@camstack/addon-pipeline 1.2.344 → 1.2.346

package/dist/stream-broker/index.js

@@ -3,12 +3,12 @@ Object.defineProperties(exports, {
 	[Symbol.toStringTag]: { value: "Module" }
 });
 const require_chunk = require("../chunk-emK7D4bc.js");
-const require_dist = require("../dist-DNsARIUD.js");
-const require_node = require("../node-D-U3T-h-.js");
+const require_dist = require("../dist-qBCVRsyf.js");
+const require_node = require("../node-DmCpPH58.js");
 const require_restream_intent = require("../restream-intent-7uDNqMKs.js");
 const require_remote_restream = require("../remote-restream-CO36Sr30.js");
 const require_retire_root_keys = require("../retire-root-keys-Csb8z2f3.js");
-const require_segment_demux_js = require("../segment-demux-js-vZ6iniOU.js");
+const require_segment_demux_js = require("../segment-demux-js-DCLzkQTf.js");
 let node_os = require("node:os");
 node_os = require_chunk.__toESM(node_os);
 let node_crypto = require("node:crypto");
@@ -21337,6 +21337,169 @@ function shouldWriteLossLine(input) {
 	};
 }
 //#endregion
+//#region src/stream-broker/webrtc/talkback-bridge.ts
+/** A frame this small is Opus DTX / comfort noise, not speech. */
+var DTX_MAX_BYTES = 3;
+/** How the push-talk triple names each WebRTC audio codec, or null when the
+*  intercom path cannot take it. */
+function talkCodecOf(format) {
+	switch (format) {
+		case "Opus": return {
+			codec: "opus",
+			sampleRate: 48e3
+		};
+		case "Pcmu": return {
+			codec: "g711ulaw",
+			sampleRate: 8e3
+		};
+		case "Pcma": return {
+			codec: "g711alaw",
+			sampleRate: 8e3
+		};
+		default: return null;
+	}
+}
+var TalkbackBridge = class {
+	deviceId;
+	port;
+	logger;
+	state = "idle";
+	/** Set by `close()`; no frame is accepted after it. */
+	closing = false;
+	/** A talk session was opened at the camera and must be ended. */
+	started = false;
+	sequence = 0;
+	chain = Promise.resolve();
+	dropped = 0;
+	refusedCodecLogged = false;
+	constructor(deviceId, port, logger) {
+		this.deviceId = deviceId;
+		this.port = port;
+		this.logger = logger;
+	}
+	/** One inbound RTP payload from the peer. Never throws. */
+	onAudio(payload, format) {
+		if (this.closing) return;
+		if (payload.length <= DTX_MAX_BYTES) return;
+		const talk = talkCodecOf(format);
+		if (talk === null) {
+			if (!this.refusedCodecLogged) {
+				this.refusedCodecLogged = true;
+				this.logger.warn("talk-back: the peer sends a codec the intercom path cannot take", {
+					tags: { deviceId: this.deviceId },
+					meta: { format }
+				});
+			}
+			return;
+		}
+		if (this.state === "idle") {
+			this.state = "opening";
+			this.chain = this.chain.then(() => this.open());
+		}
+		const frame = {
+			deviceId: this.deviceId,
+			audioBase64: payload.toString("base64"),
+			codec: talk.codec,
+			sampleRate: talk.sampleRate,
+			channels: 1,
+			sequenceNumber: ++this.sequence
+		};
+		this.chain = this.chain.then(() => this.push(frame));
+	}
+	/** End the talk session (if one opened). Idempotent. */
+	async close() {
+		if (this.closing) return;
+		this.closing = true;
+		await this.chain.catch(() => {});
+		if (!this.started) return;
+		await this.port.endTalkSession(this.deviceId).catch((err) => {
+			this.logger.warn("talk-back: endTalkSession failed", {
+				tags: { deviceId: this.deviceId },
+				meta: { error: require_dist.errMsg(err) }
+			});
+		});
+		this.logger.info("talk-back: ended", {
+			tags: { deviceId: this.deviceId },
+			meta: {
+				frames: this.sequence,
+				dropped: this.dropped
+			}
+		});
+	}
+	async open() {
+		if (this.state !== "opening") return;
+		try {
+			await this.port.startTalkSession(this.deviceId);
+			this.started = true;
+			if (this.state === "opening") this.state = "open";
+			this.logger.info("talk-back: the peer is talking — camera talk session opened", { tags: { deviceId: this.deviceId } });
+		} catch (err) {
+			if (this.state === "opening") this.state = "idle";
+			this.logger.warn("talk-back: startTalkSession failed — will retry on the next frame", {
+				tags: { deviceId: this.deviceId },
+				meta: { error: require_dist.errMsg(err) }
+			});
+		}
+	}
+	async push(frame) {
+		if (this.state !== "open") {
+			this.dropped += 1;
+			return;
+		}
+		try {
+			if (!(await this.port.pushTalkAudio(frame)).accepted) this.dropped += 1;
+		} catch (err) {
+			this.dropped += 1;
+			if (this.state === "open") this.state = "idle";
+			this.logger.warn("talk-back: pushTalkAudio failed — reopening on the next frame", {
+				tags: { deviceId: this.deviceId },
+				meta: {
+					error: require_dist.errMsg(err),
+					sequence: frame.sequenceNumber
+				}
+			});
+		}
+	}
+};
+/**
+* Does the client's offer let the PEER send audio? (Its `m=audio` section is
+* `sendrecv` or `sendonly`; absent direction means `sendrecv`, RFC 3264.)
+*/
+function offerLetsPeerSendAudio(offerSdp) {
+	let inAudio = false;
+	let direction = null;
+	let sawAudio = false;
+	for (const raw of offerSdp.split(/\r?\n/)) {
+		const line = raw.trim();
+		if (line.startsWith("m=")) {
+			if (inAudio) break;
+			inAudio = line.startsWith("m=audio");
+			if (inAudio) sawAudio = true;
+			continue;
+		}
+		if (inAudio && /^a=(sendrecv|sendonly|recvonly|inactive)$/.test(line)) direction = line.slice(2);
+	}
+	if (!sawAudio) return false;
+	return direction === null || direction === "sendrecv" || direction === "sendonly";
+}
+/**
+* The direction to ANSWER a client offer's audio with: `sendrecv` only when
+* talk-back was asked for AND the peer offered to send audio; otherwise the
+* `sendonly` every viewer has always had.
+*/
+function clientOfferAudioDirection(talkback, offerSdp) {
+	return talkback && offerLetsPeerSendAudio(offerSdp) ? "sendrecv" : "sendonly";
+}
+/** The codec an inbound RTP packet carries, by its payload type. */
+function audioCodecForPayloadType(codecs, payloadType) {
+	switch (codecs.find((c) => c.payloadType === payloadType)?.mimeType.toLowerCase()) {
+		case "audio/opus": return "Opus";
+		case "audio/pcmu": return "Pcmu";
+		case "audio/pcma": return "Pcma";
+		default: return null;
+	}
+}
+//#endregion
 //#region src/stream-broker/webrtc/mdns-resolve.ts
 /**
 * Rewrite mDNS (`*.local`) candidates in an SDP by resolving each
@@ -23531,23 +23694,7 @@ var AdaptiveSession = class AdaptiveSession {
 		const audioDir = this.intercom ? "sendrecv" : "sendonly";
 		const audioTransceiver = this.pc.addTransceiver(this.audioTrack, { direction: audioDir });
 		this.audioSender = audioTransceiver.sender;
-		if (this.intercom) {
-			const cb = this.intercom.onAudioReceived;
-			audioTransceiver.onTrack.subscribe((track) => {
-				track.onReceiveRtp.subscribe((pkt) => {
-					try {
-						const payload = pkt.payload;
-						if (payload?.length > 0) cb(payload, "Opus");
-					} catch (err) {
-						this.logger.error("Intercom error", { meta: {
-							phase: "session",
-							sessionId: this.sessionId,
-							error: require_dist.errMsg(err)
-						} });
-					}
-				});
-			});
-		}
+		if (this.intercom) this.subscribeInboundAudio(audioTransceiver);
 		const offer = await this.pc.createOffer();
 		await this.pc.setLocalDescription(offer);
 		await this.waitForIceGatheringFast();
@@ -23647,6 +23794,33 @@ var AdaptiveSession = class AdaptiveSession {
 		this.startFeedingFrames();
 	}
 	/**
+	* Relay the peer's audio to `intercom.onAudioReceived`, naming each packet's
+	* codec from the payload type werift negotiated — never assumed: the offer
+	* carries PCMU as well as Opus, and a PCMU frame labelled Opus is noise at
+	* the camera. Packets of a codec the intercom path cannot take are dropped.
+	*/
+	subscribeInboundAudio(transceiver) {
+		const intercom = this.intercom;
+		if (intercom === void 0) return;
+		transceiver.onTrack.subscribe((track) => {
+			track.onReceiveRtp.subscribe((pkt) => {
+				try {
+					const payload = pkt.payload;
+					if (payload === void 0 || payload.length === 0) return;
+					const format = audioCodecForPayloadType(transceiver.codecs ?? [], pkt.header.payloadType);
+					if (format === null) return;
+					intercom.onAudioReceived(payload, format);
+				} catch (err) {
+					this.logger.error("Intercom error", { meta: {
+						phase: "session",
+						sessionId: this.sessionId,
+						error: require_dist.errMsg(err)
+					} });
+				}
+			});
+		});
+	}
+	/**
 	* Handle WHEP offer: client sends SDP offer, we create answer.
 	*
 	* Uses the server-creates-offer pattern internally: we create our own offer
@@ -23730,10 +23904,12 @@ var AdaptiveSession = class AdaptiveSession {
 					} });
 				}
 			} else if (kind === "audio" && !this.audioTrack) {
-				t.setDirection("sendonly");
+				const audioDirection = clientOfferAudioDirection(this.intercom !== void 0, clientOffer.sdp);
+				t.setDirection(audioDirection);
 				this.audioTrack = new werift.MediaStreamTrack({ kind: "audio" });
 				await t.sender.replaceTrack(this.audioTrack);
 				this.audioSender = t.sender;
+				if (audioDirection === "sendrecv") this.subscribeInboundAudio(t);
 			}
 		}
 		if (!this.videoTrack) {
@@ -33836,6 +34012,7 @@ var BrokerWebrtcServer = class {
 	staticIceServers;
 	icePortRange;
 	iceAdditionalHostAddresses;
+	talkbackIntercom;
 	clientNetworkStats;
 	locateRecordedSegment;
 	readRecordedSegmentBytes;
@@ -33938,6 +34115,7 @@ var BrokerWebrtcServer = class {
 		this.staticIceServers = options.iceServers;
 		this.icePortRange = options.icePortRange;
 		this.iceAdditionalHostAddresses = options.iceAdditionalHostAddresses;
+		this.talkbackIntercom = options.talkbackIntercom;
 		this.clientNetworkStats = options.clientNetworkStats;
 		this.locateRecordedSegment = options.locateRecordedSegment;
 		this.readRecordedSegmentBytes = options.readRecordedSegmentBytes;
@@ -34460,9 +34638,12 @@ var BrokerWebrtcServer = class {
 			sourceProfileLevelId,
 			offeredProfileLevelIds
 		} });
+		const talkback = opts.talkback === true && this.talkbackIntercom !== void 0 ? new TalkbackBridge(deviceId, this.talkbackIntercom, sessionLogger) : void 0;
+		if (opts.talkback === true && talkback === void 0) sessionLogger.warn("talk-back requested but no intercom port is wired — audio stays one-way", { meta: { sessionId } });
 		session = new AdaptiveSession({
 			sessionId,
 			source,
+			...talkback !== void 0 ? { intercom: { onAudioReceived: (data, format) => talkback.onAudio(data, format) } } : {},
 			sourceCodec: sessionCodec,
 			transcodeToBaseline,
 			transcodeProfile: effectiveTranscodeProfile,
@@ -34558,6 +34739,7 @@ var BrokerWebrtcServer = class {
 			unsubSdpParams,
 			unsubSourceRestart,
 			closeSource,
+			...talkback !== void 0 ? { closeTalkback: () => void talkback.close() } : {},
 			rebind: {
 				deviceId,
 				sessionCodec,
@@ -35787,6 +35969,7 @@ var BrokerWebrtcServer = class {
 		entry.unsubSourceRestart();
 		entry.releaseOpusEgress?.();
 		entry.releaseOpusEgress = void 0;
+		entry.closeTalkback?.();
 		entry.closeSource();
 	}
 	resolveBrokerId(streamId, hints = {}) {
@@ -36392,6 +36575,7 @@ var WebrtcSessionProvider = class WebrtcSessionProvider {
 				nonTrickle: input.nonTrickle === true,
 				disableTurn: input.disableTurn === true,
 				disableIpv6: input.disableIpv6 === true,
+				talkback: input.talkback === true,
 				transcodeProfile,
 				...input.consumerAttribution !== void 0 ? { consumerAttribution: input.consumerAttribution } : {}
 			});
@@ -36884,8 +37068,25 @@ var StreamBrokerAddon = class extends require_dist.BaseAddon {
 		});
 		const cameraStreamsProvider = new CameraStreamsProvider(this.brokerManager);
 		const turnApi = this.ctx.api;
+		const intercomApi = this.ctx.api.intercom;
 		const webrtcServer = new BrokerWebrtcServer({
 			logger: this.ctx.logger.child("webrtc"),
+			talkbackIntercom: {
+				startTalkSession: async (deviceId) => {
+					await intercomApi.startTalkSession.mutate({ deviceId });
+				},
+				pushTalkAudio: (frame) => intercomApi.pushTalkAudio.mutate({
+					deviceId: frame.deviceId,
+					audioBase64: frame.audioBase64,
+					codec: frame.codec,
+					sampleRate: frame.sampleRate,
+					channels: frame.channels,
+					sequenceNumber: frame.sequenceNumber
+				}),
+				endTalkSession: async (deviceId) => {
+					await intercomApi.endTalkSession.mutate({ deviceId });
+				}
+			},
 			getIceServers: async () => {
 				if (!turnApi.turnProvider) return [];
 				try {

package/dist/stream-broker/index.mjs

@@ -1,12 +1,12 @@
 import { r as __toESM } from "../chunk-DnnnRqeS.mjs";
-import { An as CAM_PROFILE_ORDER, Bn as makeSourceBrokerId, C as EncodeProfileSchema, Cn as invocationFromEncodeProfile, Dn as errMsg, E as G711_SCALE_CORRECTION_DB, En as DeviceType, F as RATE_CONTROL_TIGHT, Gn as _enum, H as RingBuffer, Hn as parseProfileBrokerId, In as expandAudioChunkToF32le, Jn as discriminatedUnion, K as addonWidgetsSourceCapability, Kn as array, Lt as memoizePinnedFfmpeg, Mt as logChannelsCapability, Nn as createEvent, On as BOOT_RECOVERY_BACKOFF_MS, P as RATE_CONTROL_RELAXED, Pt as maskUrlCredentials, Q as batteryCapability, Qn as object, Rn as isEvent, St as ffmpegRate, Tn as DeviceFeature, Tt as gifFromMp4Args, Vn as nodePin, Xn as literal, Yt as rankStreams, Zn as number, _n as webrtcSessionCapability, at as createHwAccelCache, c as CLIP_STILL_REASON_CODE_HEADER, er as record, et as cameraStreamsCapability, ht as egressTransportFromRequest, jn as CamProfileSchema, jt as loadContributionCapability, kn as BaseAddon, lt as deriveBatteryPresence, mt as egressTranscodeSharingKey, nn as resolveEgressDecodeHwAccel, nr as union, nt as clipExportFailureOf, o as BatteryStatusSchema, ot as createLogChannelsProvider, pn as streamBrokerCapability, qn as boolean, rr as EventCategory, s as CLIP_BROKER_PACED_RATES, st as declareLogChannel, tr as string, vn as AUDIO_PRESETS, wn as isSoftwareDecode, wt as gifDeriveSpeed, xn as buildFfmpegArgs, zn as makeProfileBrokerId } from "../dist-DUr_0s5c.mjs";
-import { i as NO_COST_CLAIM, n as Fmp4FragmentChild, o as probeFfmpegBinary, r as Fmp4FragmentPlane, t as ChildCostRegistry } from "../node-DhrskFZ_.mjs";
+import { $n as discriminatedUnion, An as DeviceFeature, At as gifFromMp4Args, Cn as AUDIO_PRESETS, D as G711_SCALE_CORRECTION_DB, Dt as ffmpegRate, En as buildFfmpegArgs, Fn as CAM_PROFILE_ORDER, Gn as makeSourceBrokerId, Ht as memoizePinnedFfmpeg, In as CamProfileSchema, It as loadContributionCapability, K as RingBuffer, Kn as nodePin, Lt as logChannelsCapability, Mn as errMsg, Nn as BOOT_RECOVERY_BACKOFF_MS, On as invocationFromEncodeProfile, Pn as BaseAddon, Qn as boolean, R as RATE_CONTROL_RELAXED, Rn as createEvent, Sn as webrtcSessionCapability, Un as isEvent, Vn as expandAudioChunkToF32le, Wn as makeProfileBrokerId, X as addonWidgetsSourceCapability, Xn as _enum, Zn as array, ar as record, at as cameraStreamsCapability, bt as egressTransportFromRequest, c as CLIP_STILL_REASON_CODE_HEADER, cr as EventCategory, dt as createLogChannelsProvider, en as rankStreams, ft as declareLogChannel, jn as DeviceType, kn as isSoftwareDecode, kt as gifDeriveSpeed, mt as deriveBatteryPresence, nr as number, nt as batteryCapability, o as BatteryStatusSchema, or as string, qn as parseProfileBrokerId, rr as object, s as CLIP_BROKER_PACED_RATES, sn as resolveEgressDecodeHwAccel, sr as union, st as clipExportFailureOf, tr as literal, ut as createHwAccelCache, vn as streamBrokerCapability, w as EncodeProfileSchema, yt as egressTranscodeSharingKey, z as RATE_CONTROL_TIGHT, zt as maskUrlCredentials } from "../dist-Dj26w1od.mjs";
+import { i as NO_COST_CLAIM, n as Fmp4FragmentChild, o as probeFfmpegBinary, r as Fmp4FragmentPlane, t as ChildCostRegistry } from "../node-Cc9bQ8ZM.mjs";
 import { a as parseRestreamPath, d as PREBUFFER_MAX_MS, f as capPreRollRequestMs, i as isLiveEdgeOnlyDial, m as resolvePrebuffer, n as NO_DECLARATION, p as prebufferRetentionMs, s as sdpVariantFor, u as withSentryIntent } from "../restream-intent-DyGmLAU9.mjs";
 import { n as profileForStreamId } from "../remote-restream-BeHi78PZ.mjs";
 import { n as RowMapStore, t as retireRootKeys } from "../retire-root-keys-DZvlso2l.mjs";
-import { a as require_mp4box_all, c as parseNalLengthSize, i as runFfmpegOnce, n as demuxSegmentPreferJs, o as avccSampleToAnnexB, r as FfmpegFallbackRefusedError, s as buildKeyframeAnnexB, t as demuxSegmentFullJs } from "../segment-demux-js-Bv7s-_uv.mjs";
+import { a as require_mp4box_all, c as parseNalLengthSize, i as runFfmpegOnce, n as demuxSegmentPreferJs, o as avccSampleToAnnexB, r as FfmpegFallbackRefusedError, s as buildKeyframeAnnexB, t as demuxSegmentFullJs } from "../segment-demux-js-CmuAuckE.mjs";
 import { createRequire } from "node:module";
-import * as os from "node:os";
+import * as os$1 from "node:os";
 import { networkInterfaces, tmpdir } from "node:os";
 import * as crypto$1 from "node:crypto";
 import crypto, { createHash, randomBytes, randomUUID } from "node:crypto";
@@ -21333,6 +21333,169 @@ function shouldWriteLossLine(input) {
 	};
 }
 //#endregion
+//#region src/stream-broker/webrtc/talkback-bridge.ts
+/** A frame this small is Opus DTX / comfort noise, not speech. */
+var DTX_MAX_BYTES = 3;
+/** How the push-talk triple names each WebRTC audio codec, or null when the
+*  intercom path cannot take it. */
+function talkCodecOf(format) {
+	switch (format) {
+		case "Opus": return {
+			codec: "opus",
+			sampleRate: 48e3
+		};
+		case "Pcmu": return {
+			codec: "g711ulaw",
+			sampleRate: 8e3
+		};
+		case "Pcma": return {
+			codec: "g711alaw",
+			sampleRate: 8e3
+		};
+		default: return null;
+	}
+}
+var TalkbackBridge = class {
+	deviceId;
+	port;
+	logger;
+	state = "idle";
+	/** Set by `close()`; no frame is accepted after it. */
+	closing = false;
+	/** A talk session was opened at the camera and must be ended. */
+	started = false;
+	sequence = 0;
+	chain = Promise.resolve();
+	dropped = 0;
+	refusedCodecLogged = false;
+	constructor(deviceId, port, logger) {
+		this.deviceId = deviceId;
+		this.port = port;
+		this.logger = logger;
+	}
+	/** One inbound RTP payload from the peer. Never throws. */
+	onAudio(payload, format) {
+		if (this.closing) return;
+		if (payload.length <= DTX_MAX_BYTES) return;
+		const talk = talkCodecOf(format);
+		if (talk === null) {
+			if (!this.refusedCodecLogged) {
+				this.refusedCodecLogged = true;
+				this.logger.warn("talk-back: the peer sends a codec the intercom path cannot take", {
+					tags: { deviceId: this.deviceId },
+					meta: { format }
+				});
+			}
+			return;
+		}
+		if (this.state === "idle") {
+			this.state = "opening";
+			this.chain = this.chain.then(() => this.open());
+		}
+		const frame = {
+			deviceId: this.deviceId,
+			audioBase64: payload.toString("base64"),
+			codec: talk.codec,
+			sampleRate: talk.sampleRate,
+			channels: 1,
+			sequenceNumber: ++this.sequence
+		};
+		this.chain = this.chain.then(() => this.push(frame));
+	}
+	/** End the talk session (if one opened). Idempotent. */
+	async close() {
+		if (this.closing) return;
+		this.closing = true;
+		await this.chain.catch(() => {});
+		if (!this.started) return;
+		await this.port.endTalkSession(this.deviceId).catch((err) => {
+			this.logger.warn("talk-back: endTalkSession failed", {
+				tags: { deviceId: this.deviceId },
+				meta: { error: errMsg(err) }
+			});
+		});
+		this.logger.info("talk-back: ended", {
+			tags: { deviceId: this.deviceId },
+			meta: {
+				frames: this.sequence,
+				dropped: this.dropped
+			}
+		});
+	}
+	async open() {
+		if (this.state !== "opening") return;
+		try {
+			await this.port.startTalkSession(this.deviceId);
+			this.started = true;
+			if (this.state === "opening") this.state = "open";
+			this.logger.info("talk-back: the peer is talking — camera talk session opened", { tags: { deviceId: this.deviceId } });
+		} catch (err) {
+			if (this.state === "opening") this.state = "idle";
+			this.logger.warn("talk-back: startTalkSession failed — will retry on the next frame", {
+				tags: { deviceId: this.deviceId },
+				meta: { error: errMsg(err) }
+			});
+		}
+	}
+	async push(frame) {
+		if (this.state !== "open") {
+			this.dropped += 1;
+			return;
+		}
+		try {
+			if (!(await this.port.pushTalkAudio(frame)).accepted) this.dropped += 1;
+		} catch (err) {
+			this.dropped += 1;
+			if (this.state === "open") this.state = "idle";
+			this.logger.warn("talk-back: pushTalkAudio failed — reopening on the next frame", {
+				tags: { deviceId: this.deviceId },
+				meta: {
+					error: errMsg(err),
+					sequence: frame.sequenceNumber
+				}
+			});
+		}
+	}
+};
+/**
+* Does the client's offer let the PEER send audio? (Its `m=audio` section is
+* `sendrecv` or `sendonly`; absent direction means `sendrecv`, RFC 3264.)
+*/
+function offerLetsPeerSendAudio(offerSdp) {
+	let inAudio = false;
+	let direction = null;
+	let sawAudio = false;
+	for (const raw of offerSdp.split(/\r?\n/)) {
+		const line = raw.trim();
+		if (line.startsWith("m=")) {
+			if (inAudio) break;
+			inAudio = line.startsWith("m=audio");
+			if (inAudio) sawAudio = true;
+			continue;
+		}
+		if (inAudio && /^a=(sendrecv|sendonly|recvonly|inactive)$/.test(line)) direction = line.slice(2);
+	}
+	if (!sawAudio) return false;
+	return direction === null || direction === "sendrecv" || direction === "sendonly";
+}
+/**
+* The direction to ANSWER a client offer's audio with: `sendrecv` only when
+* talk-back was asked for AND the peer offered to send audio; otherwise the
+* `sendonly` every viewer has always had.
+*/
+function clientOfferAudioDirection(talkback, offerSdp) {
+	return talkback && offerLetsPeerSendAudio(offerSdp) ? "sendrecv" : "sendonly";
+}
+/** The codec an inbound RTP packet carries, by its payload type. */
+function audioCodecForPayloadType(codecs, payloadType) {
+	switch (codecs.find((c) => c.payloadType === payloadType)?.mimeType.toLowerCase()) {
+		case "audio/opus": return "Opus";
+		case "audio/pcmu": return "Pcmu";
+		case "audio/pcma": return "Pcma";
+		default: return null;
+	}
+}
+//#endregion
 //#region src/stream-broker/webrtc/mdns-resolve.ts
 /**
 * Rewrite mDNS (`*.local`) candidates in an SDP by resolving each
@@ -23527,23 +23690,7 @@ var AdaptiveSession = class AdaptiveSession {
 		const audioDir = this.intercom ? "sendrecv" : "sendonly";
 		const audioTransceiver = this.pc.addTransceiver(this.audioTrack, { direction: audioDir });
 		this.audioSender = audioTransceiver.sender;
-		if (this.intercom) {
-			const cb = this.intercom.onAudioReceived;
-			audioTransceiver.onTrack.subscribe((track) => {
-				track.onReceiveRtp.subscribe((pkt) => {
-					try {
-						const payload = pkt.payload;
-						if (payload?.length > 0) cb(payload, "Opus");
-					} catch (err) {
-						this.logger.error("Intercom error", { meta: {
-							phase: "session",
-							sessionId: this.sessionId,
-							error: errMsg(err)
-						} });
-					}
-				});
-			});
-		}
+		if (this.intercom) this.subscribeInboundAudio(audioTransceiver);
 		const offer = await this.pc.createOffer();
 		await this.pc.setLocalDescription(offer);
 		await this.waitForIceGatheringFast();
@@ -23643,6 +23790,33 @@ var AdaptiveSession = class AdaptiveSession {
 		this.startFeedingFrames();
 	}
 	/**
+	* Relay the peer's audio to `intercom.onAudioReceived`, naming each packet's
+	* codec from the payload type werift negotiated — never assumed: the offer
+	* carries PCMU as well as Opus, and a PCMU frame labelled Opus is noise at
+	* the camera. Packets of a codec the intercom path cannot take are dropped.
+	*/
+	subscribeInboundAudio(transceiver) {
+		const intercom = this.intercom;
+		if (intercom === void 0) return;
+		transceiver.onTrack.subscribe((track) => {
+			track.onReceiveRtp.subscribe((pkt) => {
+				try {
+					const payload = pkt.payload;
+					if (payload === void 0 || payload.length === 0) return;
+					const format = audioCodecForPayloadType(transceiver.codecs ?? [], pkt.header.payloadType);
+					if (format === null) return;
+					intercom.onAudioReceived(payload, format);
+				} catch (err) {
+					this.logger.error("Intercom error", { meta: {
+						phase: "session",
+						sessionId: this.sessionId,
+						error: errMsg(err)
+					} });
+				}
+			});
+		});
+	}
+	/**
 	* Handle WHEP offer: client sends SDP offer, we create answer.
 	*
 	* Uses the server-creates-offer pattern internally: we create our own offer
@@ -23726,10 +23900,12 @@ var AdaptiveSession = class AdaptiveSession {
 					} });
 				}
 			} else if (kind === "audio" && !this.audioTrack) {
-				t.setDirection("sendonly");
+				const audioDirection = clientOfferAudioDirection(this.intercom !== void 0, clientOffer.sdp);
+				t.setDirection(audioDirection);
 				this.audioTrack = new werift.MediaStreamTrack({ kind: "audio" });
 				await t.sender.replaceTrack(this.audioTrack);
 				this.audioSender = t.sender;
+				if (audioDirection === "sendrecv") this.subscribeInboundAudio(t);
 			}
 		}
 		if (!this.videoTrack) {
@@ -33832,6 +34008,7 @@ var BrokerWebrtcServer = class {
 	staticIceServers;
 	icePortRange;
 	iceAdditionalHostAddresses;
+	talkbackIntercom;
 	clientNetworkStats;
 	locateRecordedSegment;
 	readRecordedSegmentBytes;
@@ -33934,6 +34111,7 @@ var BrokerWebrtcServer = class {
 		this.staticIceServers = options.iceServers;
 		this.icePortRange = options.icePortRange;
 		this.iceAdditionalHostAddresses = options.iceAdditionalHostAddresses;
+		this.talkbackIntercom = options.talkbackIntercom;
 		this.clientNetworkStats = options.clientNetworkStats;
 		this.locateRecordedSegment = options.locateRecordedSegment;
 		this.readRecordedSegmentBytes = options.readRecordedSegmentBytes;
@@ -34456,9 +34634,12 @@ var BrokerWebrtcServer = class {
 			sourceProfileLevelId,
 			offeredProfileLevelIds
 		} });
+		const talkback = opts.talkback === true && this.talkbackIntercom !== void 0 ? new TalkbackBridge(deviceId, this.talkbackIntercom, sessionLogger) : void 0;
+		if (opts.talkback === true && talkback === void 0) sessionLogger.warn("talk-back requested but no intercom port is wired — audio stays one-way", { meta: { sessionId } });
 		session = new AdaptiveSession({
 			sessionId,
 			source,
+			...talkback !== void 0 ? { intercom: { onAudioReceived: (data, format) => talkback.onAudio(data, format) } } : {},
 			sourceCodec: sessionCodec,
 			transcodeToBaseline,
 			transcodeProfile: effectiveTranscodeProfile,
@@ -34554,6 +34735,7 @@ var BrokerWebrtcServer = class {
 			unsubSdpParams,
 			unsubSourceRestart,
 			closeSource,
+			...talkback !== void 0 ? { closeTalkback: () => void talkback.close() } : {},
 			rebind: {
 				deviceId,
 				sessionCodec,
@@ -35783,6 +35965,7 @@ var BrokerWebrtcServer = class {
 		entry.unsubSourceRestart();
 		entry.releaseOpusEgress?.();
 		entry.releaseOpusEgress = void 0;
+		entry.closeTalkback?.();
 		entry.closeSource();
 	}
 	resolveBrokerId(streamId, hints = {}) {
@@ -36388,6 +36571,7 @@ var WebrtcSessionProvider = class WebrtcSessionProvider {
 				nonTrickle: input.nonTrickle === true,
 				disableTurn: input.disableTurn === true,
 				disableIpv6: input.disableIpv6 === true,
+				talkback: input.talkback === true,
 				transcodeProfile,
 				...input.consumerAttribution !== void 0 ? { consumerAttribution: input.consumerAttribution } : {}
 			});
@@ -36880,8 +37064,25 @@ var StreamBrokerAddon = class extends BaseAddon {
 		});
 		const cameraStreamsProvider = new CameraStreamsProvider(this.brokerManager);
 		const turnApi = this.ctx.api;
+		const intercomApi = this.ctx.api.intercom;
 		const webrtcServer = new BrokerWebrtcServer({
 			logger: this.ctx.logger.child("webrtc"),
+			talkbackIntercom: {
+				startTalkSession: async (deviceId) => {
+					await intercomApi.startTalkSession.mutate({ deviceId });
+				},
+				pushTalkAudio: (frame) => intercomApi.pushTalkAudio.mutate({
+					deviceId: frame.deviceId,
+					audioBase64: frame.audioBase64,
+					codec: frame.codec,
+					sampleRate: frame.sampleRate,
+					channels: frame.channels,
+					sequenceNumber: frame.sequenceNumber
+				}),
+				endTalkSession: async (deviceId) => {
+					await intercomApi.endTalkSession.mutate({ deviceId });
+				}
+			},
 			getIceServers: async () => {
 				if (!turnApi.turnProvider) return [];
 				try {
@@ -37626,7 +37827,7 @@ var URL_SCHEME$1 = "unix://";
 var UdsTransport = class {
 	kind = "uds";
 	async listen(options) {
-		const socketPath = options?.socketPath ?? path$1.join(os.tmpdir(), `camstack-frame-${randomUUID()}.sock`);
+		const socketPath = options?.socketPath ?? path$1.join(os$1.tmpdir(), `camstack-frame-${randomUUID()}.sock`);
 		try {
 			fs.unlinkSync(socketPath);
 		} catch (err) {

package/dist/stream-broker/remoteEntry.js

@@ -30,7 +30,7 @@ async function d(e) {
 	}
 }
 async function f() {
-	return l ||= d(() => import("./_virtual_mf-localSharedImportMap___mfe_internal__addon_stream_broker_widgets-BH3o1g9i.mjs")).catch((e) => {
+	return l ||= d(() => import("./_virtual_mf-localSharedImportMap___mfe_internal__addon_stream_broker_widgets-BX73joE9.mjs")).catch((e) => {
 		throw l = void 0, e;
 	}), l;
 }