@absolutejs/voice 0.0.22-beta.320 → 0.0.22-beta.322

package/README.md

@@ -4882,3 +4882,132 @@ Browser and framework helpers sit on top of the same connection core:
 - `VoiceStreamService` in `@absolutejs/voice/angular`
 
 For plain HTML or HTMX flows, use `@absolutejs/voice/client` directly.
+
+### Browser Media Proof
+
+If your app owns a browser `RTCPeerConnection`, pass it through `browserMedia` so AbsoluteJS can persist real `RTCPeerConnection.getStats()` evidence and feed production readiness. The default WebSocket microphone flow does not require this; this is for WebRTC voice surfaces where browser transport quality matters.
+
+Server route setup:
+
+```ts
+import {
+	createVoiceBrowserMediaRoutes,
+	createVoiceProductionReadinessRoutes,
+	getLatestVoiceBrowserMediaReport
+} from '@absolutejs/voice';
+
+app
+	.use(createVoiceBrowserMediaRoutes({ store: runtime.traces }))
+	.use(
+		createVoiceProductionReadinessRoutes({
+			browserMedia: () =>
+				getLatestVoiceBrowserMediaReport({ store: runtime.traces }),
+			links: {
+				browserMedia: '/voice/browser-media'
+			}
+		})
+	);
+```
+
+Shared stream options:
+
+```ts
+const browserMedia = {
+	continuity: {
+		maxGapMs: 7000,
+		maxInboundPacketStallMs: 7000,
+		maxOutboundPacketStallMs: 7000,
+		requireInboundAudio: true,
+		requireOutboundAudio: true
+	},
+	getPeerConnection: () => peerConnection,
+	maxJitterMs: 30,
+	maxPacketLossRatio: 0.02,
+	maxRoundTripTimeMs: 250,
+	requireConnectedCandidatePair: true,
+	requireLiveAudioTrack: true
+};
+```
+
+React:
+
+```tsx
+import { useRef } from 'react';
+import { useVoiceStream } from '@absolutejs/voice/react';
+
+export function WebRTCVoice() {
+	const peerConnection = useRef<RTCPeerConnection | null>(null);
+	const voice = useVoiceStream('/voice/support', {
+		browserMedia: {
+			...browserMedia,
+			getPeerConnection: () => peerConnection.current
+		}
+	});
+
+	return <button onClick={() => voice.close()}>End call</button>;
+}
+```
+
+Vue:
+
+```ts
+import { shallowRef } from 'vue';
+import { useVoiceStream } from '@absolutejs/voice/vue';
+
+const peerConnection = shallowRef<RTCPeerConnection | null>(null);
+const voice = useVoiceStream('/voice/support', {
+	browserMedia: {
+		...browserMedia,
+		getPeerConnection: () => peerConnection.value
+	}
+});
+```
+
+Svelte:
+
+```ts
+import { createVoiceStream } from '@absolutejs/voice/svelte';
+
+let peerConnection: RTCPeerConnection | null = null;
+const voice = createVoiceStream('/voice/support', {
+	browserMedia: {
+		...browserMedia,
+		getPeerConnection: () => peerConnection
+	}
+});
+```
+
+Angular:
+
+```ts
+import { Component, inject } from '@angular/core';
+import { VoiceStreamService } from '@absolutejs/voice/angular';
+
+@Component({
+	selector: 'app-webrtc-voice',
+	template: `<button type="button" (click)="stream.close()">End call</button>`
+})
+export class WebRTCVoiceComponent {
+	private readonly voice = inject(VoiceStreamService);
+	private peerConnection: RTCPeerConnection | null = null;
+
+	readonly stream = this.voice.connect('/voice/support', {
+		browserMedia: {
+			...browserMedia,
+			getPeerConnection: () => this.peerConnection
+		}
+	});
+}
+```
+
+HTMX/plain browser:
+
+```ts
+import { createVoiceController } from '@absolutejs/voice/client';
+
+const voice = createVoiceController('/voice/support', {
+	browserMedia
+});
+
+voice.bindHTMX({ element: '#voice-htmx-sync' });
+```

package/dist/angular/index.js

@@ -1412,6 +1412,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -1658,12 +1659,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -1749,21 +1802,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/browserMediaRoutes.d.ts

@@ -1,9 +1,10 @@
 import { Elysia } from 'elysia';
-import type { MediaWebRTCStatsReport } from '@absolutejs/media';
+import type { MediaWebRTCStatsReport, MediaWebRTCStreamContinuityReport } from '@absolutejs/media';
 import type { VoiceTraceEventStore } from './trace';
 export type VoiceBrowserMediaStatus = 'empty' | 'fail' | 'pass' | 'warn';
 export type VoiceBrowserMediaSample = {
     at: number;
+    continuity?: MediaWebRTCStreamContinuityReport;
     report: MediaWebRTCStatsReport;
     scenarioId?: string;
     sessionId: string;

package/dist/client/htmxBootstrap.js

@@ -260,6 +260,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -334,12 +335,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 
 // src/client/browserMedia.ts
@@ -362,21 +415,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/client/index.js

@@ -805,6 +805,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -1051,12 +1052,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -1142,21 +1195,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/index.js

@@ -11637,6 +11637,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -11883,12 +11884,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -12203,16 +12256,36 @@ var isMediaWebRTCStatsReport = (value) => {
   const report = value;
   return !!report && typeof report === "object" && (report.status === "pass" || report.status === "warn" || report.status === "fail") && typeof report.activeCandidatePairs === "number" && typeof report.liveAudioTracks === "number" && typeof report.packetLossRatio === "number" && typeof report.bytesReceived === "number" && typeof report.bytesSent === "number" && Array.isArray(report.issues);
 };
+var isMediaWebRTCStreamContinuityReport = (value) => {
+  const report = value;
+  return !!report && typeof report === "object" && (report.status === "pass" || report.status === "warn" || report.status === "fail") && typeof report.inboundAudioStreams === "number" && typeof report.outboundAudioStreams === "number" && typeof report.stalledInboundStreams === "number" && typeof report.stalledOutboundStreams === "number" && Array.isArray(report.issues) && Array.isArray(report.streams);
+};
 var isBrowserMediaPostBody = (value) => {
   const body = value;
   return !!body && isMediaWebRTCStatsReport(body.report);
 };
+var mergeIssues = (report, continuity) => {
+  if (!continuity) {
+    return report;
+  }
+  const issues = [
+    ...report.issues,
+    ...continuity.issues
+  ];
+  const status = report.status === "fail" || continuity.status === "fail" ? "fail" : report.status === "warn" || continuity.status === "warn" ? "warn" : "pass";
+  return {
+    ...report,
+    issues,
+    status
+  };
+};
 var toBrowserMediaSample = (event) => {
   if (event.type !== "client.browser_media" || !isMediaWebRTCStatsReport(event.payload.report)) {
     return;
   }
   return {
     at: event.at,
+    continuity: isMediaWebRTCStreamContinuityReport(event.payload.continuity) ? event.payload.continuity : undefined,
     report: event.payload.report,
     scenarioId: event.scenarioId,
     sessionId: event.sessionId,
@@ -12228,24 +12301,29 @@ var summarizeVoiceBrowserMedia = async (options) => {
   const latest = recent[0];
   const maxAgeMs = options.maxAgeMs ?? 30000;
   const stale = latest ? Date.now() - latest.at > maxAgeMs : false;
+  const latestReport = latest ? mergeIssues(latest.report, latest.continuity) : undefined;
   return {
     checkedAt: Date.now(),
     latest,
     recent,
     stale,
-    status: latest ? latest.report.status === "pass" && !stale ? "pass" : stale ? "warn" : latest.report.status : "empty",
+    status: latest ? latestReport?.status === "pass" && !stale ? "pass" : stale ? "warn" : latestReport?.status ?? latest.report.status : "empty",
     total: recent.length
   };
 };
 var getLatestVoiceBrowserMediaReport = async (options) => {
   const summary = await summarizeVoiceBrowserMedia(options);
-  return summary.latest?.report;
+  return summary.latest ? mergeIssues(summary.latest.report, summary.latest.continuity) : undefined;
 };
 var renderVoiceBrowserMediaHTML = (report, options = {}) => {
   const title = options.title ?? "Voice Browser Media";
   const latest = report.latest?.report;
-  const rows = report.recent.slice(0, 20).map((sample) => `<tr><td>${escapeHtml16(sample.sessionId)}</td><td>${escapeHtml16(sample.report.status)}</td><td>${String(sample.report.activeCandidatePairs)}</td><td>${String(sample.report.liveAudioTracks)}</td><td>${String(sample.report.roundTripTimeMs ?? "n/a")}ms</td><td>${String(sample.report.jitterMs ?? "n/a")}ms</td><td>${String(sample.report.packetLossRatio)}</td><td>${escapeHtml16(new Date(sample.at).toLocaleString())}</td></tr>`).join("");
-  return `<!doctype html><html lang="en"><head><meta charset="utf-8" /><meta name="viewport" content="width=device-width, initial-scale=1" /><title>${escapeHtml16(title)}</title><style>body{background:#0f172a;color:#e2e8f0;font-family:ui-sans-serif,system-ui,sans-serif;margin:0}main{margin:auto;max-width:1120px;padding:32px}.hero,.primitive,table{background:#111827;border:1px solid #334155;border-radius:22px;margin-bottom:16px}.hero,.primitive{padding:22px}.eyebrow{color:#93c5fd;font-weight:900;letter-spacing:.12em;text-transform:uppercase}h1{font-size:clamp(2.3rem,6vw,4.8rem);line-height:.92;margin:.2rem 0 1rem}.status{border:1px solid #64748b;border-radius:999px;display:inline-flex;font-weight:900;padding:8px 12px}.pass{color:#86efac}.warn,.empty{color:#fde68a}.fail{color:#fecaca}.metrics{display:grid;gap:12px;grid-template-columns:repeat(auto-fit,minmax(160px,1fr));margin-top:18px}.metric{background:#0b1220;border:1px solid #263244;border-radius:16px;padding:14px}.metric span{color:#94a3b8}.metric strong{display:block;font-size:1.7rem;margin-top:4px}.primitive code{color:#bfdbfe}table{border-collapse:collapse;overflow:hidden;width:100%}td,th{border-bottom:1px solid #334155;padding:10px;text-align:left}</style></head><body><main><section class="hero"><p class="eyebrow">Browser WebRTC media proof</p><h1>${escapeHtml16(title)}</h1><p class="status ${escapeHtml16(report.status)}">Status: ${escapeHtml16(report.status)}</p><p>Recent <code>client.browser_media</code> traces from browser <code>RTCPeerConnection.getStats()</code> reports.</p><section class="metrics"><div class="metric"><span>Reports</span><strong>${String(report.total)}</strong></div><div class="metric"><span>Candidate pairs</span><strong>${String(latest?.activeCandidatePairs ?? 0)}</strong></div><div class="metric"><span>Live audio tracks</span><strong>${String(latest?.liveAudioTracks ?? 0)}</strong></div><div class="metric"><span>RTT</span><strong>${String(latest?.roundTripTimeMs ?? "n/a")}ms</strong></div><div class="metric"><span>Jitter</span><strong>${String(latest?.jitterMs ?? "n/a")}ms</strong></div><div class="metric"><span>Loss</span><strong>${String(latest?.packetLossRatio ?? "n/a")}</strong></div></section></section><section class="primitive"><p class="eyebrow">Copy into your app</p><p><code>createVoiceBrowserMediaReporter({ peerConnection })</code> runs in the browser and posts reports here. <code>getLatestVoiceBrowserMediaReport(...)</code> can feed production readiness.</p></section><table><thead><tr><th>Session</th><th>Status</th><th>Pairs</th><th>Tracks</th><th>RTT</th><th>Jitter</th><th>Loss</th><th>Measured</th></tr></thead><tbody>${rows || '<tr><td colspan="8">No browser media reports yet.</td></tr>'}</tbody></table></main></body></html>`;
+  const latestContinuity = report.latest?.continuity;
+  const rows = report.recent.slice(0, 20).map((sample) => {
+    const stalledStreams = (sample.continuity?.stalledInboundStreams ?? 0) + (sample.continuity?.stalledOutboundStreams ?? 0);
+    return `<tr><td>${escapeHtml16(sample.sessionId)}</td><td>${escapeHtml16(mergeIssues(sample.report, sample.continuity).status)}</td><td>${String(sample.report.activeCandidatePairs)}</td><td>${String(sample.report.liveAudioTracks)}</td><td>${String(sample.continuity?.inboundAudioStreams ?? "n/a")}</td><td>${String(sample.continuity?.outboundAudioStreams ?? "n/a")}</td><td>${String(stalledStreams)}</td><td>${String(sample.report.roundTripTimeMs ?? "n/a")}ms</td><td>${String(sample.report.jitterMs ?? "n/a")}ms</td><td>${String(sample.report.packetLossRatio)}</td><td>${escapeHtml16(new Date(sample.at).toLocaleString())}</td></tr>`;
+  }).join("");
+  return `<!doctype html><html lang="en"><head><meta charset="utf-8" /><meta name="viewport" content="width=device-width, initial-scale=1" /><title>${escapeHtml16(title)}</title><style>body{background:#0f172a;color:#e2e8f0;font-family:ui-sans-serif,system-ui,sans-serif;margin:0}main{margin:auto;max-width:1120px;padding:32px}.hero,.primitive,table{background:#111827;border:1px solid #334155;border-radius:22px;margin-bottom:16px}.hero,.primitive{padding:22px}.eyebrow{color:#93c5fd;font-weight:900;letter-spacing:.12em;text-transform:uppercase}h1{font-size:clamp(2.3rem,6vw,4.8rem);line-height:.92;margin:.2rem 0 1rem}.status{border:1px solid #64748b;border-radius:999px;display:inline-flex;font-weight:900;padding:8px 12px}.pass{color:#86efac}.warn,.empty{color:#fde68a}.fail{color:#fecaca}.metrics{display:grid;gap:12px;grid-template-columns:repeat(auto-fit,minmax(160px,1fr));margin-top:18px}.metric{background:#0b1220;border:1px solid #263244;border-radius:16px;padding:14px}.metric span{color:#94a3b8}.metric strong{display:block;font-size:1.7rem;margin-top:4px}.primitive code{color:#bfdbfe}table{border-collapse:collapse;overflow:hidden;width:100%}td,th{border-bottom:1px solid #334155;padding:10px;text-align:left}</style></head><body><main><section class="hero"><p class="eyebrow">Browser WebRTC media proof</p><h1>${escapeHtml16(title)}</h1><p class="status ${escapeHtml16(report.status)}">Status: ${escapeHtml16(report.status)}</p><p>Recent <code>client.browser_media</code> traces from browser <code>RTCPeerConnection.getStats()</code> reports, including aggregate transport stats and per-stream continuity.</p><section class="metrics"><div class="metric"><span>Reports</span><strong>${String(report.total)}</strong></div><div class="metric"><span>Candidate pairs</span><strong>${String(latest?.activeCandidatePairs ?? 0)}</strong></div><div class="metric"><span>Live audio tracks</span><strong>${String(latest?.liveAudioTracks ?? 0)}</strong></div><div class="metric"><span>Inbound streams</span><strong>${String(latestContinuity?.inboundAudioStreams ?? "n/a")}</strong></div><div class="metric"><span>Outbound streams</span><strong>${String(latestContinuity?.outboundAudioStreams ?? "n/a")}</strong></div><div class="metric"><span>Stalled streams</span><strong>${String((latestContinuity?.stalledInboundStreams ?? 0) + (latestContinuity?.stalledOutboundStreams ?? 0))}</strong></div><div class="metric"><span>RTT</span><strong>${String(latest?.roundTripTimeMs ?? "n/a")}ms</strong></div><div class="metric"><span>Jitter</span><strong>${String(latest?.jitterMs ?? "n/a")}ms</strong></div><div class="metric"><span>Loss</span><strong>${String(latest?.packetLossRatio ?? "n/a")}</strong></div></section></section><section class="primitive"><p class="eyebrow">Copy into your app</p><p><code>createVoiceBrowserMediaReporter({ peerConnection, continuity })</code> runs in the browser and posts reports here. <code>getLatestVoiceBrowserMediaReport(...)</code> can feed production readiness with aggregate and continuity issues merged.</p></section><table><thead><tr><th>Session</th><th>Status</th><th>Pairs</th><th>Tracks</th><th>Inbound</th><th>Outbound</th><th>Stalled</th><th>RTT</th><th>Jitter</th><th>Loss</th><th>Measured</th></tr></thead><tbody>${rows || '<tr><td colspan="11">No browser media reports yet.</td></tr>'}</tbody></table></main></body></html>`;
 };
 var createVoiceBrowserMediaRoutes = (options) => {
   const path = options.path ?? "/api/voice/browser-media";
@@ -12261,6 +12339,7 @@ var createVoiceBrowserMediaRoutes = (options) => {
     await options.store.append({
       at: typeof body.at === "number" ? body.at : Date.now(),
       payload: {
+        continuity: isMediaWebRTCStreamContinuityReport(body.continuity) ? body.continuity : undefined,
         report: body.report
       },
       scenarioId: typeof body.scenarioId === "string" ? body.scenarioId : undefined,

package/dist/react/index.js

@@ -5048,6 +5048,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -5294,12 +5295,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -5385,21 +5438,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/svelte/index.js

@@ -3382,6 +3382,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -3628,12 +3629,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -3719,21 +3772,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/testing/index.js

@@ -2184,6 +2184,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -2430,12 +2431,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -2521,21 +2574,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/dist/types.d.ts

@@ -1,5 +1,5 @@
 import type { SessionStore } from '@absolutejs/absolute';
-import type { MediaWebRTCStatsCollector, MediaWebRTCStatsReport, MediaWebRTCStatsReportInput } from '@absolutejs/media';
+import type { MediaWebRTCStatsCollector, MediaWebRTCStatsReport, MediaWebRTCStatsReportInput, MediaWebRTCStreamContinuityInput, MediaWebRTCStreamContinuityReport } from '@absolutejs/media';
 import type { VoiceOpsDispositionTaskPolicies, VoiceOpsTaskAssignmentRule, VoiceOpsTaskAssignmentRules, VoiceIntegrationWebhookConfig, StoredVoiceIntegrationEvent, StoredVoiceOpsTask, VoiceIntegrationEventStore, VoiceOpsTaskPolicy, VoiceOpsTask, VoiceOpsTaskStore } from './ops';
 import type { VoiceIntegrationSink } from './opsSinks';
 import type { StoredVoiceCallReviewArtifact, VoiceCallReviewArtifact, VoiceCallReviewStore } from './testing/review';
@@ -774,6 +774,7 @@ export type VoiceConnectionOptions = {
 };
 export type VoiceBrowserMediaReportPayload = {
     at: number;
+    continuity?: MediaWebRTCStreamContinuityReport;
     report: MediaWebRTCStatsReport;
     scenarioId?: string | null;
     sessionId?: string | null;
@@ -784,6 +785,7 @@ export type VoiceBrowserMediaReporterOptions = Omit<MediaWebRTCStatsReportInput,
     getScenarioId?: () => string | null | undefined;
     getSessionId?: () => string | null | undefined;
     intervalMs?: number;
+    continuity?: false | Omit<MediaWebRTCStreamContinuityInput, 'previousStats' | 'stats'>;
     onError?: (error: unknown) => void;
     onReport?: (payload: VoiceBrowserMediaReportPayload) => void;
     path?: string;

package/dist/vue/index.js

@@ -4766,6 +4766,7 @@ var stringStat = (stat, key) => {
   const value = stat[key];
   return typeof value === "string" ? value : undefined;
 };
+var statKey = (stat) => String(stat.id ?? stringStat(stat, "ssrc") ?? numericStat(stat, "ssrc") ?? stringStat(stat, "trackIdentifier") ?? stringStat(stat, "mid") ?? "unknown");
 var secondsToMs = (value) => value === undefined ? undefined : value * 1000;
 var normalizeWebRTCStat = (stat) => {
   const sample = {};
@@ -5012,12 +5013,64 @@ var collectMediaWebRTCStats = async (input) => {
   const report = await input.peerConnection.getStats(input.selector ?? null);
   return [...report.values()].map(normalizeWebRTCStat);
 };
-var collectMediaWebRTCStatsReport = async (input) => {
-  const stats = await collectMediaWebRTCStats(input);
-  return buildMediaWebRTCStatsReport({
-    ...input,
-    stats
+var buildMediaWebRTCStreamContinuityReport = (input = {}) => {
+  const stats = input.stats ?? [];
+  const previousStats = input.previousStats ?? [];
+  const issues = [];
+  const previousByKey = new Map(previousStats.map((stat) => [statKey(stat), stat]));
+  const audioRtp = stats.filter((stat) => (stat.type === "inbound-rtp" || stat.type === "outbound-rtp") && stringStat(stat, "kind") !== "video" && stringStat(stat, "mediaType") !== "video");
+  const streams = audioRtp.map((stat) => {
+    const direction = stat.type === "outbound-rtp" ? "outbound" : "inbound";
+    const packetsKey = direction === "outbound" ? "packetsSent" : "packetsReceived";
+    const bytesKey = direction === "outbound" ? "bytesSent" : "bytesReceived";
+    const previous = previousByKey.get(statKey(stat));
+    const currentPackets = numericStat(stat, packetsKey);
+    const previousPackets = previous ? numericStat(previous, packetsKey) : undefined;
+    const currentBytes = numericStat(stat, bytesKey);
+    const previousBytes = previous ? numericStat(previous, bytesKey) : undefined;
+    const timeDeltaMs = stat.timestamp !== undefined && previous?.timestamp !== undefined ? stat.timestamp - previous.timestamp : undefined;
+    return {
+      bytesDelta: currentBytes !== undefined && previousBytes !== undefined ? currentBytes - previousBytes : undefined,
+      currentPackets,
+      direction,
+      id: statKey(stat),
+      packetDelta: currentPackets !== undefined && previousPackets !== undefined ? currentPackets - previousPackets : undefined,
+      previousPackets,
+      timeDeltaMs
+    };
   });
+  const inbound = streams.filter((stream) => stream.direction === "inbound");
+  const outbound = streams.filter((stream) => stream.direction === "outbound");
+  const maxObservedGapMs = max(streams.map((stream) => stream.timeDeltaMs).filter((value) => value !== undefined));
+  const stalledInboundStreams = inbound.filter((stream) => input.maxInboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxInboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  const stalledOutboundStreams = outbound.filter((stream) => input.maxOutboundPacketStallMs !== undefined && stream.timeDeltaMs !== undefined && stream.timeDeltaMs >= input.maxOutboundPacketStallMs && stream.packetDelta !== undefined && stream.packetDelta <= 0).length;
+  if (input.requireInboundAudio && inbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_audio_missing", "No inbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.requireOutboundAudio && outbound.length === 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_audio_missing", "No outbound WebRTC audio RTP stream was observed.");
+  }
+  if (input.maxGapMs !== undefined && maxObservedGapMs !== undefined && maxObservedGapMs > input.maxGapMs) {
+    pushIssue(issues, "warning", "media.webrtc_stream_gap", `Observed WebRTC stream sample gap ${String(maxObservedGapMs)}ms above ${String(input.maxGapMs)}ms.`);
+  }
+  if (stalledInboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_inbound_stalled", `${String(stalledInboundStreams)} inbound WebRTC audio stream(s) stopped receiving packets.`);
+  }
+  if (stalledOutboundStreams > 0) {
+    pushIssue(issues, "error", "media.webrtc_outbound_stalled", `${String(stalledOutboundStreams)} outbound WebRTC audio stream(s) stopped sending packets.`);
+  }
+  return {
+    checkedAt: Date.now(),
+    inboundAudioStreams: inbound.length,
+    issues,
+    maxObservedGapMs,
+    outboundAudioStreams: outbound.length,
+    stalledInboundStreams,
+    stalledOutboundStreams,
+    status: issues.some((issue) => issue.severity === "error") ? "fail" : issues.length > 0 ? "warn" : "pass",
+    streams,
+    totalStats: stats.length
+  };
 };
 var buildMediaPipelineCalibrationReport = (input = {}) => {
   const frames = input.frames ?? [];
@@ -5103,21 +5156,30 @@ var postBrowserMediaReport = async (payload, options) => {
 };
 var createVoiceBrowserMediaReporter = (options) => {
   let interval = null;
+  let previousStats = [];
   const reportOnce = async () => {
     const peerConnection = await resolvePeerConnection(options);
     if (!peerConnection) {
       return;
     }
-    const report = await collectMediaWebRTCStatsReport({
+    const stats = await collectMediaWebRTCStats({ peerConnection });
+    const report = buildMediaWebRTCStatsReport({
       ...options,
-      peerConnection
+      stats
+    });
+    const continuity = options.continuity === false ? undefined : buildMediaWebRTCStreamContinuityReport({
+      ...options.continuity,
+      previousStats,
+      stats
     });
     const payload = {
       at: Date.now(),
+      continuity,
       report,
       scenarioId: options.getScenarioId?.() ?? null,
       sessionId: options.getSessionId?.() ?? null
     };
+    previousStats = stats;
     options.onReport?.(payload);
     await postBrowserMediaReport(payload, options);
     return payload;

package/package.json

@@ -1,6 +1,6 @@
 {
 	"name": "@absolutejs/voice",
-	"version": "0.0.22-beta.320",
+	"version": "0.0.22-beta.322",
 	"description": "Voice primitives and Elysia plugin for AbsoluteJS",
 	"repository": {
 		"type": "git",
@@ -246,7 +246,7 @@
 		}
 	},
 	"dependencies": {
-		"@absolutejs/media": "0.0.1-beta.4"
+		"@absolutejs/media": "0.0.1-beta.5"
 	},
 	"devDependencies": {
 		"@absolutejs/absolute": "0.19.0-beta.646",