytes;
			}
			Err(error) => warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			error = %safe_error(&error.to_string()),
			"runtime helper stream_audio_pcm_debug_dump_failed"
			),
			}
			}
			}
			emit_activity(
			call_id,
			trace_id,
			@@ -1453,51 +2038,20 @@
			"replyPlaybackMode": state.reply_playback_mode.as_str(),
			"format": format.as_str(),
			"chunkSeq": audio_chunk.chunk_seq,
			"networkChunkCount": state.pcm_stream_network_chunk_count,
			"debugSourcePath": debug_source_path,
			"debugPcmWavPath": debug_pcm_wav_path,
			"debugPcmWavSizeBytes": debug_pcm_wav_size_bytes,
			"sourceSampleRate": audio_chunk.sample_rate,
			"sourceChannels": audio_chunk.channels,
			"targetAudioProfile": sink.profile.as_str(),
			"targetSampleRate": sink.sample_rate_hz,
			"targetChannels": sink.num_channels,
			"replyTotalAfterVadEndMs": turn_pipeline_started_at.elapsed().as_millis() as u64,
			}),
			);
			}
			Ok(frames.len())
			}

			fn pcm_s16le_payload_to_frames(
			payload: &[u8],
			sample_rate: u32,
			channels: u32,
			) -> Result<Vec<PcmFrame>> {
			if sample_rate != TARGET_SAMPLE_RATE_HZ \|\| channels != u32::from(TARGET_NUM_CHANNELS) {
			return Err(anyhow!(
			"unsupported pcm_s16le stream format: sample_rate={}, channels={}",
			sample_rate,
			channels
			));
			}
			if payload.len() % 2 != 0 {
			return Err(anyhow!("pcm_s16le payload has odd byte length"));
			}
			let samples: Vec<i16> = payload
			.chunks_exact(2)
			.map(\|chunk\| i16::from_le_bytes([chunk[0], chunk[1]]))
			.collect();
			if samples.is_empty() {
			return Ok(Vec::new());
			}
			let samples_per_channel = (sample_rate / 50).max(1);
			let frame_sample_count = samples_per_channel as usize * channels as usize;
			let mut frames = Vec::new();
			for chunk in samples.chunks(frame_sample_count) {
			let chunk_samples_per_channel = (chunk.len() / channels as usize) as u32;
			if chunk_samples_per_channel == 0 {
			continue;
			}
			frames.push(PcmFrame::new(
			chunk.to_vec(),
			sample_rate,
			channels,
			chunk_samples_per_channel,
			));
			}
			Ok(frames)
			}

			fn trim_ascii_whitespace(value: &[u8]) -> &[u8] {
			@@ -1520,6 +2074,7 @@
			turn: &FinishedSpeechTurn,
			path_ref: &str,
			byte_size: u64,
			asr_result_ref: Option<&str>,
			turn_pipeline_started_at: Instant,
			) -> Result<RuntimeTurnBridgeOutcome> {
			let bridge_started_at = Instant::now();
			@@ -1527,15 +2082,16 @@
			call_id: call_id.to_string(),
			trace_id: trace_id.to_string(),
			turn_id: turn.turn_id.clone(),
			audio_artifact: RuntimeTurnAudioArtifact {
			audio_artifact: Some(RuntimeTurnAudioArtifact {
			artifact_type: "local_file".to_string(),
			path_ref: path_ref.to_string(),
			format: "wav".to_string(),
			sample_rate: TARGET_SAMPLE_RATE_HZ,
			channels: u32::from(TARGET_NUM_CHANNELS),
			sample_rate: USER_AUDIO_SAMPLE_RATE_HZ,
			channels: u32::from(USER_AUDIO_NUM_CHANNELS),
			duration_ms: turn.duration_ms,
			byte_size,
			},
			}),
			asr_result_ref: asr_result_ref.map(str::to_string),
			};
			let response = http
			.post(bridge_config.bridge_url.as_deref().unwrap_or_default())
			@@ -1695,8 +2251,8 @@
			http,
			Some(&audio_path_string),
			None,
			TARGET_SAMPLE_RATE_HZ,
			TARGET_NUM_CHANNELS,
			sink.sample_rate_hz,
			sink.num_channels,
			bridge_config.audio_debug_dump_dir.as_deref(),
			call_id,
			&reply_debug_label,
			@@ -1826,7 +2382,10 @@
			#[serde(rename = "turnId")]
			turn_id: String,
			#[serde(rename = "audioArtifact")]
			audio_artifact: RuntimeTurnAudioArtifact,
			#[serde(skip_serializing_if = "Option::is_none")]
			audio_artifact: Option<RuntimeTurnAudioArtifact>,
			#[serde(rename = "asrResultRef", skip_serializing_if = "Option::is_none")]
			asr_result_ref: Option<String>,
			}

			#[derive(Serialize)]
			@@ -1854,6 +2413,20 @@
			retryable: Option<bool>,
			}

			#[derive(Deserialize)]
			#[serde(rename_all = "camelCase")]
			struct RuntimeAsrStreamResp {
			status: Option<String>,
			asr_result_ref: Option<String>,
			chunk_count: Option<u64>,
			audio_bytes: Option<u64>,
			asr_duration_ms: Option<u64>,
			provider_alias: Option<String>,
			text_len: Option<u64>,
			fallback_reason: Option<String>,
			fallback_stage: Option<String>,
			}

			#[derive(Default)]
			struct RuntimeTurnBridgeOutcome {
			reply_audio_artifact: Option<RuntimeTurnReplyAudioArtifact>,
			@@ -1874,7 +2447,12 @@
			completed: bool,
			audio_chunk_count: u64,
			device_output_count: u64,
			published_device_output_ids: HashSet<String>,
			encoded_audio_buffer: Vec<u8>,
			pcm_stream_decoder: Option<audio::PcmS16leStreamDecoder>,
			pcm_stream_network_chunk_count: u64,
			reply_chunk_markers: ReplyChunkMarkerState,
			timing: RuntimeTurnStreamTimingState,
			}

			impl Default for RuntimeTurnStreamState {
			@@ -1887,8 +2465,266 @@
			completed: false,
			audio_chunk_count: 0,
			device_output_count: 0,
			published_device_output_ids: HashSet::new(),
			encoded_audio_buffer: Vec::new(),
			pcm_stream_decoder: None,
			pcm_stream_network_chunk_count: 0,
			reply_chunk_markers: ReplyChunkMarkerState::default(),
			timing: RuntimeTurnStreamTimingState::default(),
			}
			}
			}

			impl RuntimeTurnStreamState {
			fn arm_timing_anchor(
			&mut self,
			call_id: &str,
			trace_id: &str,
			turn_id: &str,
			runtime_session_nonce: &str,
			event: &RuntimeTurnStreamEvent,
			) -> bool {
			if self.timing.phase != RuntimeTurnStreamTimingPhase::Empty
			\|\| event.call_id.as_deref() != Some(call_id)
			\|\| event.trace_id.as_deref() != Some(trace_id)
			\|\| event.turn_id.as_deref() != Some(turn_id)
			{
			return false;
			}
			let Some(activity) = event.activity.as_ref() else {
			return false;
			};
			if activity.event_type.as_deref() != Some("tts_first_audio_chunk_ready") {
			return false;
			}
			let Some(extension) = activity.extension.as_ref() else {
			return false;
			};
			let Some(anchor_id) = extension.stream_anchor_id.as_deref() else {
			return false;
			};
			let valid_anchor_id = (16..=64).contains(&anchor_id.len()) && anchor_id.is_ascii();
			let expected_nonce_hash = runtime_session_nonce_hash(runtime_session_nonce);
			if extension.stream_timing_version != Some(STREAM_TIMING_VERSION)
			\|\| !valid_anchor_id
			\|\| extension.runtime_session_nonce_hash.as_deref() != Some(expected_nonce_hash.as_str())
			\|\| extension.segment_seq != Some(STREAM_TIMING_FIRST_SEGMENT)
			\|\| extension.stream_timing_validation.as_deref() != Some("bound")
			{
			return false;
			}
			let Some(server_delta_ms) = extension.stream_anchor_server_delta_ms else {
			return false;
			};
			self.timing.anchor = Some(RuntimeTurnStreamTimingAnchor {
			call_id: call_id.to_string(),
			trace_id: trace_id.to_string(),
			turn_id: turn_id.to_string(),
			anchor_id: anchor_id.to_string(),
			server_delta_ms,
			runtime_session_nonce_hash: expected_nonce_hash,
			segment_seq: STREAM_TIMING_FIRST_SEGMENT,
			received_at: Instant::now(),
			});
			self.timing.phase = RuntimeTurnStreamTimingPhase::Armed;
			true
			}

			fn record_m6(
			&mut self,
			call_id: &str,
			trace_id: &str,
			turn_id: &str,
			event: &RuntimeTurnStreamEvent,
			audio_chunk: &RuntimeTurnStreamAudioChunk,
			) -> Option<RuntimeTurnStreamTimingMarker> {
			if self.timing.phase != RuntimeTurnStreamTimingPhase::Armed {
			return None;
			}
			let anchor = self.timing.anchor.as_ref()?;
			if anchor.call_id != call_id
			\|\| anchor.trace_id != trace_id
			\|\| anchor.turn_id != turn_id
			\|\| event.call_id.as_deref() != Some(call_id)
			\|\| event.trace_id.as_deref() != Some(trace_id)
			\|\| event.turn_id.as_deref() != Some(turn_id)
			\|\| audio_chunk.segment_seq != Some(anchor.segment_seq)
			\|\| audio_chunk.chunk_seq != Some(STREAM_TIMING_FIRST_CHUNK)
			\|\| audio_chunk.stream_timing_version != Some(STREAM_TIMING_VERSION)
			\|\| audio_chunk.stream_anchor_id.as_deref() != Some(anchor.anchor_id.as_str())
			{
			return None;
			}
			let Some(marker) = RuntimeTurnStreamTimingMarker::from_anchor(
			anchor,
			audio_chunk.chunk_seq.unwrap_or(STREAM_TIMING_FIRST_CHUNK),
			) else {
			self.close_timing();
			return None;
			};
			self.timing.chunk_seq = Some(marker.chunk_seq);
			self.timing.phase = RuntimeTurnStreamTimingPhase::M6Recorded;
			Some(marker)
			}

			fn record_m7(&mut self) -> Option<RuntimeTurnStreamTimingMarker> {
			if self.timing.phase != RuntimeTurnStreamTimingPhase::M6Recorded {
			return None;
			}
			let anchor = self.timing.anchor.as_ref()?;
			let Some(marker) = RuntimeTurnStreamTimingMarker::from_anchor(
			anchor,
			self.timing.chunk_seq.unwrap_or(STREAM_TIMING_FIRST_CHUNK),
			) else {
			self.close_timing();
			return None;
			};
			self.timing.phase = RuntimeTurnStreamTimingPhase::M7Recorded;
			Some(marker)
			}

			fn close_timing(&mut self) {
			self.timing.close();
			}
			}

			#[derive(Debug, Clone, Copy, PartialEq, Eq)]
			enum RuntimeTurnStreamTimingPhase {
			Empty,
			Armed,
			M6Recorded,
			M7Recorded,
			Closed,
			}

			struct RuntimeTurnStreamTimingState {
			phase: RuntimeTurnStreamTimingPhase,
			anchor: Option<RuntimeTurnStreamTimingAnchor>,
			chunk_seq: Option<u64>,
			}

			impl Default for RuntimeTurnStreamTimingState {
			fn default() -> Self {
			Self {
			phase: RuntimeTurnStreamTimingPhase::Empty,
			anchor: None,
			chunk_seq: None,
			}
			}
			}

			impl RuntimeTurnStreamTimingState {
			fn close(&mut self) {
			self.anchor = None;
			self.chunk_seq = None;
			self.phase = RuntimeTurnStreamTimingPhase::Closed;
			}
			}

			impl Drop for RuntimeTurnStreamTimingState {
			fn drop(&mut self) {
			self.anchor = None;
			self.chunk_seq = None;
			}
			}

			struct RuntimeTurnStreamTimingAnchor {
			call_id: String,
			trace_id: String,
			turn_id: String,
			anchor_id: String,
			server_delta_ms: u64,
			runtime_session_nonce_hash: String,
			segment_seq: u64,
			received_at: Instant,
			}

			struct RuntimeTurnStreamTimingMarker {
			version: u32,
			anchor_id: String,
			anchor_server_delta_ms: u64,
			anchor_elapsed_ms: u64,
			server_delta_ms: u64,
			runtime_session_nonce_hash: String,
			segment_seq: u64,
			chunk_seq: u64,
			}

			impl RuntimeTurnStreamTimingMarker {
			fn from_anchor(anchor: &RuntimeTurnStreamTimingAnchor, chunk_seq: u64) -> Option<Self> {
			let elapsed_ms = u64::try_from(anchor.received_at.elapsed().as_millis()).ok()?;
			if elapsed_ms > STREAM_TIMING_MAX_ELAPSED_MS {
			return None;
			}
			Some(Self {
			version: STREAM_TIMING_VERSION,
			anchor_id: anchor.anchor_id.clone(),
			anchor_server_delta_ms: anchor.server_delta_ms,
			anchor_elapsed_ms: elapsed_ms,
			server_delta_ms: anchor.server_delta_ms.checked_add(elapsed_ms)?,
			runtime_session_nonce_hash: anchor.runtime_session_nonce_hash.clone(),
			segment_seq: anchor.segment_seq,
			chunk_seq,
			})
			}

			fn extension_with(&self, extra: serde_json::Value) -> serde_json::Value {
			let mut extension = match extra {
			serde_json::Value::Object(value) => value,
			_ => serde_json::Map::new(),
			};
			extension.insert("streamTimingVersion".to_string(), json!(self.version));
			extension.insert("streamAnchorId".to_string(), json!(self.anchor_id));
			extension.insert(
			"streamAnchorServerDeltaMs".to_string(),
			json!(self.anchor_server_delta_ms),
			);
			extension.insert("anchorElapsedMs".to_string(), json!(self.anchor_elapsed_ms));
			extension.insert(
			"runtimeSessionNonceHash".to_string(),
			json!(self.runtime_session_nonce_hash),
			);
			extension.insert("segmentSeq".to_string(), json!(self.segment_seq));
			extension.insert("chunkSeq".to_string(), json!(self.chunk_seq));
			extension.insert("streamTimingValidation".to_string(), json!("bound"));
			serde_json::Value::Object(extension)
			}
			}

			fn runtime_session_nonce_hash(value: &str) -> String {
			let digest = Sha256::digest(value.as_bytes());
			digest[..6]
			.iter()
			.map(\|byte\| format!("{byte:02x}"))
			.collect()
			}

			#[derive(Debug, PartialEq, Eq)]
			enum ReplyChunkMarker {
			FirstReply,
			SegmentFirst,
			None,
			}

			#[derive(Default)]
			struct ReplyChunkMarkerState {
			first_reply_seen: bool,
			seen_segments: HashSet<u64>,
			}

			impl ReplyChunkMarkerState {
			fn observe(&mut self, segment_seq: Option<u64>) -> ReplyChunkMarker {
			let first_for_segment = segment_seq
			.map(\|value\| self.seen_segments.insert(value))
			.unwrap_or(false);
			if !self.first_reply_seen {
			self.first_reply_seen = true;
			return ReplyChunkMarker::FirstReply;
			}
			if first_for_segment {
			return ReplyChunkMarker::SegmentFirst;
			}
			ReplyChunkMarker::None
			}
			}

			@@ -1908,6 +2744,12 @@
			struct RuntimeTurnStreamEvent {
			#[serde(rename = "type", alias = "event")]
			event_type: Option<String>,
			#[serde(rename = "callId")]
			call_id: Option<String>,
			#[serde(rename = "traceId")]
			trace_id: Option<String>,
			#[serde(rename = "turnId")]
			turn_id: Option<String>,
			#[serde(rename = "seq")]
			seq: Option<u64>,
			#[serde(rename = "replyPlaybackMode")]
			@@ -1943,6 +2785,12 @@
			struct RuntimeTurnStreamAudioChunk {
			#[serde(rename = "chunkSeq", alias = "seq")]
			chunk_seq: Option<u64>,
			#[serde(rename = "segmentSeq")]
			segment_seq: Option<u64>,
			#[serde(rename = "streamTimingVersion")]
			stream_timing_version: Option<u32>,
			#[serde(rename = "streamAnchorId")]
			stream_anchor_id: Option<String>,
			format: Option<String>,
			#[serde(rename = "sampleRate")]
			sample_rate: Option<u32>,
			@@ -1959,6 +2807,23 @@
			stage: Option<String>,
			#[serde(rename = "reasonCode")]
			reason_code: Option<String>,
			extension: Option<RuntimeTurnStreamTimingExtension>,
			}

			#[derive(Clone, Deserialize)]
			struct RuntimeTurnStreamTimingExtension {
			#[serde(rename = "streamTimingVersion")]
			stream_timing_version: Option<u32>,
			#[serde(rename = "streamAnchorId")]
			stream_anchor_id: Option<String>,
			#[serde(rename = "streamAnchorServerDeltaMs")]
			stream_anchor_server_delta_ms: Option<u64>,
			#[serde(rename = "runtimeSessionNonceHash")]
			runtime_session_nonce_hash: Option<String>,
			#[serde(rename = "segmentSeq")]
			segment_seq: Option<u64>,
			#[serde(rename = "streamTimingValidation")]
			stream_timing_validation: Option<String>,
			}

			#[derive(Deserialize)]
			@@ -1991,6 +2856,30 @@
			#[serde(rename = "commandCode")]
			command_code: Option<String>,
			params: Option<serde_json::Value>,
			}

			fn should_publish_device_output(
			published_ids: &mut HashSet<String>,
			output: &RuntimeTurnDeviceOutput,
			) -> bool {
			let Some(command_id) = output
			.command_id
			.as_deref()
			.map(str::trim)
			.filter(\|value\| !value.is_empty())
			else {
			return false;
			};
			if output
			.command_code
			.as_deref()
			.map(str::trim)
			.filter(\|value\| !value.is_empty())
			.is_none()
			{
			return false;
			}
			published_ids.insert(command_id.to_string())
			}

			fn require_safe_segment(value: &str) -> Result<()> {
			@@ -2032,14 +2921,77 @@
			turn_bridge_config: TurnBridgeConfig,
			http: Client,
			sink: Arc<BotAudioOutputSink>,
			participant: RemoteParticipant,
			) -> JoinHandle<()> {
			tokio::spawn(async move {
			let mut stream = NativeAudioStream::new(
			track.rtc_track(),
			TARGET_SAMPLE_RATE_HZ as i32,
			i32::from(TARGET_NUM_CHANNELS),
			USER_AUDIO_SAMPLE_RATE_HZ as i32,
			i32::from(USER_AUDIO_NUM_CHANNELS),
			);
			let started_at = Instant::now();
			let (frame_tx, mut frame_rx) =
			mpsc::channel::<DrainedUserAudioFrame>(INBOUND_AUDIO_QUEUE_CAPACITY);
			let (drain_shutdown_tx, mut drain_shutdown_rx) = watch::channel(false);
			let drain_call_id = call_id.clone();
			let drain_trace_id = trace_id.clone();
			let drain_task = tokio::spawn(async move {
			let mut received_frame_count: u64 = 0;
			let mut dropped_frame_count: u64 = 0;
			loop {
			tokio::select! {
			changed = drain_shutdown_rx.changed() => {
			match changed {
			Ok(()) if *drain_shutdown_rx.borrow() => break,
			Ok(()) => {}
			Err(_) => break,
			}
			}
			maybe_frame = stream.next() => {
			let Some(frame) = maybe_frame else {
			break;
			};
			received_frame_count = received_frame_count.saturating_add(1);
			let drained = DrainedUserAudioFrame {
			frame_index: received_frame_count,
			captured_elapsed_ms: started_at.elapsed().as_millis() as u64,
			frame: AudioFrame {
			data: Cow::Owned(frame.data.as_ref().to_vec()),
			sample_rate: frame.sample_rate,
			num_channels: frame.num_channels,
			samples_per_channel: frame.samples_per_channel,
			},
			};
			match frame_tx.try_send(drained) {
			Ok(()) => {}
			Err(mpsc::error::TrySendError::Full(_)) => {
			dropped_frame_count = dropped_frame_count.saturating_add(1);
			if dropped_frame_count % INBOUND_AUDIO_DROP_LOG_INTERVAL == 1 {
			warn!(
			call_id = %drain_call_id,
			trace_id = %drain_trace_id,
			dropped_frame_count,
			received_frame_count,
			queue_capacity = INBOUND_AUDIO_QUEUE_CAPACITY,
			"runtime helper inbound_audio_queue_full_dropping_newest"
			);
			}
			}
			Err(mpsc::error::TrySendError::Closed(_)) => break,
			}
			}
			}
			}
			stream.close();
			info!(
			call_id = %drain_call_id,
			trace_id = %drain_trace_id,
			received_frame_count,
			dropped_frame_count,
			queue_capacity = INBOUND_AUDIO_QUEUE_CAPACITY,
			"runtime helper user_audio_drain_ended"
			);
			});
			let mut frame_count: u64 = 0;
			let mut sample_count: u64 = 0;
			let mut simple_vad = if simple_vad_enabled {
			@@ -2047,11 +2999,14 @@
			} else {
			None
			};
			let mut realtime_asr_upload: Option<RealtimeAsrUpload> = None;
			let mut last_fixture_sequence: Option<String> = None;

			while let Some(frame) = stream.next().await {
			frame_count += 1;
			while let Some(drained) = frame_rx.recv().await {
			let frame = drained.frame;
			frame_count = drained.frame_index;
			sample_count += u64::from(frame.samples_per_channel) * u64::from(frame.num_channels);
			let elapsed_ms = started_at.elapsed().as_millis() as u64;
			let elapsed_ms = drained.captured_elapsed_ms;
			if frame_count == 1 {
			info!(
			call_id = %call_id,
			@@ -2079,7 +3034,8 @@

			if let Some(vad) = simple_vad.as_mut() {
			if vad_enabled_gate.load(Ordering::Acquire) {
			if let Some(turn) = vad.observe_frame(
			let (was_in_speech, is_in_speech, turn) = observe_frame_and_start_session(
			vad,
			&call_id,
			&trace_id,
			&participant_alias,
			@@ -2087,7 +3043,40 @@
			frame_count,
			elapsed_ms,
			&frame,
			) {
			http.clone(),
			turn_bridge_config.realtime_asr_config(),
			\|\| participant.attributes(),
			&mut realtime_asr_upload,
			&mut last_fixture_sequence,
			turn_bridge_config.asr_realtime_enabled,
			);

			if was_in_speech {
			let push_failed = realtime_asr_upload
			.as_mut()
			.and_then(\|upload\| upload.push_48k_samples(frame.data.as_ref()).err());
			if let Some(error) = push_failed {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			error = %safe_error(&error.to_string()),
			"runtime helper asr_realtime_upload_failed_fallback"
			);
			if let Some(upload) = realtime_asr_upload.take() {
			tokio::spawn(async move {
			upload.cancel("upload_backpressure").await;
			});
			}
			}
			}

			if let Some(turn) = turn {
			let realtime_asr_result_ref = match realtime_asr_upload.take() {
			Some(upload) => {
			finish_realtime_asr_upload(upload, &call_id, &trace_id, &turn).await
			}
			None => None,
			};
			handle_finished_turn(
			&http,
			&turn_bridge_config,
			@@ -2095,10 +3084,22 @@
			&call_id,
			&trace_id,
			turn,
			realtime_asr_result_ref,
			)
			.await;
			} else if was_in_speech && !is_in_speech {
			if let Some(upload) = realtime_asr_upload.take() {
			tokio::spawn(async move {
			upload.cancel("speech_too_short").await;
			});
			}
			}
			} else {
			if let Some(upload) = realtime_asr_upload.take() {
			tokio::spawn(async move {
			upload.cancel("vad_disabled").await;
			});
			}
			vad.observe_disabled_frame(
			&call_id,
			&trace_id,
			@@ -2119,9 +3120,41 @@
			&track_sid_alias,
			started_at.elapsed().as_millis() as u64,
			) {
			handle_finished_turn(&http, &turn_bridge_config, &sink, &call_id, &trace_id, turn)
			.await;
			let realtime_asr_result_ref = match realtime_asr_upload.take() {
			Some(upload) => {
			finish_realtime_asr_upload(upload, &call_id, &trace_id, &turn).await
			}
			None => None,
			};
			handle_finished_turn(
			&http,
			&turn_bridge_config,
			&sink,
			&call_id,
			&trace_id,
			turn,
			realtime_asr_result_ref,
			)
			.await;
			}
			}
			if let Some(upload) = realtime_asr_upload.take() {
			upload.cancel("stream_end").await;
			}
			let _ = drain_shutdown_tx.send(true);
			let mut drain_task = drain_task;
			if timeout(AUDIO_DRAIN_STOP_GRACE, &mut drain_task)
			.await
			.is_err()
			{
			drain_task.abort();
			let _ = drain_task.await;
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			stop_grace_ms = AUDIO_DRAIN_STOP_GRACE.as_millis() as u64,
			"runtime helper user_audio_drain_stop_timeout"
			);
			}

			info!(
			@@ -2135,6 +3168,182 @@
			"runtime helper user_audio_stream_ended"
			);
			})
			}

			fn observe_frame_and_start_session<F>(
			vad: &mut SimpleVad,
			call_id: &str,
			trace_id: &str,
			participant_alias: &str,
			track_sid_alias: &str,
			frame_count: u64,
			elapsed_ms: u64,
			frame: &AudioFrame<'_>,
			http: Client,
			config: RealtimeAsrConfig,
			read_attributes: F,
			upload_slot: &mut Option<RealtimeAsrUpload>,
			last_fixture_sequence: &mut Option<String>,
			realtime_enabled: bool,
			) -> (bool, bool, Option<FinishedSpeechTurn>)
			where
			F: FnOnce() -> std::collections::HashMap<String, String>,
			{
			let was_in_speech = vad.in_speech;
			let turn = vad.observe_frame(
			call_id,
			trace_id,
			participant_alias,
			track_sid_alias,
			frame_count,
			elapsed_ms,
			frame,
			);
			let is_in_speech = vad.in_speech;
			if !was_in_speech && is_in_speech {
			start_realtime_session_for_new_speech(
			http,
			config,
			call_id,
			trace_id,
			vad,
			read_attributes,
			upload_slot,
			last_fixture_sequence,
			realtime_enabled,
			);
			}
			(was_in_speech, is_in_speech, turn)
			}

			fn start_realtime_session_for_new_speech(
			http: Client,
			config: RealtimeAsrConfig,
			call_id: &str,
			trace_id: &str,
			vad: &SimpleVad,
			read_attributes: impl FnOnce() -> std::collections::HashMap<String, String>,
			upload_slot: &mut Option<RealtimeAsrUpload>,
			last_fixture_sequence: &mut Option<String>,
			realtime_enabled: bool,
			) {
			let turn_id = format!("turn-{:04}", vad.turn_index);
			let metadata = match AudioIngressMetadata::from_participant(&read_attributes()) {
			Ok(metadata) => metadata,
			Err(reason) => {
			warn!(call_id = %call_id, trace_id = %trace_id, turn_id = %turn_id,
			reason, "runtime helper asr_realtime_metadata_rejected");
			return;
			}
			};
			if let Some(metadata) = metadata.as_ref() {
			if !fixture_sequence_is_new(
			last_fixture_sequence.as_deref(),
			&metadata.client_fixture_sequence,
			) {
			warn!(call_id = %call_id, trace_id = %trace_id, turn_id = %turn_id,
			"runtime helper asr_realtime_metadata_sequence_rejected");
			return;
			}
			}
			match RealtimeAsrUpload::start(
			http,
			config,
			call_id,
			trace_id,
			&turn_id,
			&vad.speech_samples,
			metadata.as_ref(),
			) {
			Ok(upload) => {
			if let Some(metadata) = metadata {
			*last_fixture_sequence = Some(metadata.client_fixture_sequence);
			}
			info!(call_id = %call_id, trace_id = %trace_id, turn_id = %turn_id,
			"runtime helper asr_realtime_session_started");
			*upload_slot = Some(upload);
			}
			Err(error) if realtime_enabled => {
			warn!(call_id = %call_id, trace_id = %trace_id, turn_id = %turn_id,
			error = %safe_error(&error.to_string()),
			"runtime helper asr_realtime_start_failed_fallback");
			}
			Err(_) => {}
			}
			}

			fn fixture_sequence_is_new(previous: Option<&str>, current: &str) -> bool {
			let Some(previous) = previous else {
			return true;
			};
			let current_number = current
			.rsplit_once('-')
			.and_then(\|(_, value)\| value.parse::<u64>().ok());
			let previous_number = previous
			.rsplit_once('-')
			.and_then(\|(_, value)\| value.parse::<u64>().ok());
			match (previous_number, current_number) {
			(Some(previous), Some(current)) => current > previous,
			_ => previous != current,
			}
			}

			struct DrainedUserAudioFrame {
			frame_index: u64,
			captured_elapsed_ms: u64,
			frame: AudioFrame<'static>,
			}

			async fn finish_realtime_asr_upload(
			upload: RealtimeAsrUpload,
			call_id: &str,
			trace_id: &str,
			turn: &FinishedSpeechTurn,
			) -> Option<String> {
			match upload.finish(turn.duration_ms, &turn.end_reason).await {
			Ok(RealtimeAsrOutcome {
			status,
			asr_result_ref,
			provider_alias,
			partial_count,
			fallback_reason,
			fallback_stage,
			chunk_count,
			audio_bytes,
			wall_ms,
			}) => {
			info!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			status = %status,
			provider_alias = ?provider_alias,
			partial_count,
			chunk_count,
			audio_bytes,
			wall_ms,
			asr_result_ref_present = asr_result_ref.is_some(),
			fallback_reason = ?fallback_reason,
			fallback_stage = ?fallback_stage,
			"runtime helper asr_realtime_finished"
			);
			if status == "final" {
			asr_result_ref
			} else {
			None
			}
			}
			Err(error) => {
			warn!(
			call_id = %call_id,
			trace_id = %trace_id,
			turn_id = %turn.turn_id,
			error = %safe_error(&error.to_string()),
			"runtime helper asr_realtime_failed_fallback"
			);
			None
			}
			}
			}

			#[derive(Clone)]
			@@ -2154,8 +3363,8 @@
			rms_threshold: f64_env("CV_VAD_RMS_THRESHOLD", 0.012),
			peak_threshold: f64_env("CV_VAD_PEAK_THRESHOLD", 0.08),
			start_frames: u32_env("CV_VAD_START_FRAMES", 5).max(1),
			end_silence_ms: u64_env("CV_VAD_END_SILENCE_MS", 700).max(100),
			min_speech_ms: u64_env("CV_VAD_MIN_SPEECH_MS", 300).max(1),
			end_silence_ms: u64_env("CV_VAD_END_SILENCE_MS", 400).max(100),
			min_speech_ms: u64_env("CV_VAD_MIN_SPEECH_MS", 250).max(1),
			max_turn_ms: u64_env("CV_VAD_MAX_TURN_MS", 10_000).max(1_000),
			initial_ignore_ms: u64_env("CV_VAD_INITIAL_IGNORE_MS", 500),
			}
			@@ -2268,7 +3477,9 @@
			return None;
			}

			let voiced = rms >= self.config.rms_threshold \|\| peak >= self.config.peak_threshold;
			// Align with cb-sdk's energy-based segmentation: peak is diagnostic only,
			// otherwise isolated spikes can keep a turn open until max_turn_ms.
			let voiced = rms >= self.config.rms_threshold;
			if !self.in_speech {
			self.remember_pre_speech_frame(frame);
			}
			@@ -2524,6 +3735,9 @@
			rtc_source: NativeAudioSource,
			track: LocalAudioTrack,
			device_output_destination_identity: Option<String>,
			profile: String,
			sample_rate_hz: u32,
			num_channels: u16,
			}

			impl BotAudioOutputSink {
			@@ -2534,6 +3748,7 @@
			call_id: &str,
			trace_id: &str,
			track_name: &str,
			profile: String,
			sample_rate: u32,
			num_channels: u32,
			device_output_destination_identity: Option<String>,
			@@ -2562,11 +3777,14 @@
			"failed to publish bot audio track in room {room_alias} for participant {participant_alias}: {error}"
			)
			})?;
			let num_channels_u16 = u16::try_from(num_channels)
			.map_err(\|_\| anyhow!("unsupported bot audio channel count {num_channels}"))?;

			info!(
			room_alias = %room_alias,
			participant_alias = %participant_alias,
			track_name = %track_name,
			bot_audio_profile = %profile,
			sample_rate,
			num_channels,
			"runtime helper published bot audio track"
			@@ -2581,6 +3799,7 @@
			None,
			json!({
			"trackName": track_name,
			"audioProfile": profile,
			"sampleRate": sample_rate,
			"numChannels": num_channels,
			}),
			@@ -2591,6 +3810,9 @@
			rtc_source,
			track,
			device_output_destination_identity,
			profile,
			sample_rate_hz: sample_rate,
			num_channels: num_channels_u16,
			})
			}

			@@ -2815,3 +4037,460 @@
			Ok(())
			}
			}

			#[cfg(test)]
			mod tests {
			use super::{
			ReplyChunkMarker, ReplyChunkMarkerState, RuntimeTurnDeviceOutput, RuntimeTurnStreamEvent,
			RuntimeTurnStreamState, RuntimeTurnStreamTimingPhase, runtime_session_nonce_hash,
			should_publish_device_output,
			};
			use std::{
			collections::HashSet,
			io::{Read, Write},
			net::TcpListener,
			sync::mpsc,
			thread,
			time::Duration,
			};

			#[test]
			fn production_vad_session_boundary_reads_updated_attributes() {
			let config = SimpleVadConfig {
			rms_threshold: 0.001,
			peak_threshold: 0.01,
			start_frames: 2,
			end_silence_ms: 100,
			min_speech_ms: 1,
			max_turn_ms: 1_000,
			initial_ignore_ms: 0,
			};
			let mut vad = SimpleVad::new(config);
			let samples = vec![1_000i16; 160];
			let frame = AudioFrame {
			data: samples.as_slice().into(),
			sample_rate: 16_000,
			num_channels: 1,
			samples_per_channel: 160,
			};
			let mut attributes = std::collections::HashMap::from([
			(
			"inputSourceCategory".to_string(),
			"controlled_fixture".to_string(),
			),
			(
			"clientFixtureSequence".to_string(),
			"fixture-01".to_string(),
			),
			]);
			let mut starts = Vec::new();
			for (session_index, sequence) in [(1, "fixture-01"), (2, "fixture-02")] {
			let was_in_speech = vad.in_speech;
			vad.observe_frame(
			"call-001",
			"trace-001",
			"participant",
			"track",
			session_index * 2 - 1,
			1_000 * session_index,
			&frame,
			);
			vad.observe_frame(
			"call-001",
			"trace-001",
			"participant",
			"track",
			session_index * 2,
			1_000 * session_index + 10,
			&frame,
			);
			let is_in_speech = vad.in_speech;
			assert!(!was_in_speech && is_in_speech);
			attributes.insert("clientFixtureSequence".to_string(), sequence.to_string());
			let metadata = AudioIngressMetadata::from_participant(&attributes)
			.expect("valid participant attributes")
			.expect("controlled fixture metadata");
			let session_line = asr_realtime::session_start_line(
			"call-001",
			"trace-001",
			&format!("turn-{session_index:04}"),
			"nonce-001",
			Some(&metadata),
			)
			.expect("session start line");
			let session_json: serde_json::Value =
			serde_json::from_slice(&session_line).expect("session start json");
			assert_eq!(sequence, session_json["clientFixtureSequence"]);
			starts.push(metadata.client_fixture_sequence);
			vad.reset_current_turn();
			}
			assert_eq!(vec!["fixture-01", "fixture-02"], starts);
			attributes.insert("inputSourceCategory".to_string(), "other".to_string());
			assert!(AudioIngressMetadata::from_participant(&attributes).is_err());
			assert!(
			AudioIngressMetadata::from_participant(&std::collections::HashMap::new())
			.expect("missing attributes is absent")
			.is_none()
			);
			attributes.insert(
			"clientFixtureSequence".to_string(),
			"fixture-01".to_string(),
			);
			assert!(AudioIngressMetadata::from_participant(&attributes).is_err());
			}

			#[test]
			fn production_observer_rejects_wrong_participant_before_vad_session() {
			assert!(!is_bound_user_participant(
			"participant-other",
			Some("participant-user")
			));
			assert!(is_bound_user_participant(
			"participant-user",
			Some("participant-user")
			));
			assert!(is_bound_user_participant("participant-any", None));
			}

			#[tokio::test]
			async fn production_observer_vad_to_session_entry_reads_each_updated_attribute() {
			let listener = TcpListener::bind("127.0.0.1:0").expect("bind local ASR fixture");
			let address = listener.local_addr().expect("fixture address");
			let (request_tx, request_rx) = mpsc::channel::<String>();
			let server = thread::spawn(move \|\| {
			for _ in 0..2 {
			let (mut stream, _) = listener.accept().expect("accept ASR session");
			stream
			.set_read_timeout(Some(Duration::from_secs(2)))
			.expect("set fixture timeout");
			let mut bytes = Vec::new();
			let mut buffer = [0_u8; 4096];
			loop {
			match stream.read(&mut buffer) {
			Ok(0) => break,
			Ok(size) => {
			bytes.extend_from_slice(&buffer[..size]);
			if bytes.windows(7).any(\|window\| window == b"0\r\n\r\n") {
			break;
			}
			}
			Err(_) => break,
			}
			}
			request_tx
			.send(String::from_utf8_lossy(&bytes).into_owned())
			.expect("capture ASR request");
			stream
			.write_all(b"HTTP/1.1 200 OK\r\ncontent-type: application/json\r\ncontent-length: 39\r\nconnection: close\r\n\r\n{\"code\":0,\"data\":{\"status\":\"ok\"}}")
			.expect("write fixture response");
			}
			});
			let mut vad = SimpleVad::new(SimpleVadConfig {
			rms_threshold: 0.001,
			peak_threshold: 0.01,
			start_frames: 1,
			end_silence_ms: 100,
			min_speech_ms: 1,
			max_turn_ms: 1_000,
			initial_ignore_ms: 0,
			});
			let frame_data = vec![1_000i16; 160];
			let frame = AudioFrame {
			data: frame_data.as_slice().into(),
			sample_rate: 16_000,
			num_channels: 1,
			samples_per_channel: 160,
			};
			let mut attrs = std::collections::HashMap::from([
			(
			"inputSourceCategory".to_string(),
			"controlled_fixture".to_string(),
			),
			(
			"clientFixtureSequence".to_string(),
			"fixture-01".to_string(),
			),
			]);
			let mut upload = None;
			let mut last_fixture_sequence = None;
			let config = RealtimeAsrConfig {
			enabled: true,
			url: Some(format!("http://{address}/runtime/asr/realtime")),
			runtime_token: Some("test".to_string()),
			runtime_session_nonce: Some("test".to_string()),
			chunk_duration_ms: 200,
			};
			let (was, is, turn) = observe_frame_and_start_session(
			&mut vad,
			"call-001",
			"trace-001",
			"participant-user",
			"track-001",
			1,
			1_000,
			&frame,
			Client::new(),
			config.clone(),
			\|\| attrs.clone(),
			&mut upload,
			&mut last_fixture_sequence,
			true,
			);
			assert!(!was && is && turn.is_none());
			assert!(upload.is_some());
			upload.take().unwrap().cancel("test").await;
			vad.reset_current_turn();
			attrs.insert(
			"clientFixtureSequence".to_string(),
			"fixture-02".to_string(),
			);
			let (was, is, turn) = observe_frame_and_start_session(
			&mut vad,
			"call-001",
			"trace-001",
			"participant-user",
			"track-001",
			2,
			2_000,
			&frame,
			Client::new(),
			config,
			\|\| attrs.clone(),
			&mut upload,
			&mut last_fixture_sequence,
			true,
			);
			assert!(!was && is && turn.is_none());
			assert!(upload.is_some());
			upload.take().unwrap().cancel("test").await;

			let first_request = request_rx
			.recv_timeout(Duration::from_secs(2))
			.expect("first session request");
			let second_request = request_rx
			.recv_timeout(Duration::from_secs(2))
			.expect("second session request");
			assert!(first_request.contains("\\\"clientFixtureSequence\\\":\\\"fixture-01\\\""));
			assert!(second_request.contains("\\\"clientFixtureSequence\\\":\\\"fixture-02\\\""));
			server.join().expect("fixture server");

			// The same production boundary rejects a wrong participant before VAD/session creation.
			assert!(!is_bound_user_participant(
			"participant-other",
			Some("participant-user")
			));
			assert_eq!(0, request_rx.try_iter().count());
			vad.reset_current_turn();
			attrs.insert(
			"clientFixtureSequence".to_string(),
			"fixture-01".to_string(),
			);
			let (_, _, _) = observe_frame_and_start_session(
			&mut vad,
			"call-001",
			"trace-001",
			"participant-user",
			"track-001",
			3,
			3_000,
			&frame,
			Client::new(),
			RealtimeAsrConfig {
			enabled: true,
			url: Some(format!("http://{address}/runtime/asr/realtime")),
			runtime_token: Some("test".to_string()),
			runtime_session_nonce: Some("test".to_string()),
			chunk_duration_ms: 200,
			},
			\|\| attrs.clone(),
			&mut upload,
			&mut last_fixture_sequence,
			true,
			);
			assert!(upload.is_none());
			vad.reset_current_turn();
			attrs.insert(
			"clientFixtureSequence".to_string(),
			"fixture-00".to_string(),
			);
			let (_, _, _) = observe_frame_and_start_session(
			&mut vad,
			"call-001",
			"trace-001",
			"participant-user",
			"track-001",
			4,
			4_000,
			&frame,
			Client::new(),
			RealtimeAsrConfig {
			enabled: true,
			url: Some(format!("http://{address}/runtime/asr/realtime")),
			runtime_token: Some("test".to_string()),
			runtime_session_nonce: Some("test".to_string()),
			chunk_duration_ms: 200,
			},
			\|\| attrs.clone(),
			&mut upload,
			&mut last_fixture_sequence,
			true,
			);
			assert!(upload.is_none());
			vad.reset_current_turn();
			attrs.insert("inputSourceCategory".to_string(), "other".to_string());
			let mut invalid_upload = None;
			let (_, _, invalid_turn) = observe_frame_and_start_session(
			&mut vad,
			"call-001",
			"trace-001",
			"participant-other",
			"track-001",
			3,
			3_000,
			&frame,
			Client::new(),
			RealtimeAsrConfig {
			enabled: true,
			url: Some("http://127.0.0.1:9".to_string()),
			runtime_token: Some("test".to_string()),
			runtime_session_nonce: Some("test".to_string()),
			chunk_duration_ms: 200,
			},
			\|\| attrs,
			&mut invalid_upload,
			&mut last_fixture_sequence,
			true,
			);
			assert!(invalid_turn.is_none());
			assert!(invalid_upload.is_none());
			}

			#[test]
			fn reply_chunk_marker_state_emits_turn_first_once_and_later_segment_first_once() {
			let mut state = ReplyChunkMarkerState::default();

			assert_eq!(ReplyChunkMarker::FirstReply, state.observe(Some(1)));
			assert_eq!(ReplyChunkMarker::None, state.observe(Some(1)));
			assert_eq!(ReplyChunkMarker::SegmentFirst, state.observe(Some(2)));
			assert_eq!(ReplyChunkMarker::None, state.observe(Some(2)));
			assert_eq!(ReplyChunkMarker::SegmentFirst, state.observe(Some(3)));
			}

			#[test]
			fn reply_chunk_marker_state_without_segment_only_emits_turn_first() {
			let mut state = ReplyChunkMarkerState::default();

			assert_eq!(ReplyChunkMarker::FirstReply, state.observe(None));
			assert_eq!(ReplyChunkMarker::None, state.observe(None));
			}

			#[test]
			fn runtime_turn_stream_audio_chunk_reads_segment_seq() {
			let event: RuntimeTurnStreamEvent = serde_json::from_str(
			r#"{"type":"reply_audio_chunk","audioChunk":{"chunkSeq":4,"segmentSeq":2,"format":"pcm_s16le","payloadBase64":"AA==","last":false}}"#,
			)
			.expect("turn stream event");

			assert_eq!(
			Some(2),
			event.audio_chunk.and_then(\|chunk\| chunk.segment_seq)
			);
			}

			#[test]
			fn runtime_turn_stream_reads_frozen_m5_and_audio_chunk_timing_contract() {
			let activity: RuntimeTurnStreamEvent = serde_json::from_str(
			r#"{"type":"activity","callId":"call-1","traceId":"trace-1","turnId":"turn-1","activity":{"eventType":"tts_first_audio_chunk_ready","extension":{"streamTimingVersion":1,"streamAnchorId":"0123456789abcdef","streamAnchorServerDeltaMs":1200,"runtimeSessionNonceHash":"abcdef012345","segmentSeq":1,"streamTimingValidation":"bound"}}}"#,
			)
			.expect("m5 activity event");
			let audio: RuntimeTurnStreamEvent = serde_json::from_str(
			r#"{"type":"reply_audio_chunk","callId":"call-1","traceId":"trace-1","turnId":"turn-1","audioChunk":{"chunkSeq":1,"segmentSeq":1,"streamTimingVersion":1,"streamAnchorId":"0123456789abcdef","format":"pcm_s16le","payloadBase64":"AA==","last":false}}"#,
			)
			.expect("timed audio chunk event");

			assert_eq!(Some("call-1"), activity.call_id.as_deref());
			assert_eq!(Some("trace-1"), activity.trace_id.as_deref());
			assert_eq!(Some("turn-1"), activity.turn_id.as_deref());
			let extension = activity.activity.unwrap().extension.unwrap();
			assert_eq!(Some(1), extension.stream_timing_version);
			assert_eq!(Some(1200), extension.stream_anchor_server_delta_ms);
			assert_eq!(
			Some("0123456789abcdef"),
			audio.audio_chunk.unwrap().stream_anchor_id.as_deref()
			);
			}

			#[test]
			fn runtime_turn_stream_timing_records_m6_and_m7_once_then_rejects_terminal_late_events() {
			let nonce = "runtime-nonce";
			let nonce_hash = runtime_session_nonce_hash(nonce);
			let activity: RuntimeTurnStreamEvent = serde_json::from_str(&format!(
			r#"{{"type":"activity","callId":"call-1","traceId":"trace-1","turnId":"turn-1","activity":{{"eventType":"tts_first_audio_chunk_ready","extension":{{"streamTimingVersion":1,"streamAnchorId":"0123456789abcdef","streamAnchorServerDeltaMs":1200,"runtimeSessionNonceHash":"{nonce_hash}","segmentSeq":1,"streamTimingValidation":"bound"}}}}}}"#,
			))
			.expect("m5 activity event");
			let audio: RuntimeTurnStreamEvent = serde_json::from_str(
			r#"{"type":"reply_audio_chunk","callId":"call-1","traceId":"trace-1","turnId":"turn-1","audioChunk":{"chunkSeq":1,"segmentSeq":1,"streamTimingVersion":1,"streamAnchorId":"0123456789abcdef","format":"pcm_s16le","payloadBase64":"AA==","last":false}}"#,
			)
			.expect("timed audio chunk event");
			let mut state = RuntimeTurnStreamState::default();

			assert!(state.arm_timing_anchor("call-1", "trace-1", "turn-1", nonce, &activity));
			let chunk = audio.audio_chunk.as_ref().expect("audio chunk");
			assert!(
			state
			.record_m6("call-1", "trace-1", "turn-1", &audio, chunk)
			.is_some()
			);
			assert!(
			state
			.record_m6("call-1", "trace-1", "turn-1", &audio, chunk)
			.is_none()
			);
			assert!(state.record_m7().is_some());
			assert!(state.record_m7().is_none());

			state.close_timing();
			assert_eq!(RuntimeTurnStreamTimingPhase::Closed, state.timing.phase);
			assert!(state.timing.anchor.is_none());
			assert!(!state.arm_timing_anchor("call-1", "trace-1", "turn-1", nonce, &activity));
			assert!(
			state
			.record_m6("call-1", "trace-1", "turn-1", &audio, chunk)
			.is_none()
			);
			assert!(state.record_m7().is_none());
			}

			#[test]
			fn device_output_contract_is_reliable_and_deduplicated() {
			let output = RuntimeTurnDeviceOutput {
			command_id: Some("cmd-1".to_string()),
			command_code: Some("custom.app.DeviceLevelChange".to_string()),
			params: Some(serde_json::json!({"level": 1})),
			};
			let mut published = HashSet::new();
			assert!(should_publish_device_output(&mut published, &output));
			assert!(!should_publish_device_output(&mut published, &output));
			assert_eq!(published.len(), 1);
			}

			#[test]
			fn device_output_invalid_or_missing_command_is_fail_closed() {
			for output in [
			RuntimeTurnDeviceOutput {
			command_id: None,
			command_code: Some("custom.app.DeviceLevelChange".to_string()),
			params: None,
			},
			RuntimeTurnDeviceOutput {
			command_id: Some("cmd-1".to_string()),
			command_code: None,
			params: None,
			},
			] {
			let mut published = HashSet::new();
			assert!(!should_publish_device_output(&mut published, &output));
			assert!(published.is_empty());
			}
			}
			}