@signalwire/js 4.0.0-rc.0 → 4.0.0-rc.2

package/dist/browser.mjs

@@ -9015,6 +9015,143 @@ var Destroyable = class {
 	}
 };
 
+//#endregion
+//#region src/core/constants.ts
+const INVITE_VERSION = 1e3;
+const DEFAULT_ICE_CANDIDATE_TIMEOUT_MS = 600;
+const DEFAULT_ICE_GATHERING_TIMEOUT_MS = 6e3;
+const DEFAULT_RECONNECT_CALLS_TIMEOUT_MS = 300 * 1e3;
+const DEFAULT_CONNECTION_TIMEOUT_MS = 1e4;
+const DEFAULT_RECONNECT_DELAY_MIN_MS = 100;
+const DEFAULT_RECONNECT_DELAY_MAX_MS = 3e3;
+const DEFAULT_DEVICE_DEBOUNCE_TIME_MS = 1500;
+const DEFAULT_DEVICE_POLLING_INTERVAL_MS = 0;
+const PREFERENCES_STORAGE_KEY = "sw:preferences";
+/** Scope value that enables automatic token refresh. */
+const SAT_REFRESH_SCOPE = "sat:refresh";
+/** API endpoints for device token operations. */
+const DEVICE_TOKEN_ENDPOINT = "/api/fabric/subscriber/devices/token";
+const DEVICE_REFRESH_ENDPOINT = "/api/fabric/subscriber/devices/refresh";
+/** Default device token TTL in seconds (15 minutes). */
+const DEVICE_TOKEN_DEFAULT_EXPIRE_IN = 900;
+/** Buffer time in milliseconds before expiry to trigger refresh. */
+const DEVICE_TOKEN_REFRESH_BUFFER_MS = 3e4;
+/** Maximum retry attempts for device token refresh on transient failure. */
+const DEVICE_TOKEN_REFRESH_MAX_RETRIES = 3;
+/** Base delay in milliseconds for exponential backoff on refresh retry. */
+const DEVICE_TOKEN_REFRESH_RETRY_BASE_MS = 1e3;
+/** Maximum retry attempts for developer credential refresh on transient failure. */
+const CREDENTIAL_REFRESH_MAX_RETRIES = 5;
+/** Base delay in milliseconds for exponential backoff on credential refresh retry. */
+const CREDENTIAL_REFRESH_RETRY_BASE_MS = 1e3;
+/** Maximum delay in milliseconds for credential refresh backoff. */
+const CREDENTIAL_REFRESH_MAX_DELAY_MS = 3e4;
+/** Buffer in milliseconds before token expiry to trigger refresh. */
+const CREDENTIAL_REFRESH_BUFFER_MS = 5e3;
+/**
+* Maximum time the coordinator will wait for `DeviceTokenManager.activate()`
+* to resolve before treating the activation as failed and falling back to
+* the developer-provided refresh path. Prevents a wedged HTTP layer from
+* leaving the session with no active refresh mechanism.
+*/
+const CREDENTIAL_ACTIVATE_TIMEOUT_MS = 3e4;
+/** JSON-RPC error code for requester validation failure (corrupted auth state). */
+const RPC_ERROR_REQUESTER_VALIDATION_FAILED = -32003;
+/** JSON-RPC error code for invalid params (e.g., missing authentication block). */
+const RPC_ERROR_INVALID_PARAMS = -32602;
+/** JSON-RPC error code for authentication failure (invalid token, missing DPoP, etc.). */
+const RPC_ERROR_AUTHENTICATION_FAILED = -32002;
+/** Error names browsers use for a media permission denial (user or policy). */
+const MEDIA_ACCESS_DENIAL_NAMES = [
+	"NotAllowedError",
+	"SecurityError",
+	"PermissionDeniedError"
+];
+/** Default polling interval for RTCPeerConnection.getStats() in milliseconds. */
+const DEFAULT_STATS_POLLING_INTERVAL_MS = 1e3;
+/** Number of initial samples used to build a baseline for spike detection. */
+const DEFAULT_STATS_BASELINE_SAMPLES = 10;
+/** Duration in ms with no inbound audio packets before emitting a critical issue. */
+const DEFAULT_STATS_NO_PACKET_THRESHOLD_MS = 2e3;
+/** Multiplier applied to baseline RTT to detect a warning-level RTT spike. */
+const DEFAULT_STATS_RTT_SPIKE_MULTIPLIER = 3;
+/** Packet loss fraction (0-1) above which a warning is emitted. */
+const DEFAULT_STATS_PACKET_LOSS_THRESHOLD = .05;
+/** Multiplier applied to baseline jitter to detect a jitter spike. */
+const DEFAULT_STATS_JITTER_SPIKE_MULTIPLIER = 4;
+/** Number of seconds of metrics history to retain. */
+const DEFAULT_STATS_HISTORY_SIZE = 30;
+/** Maximum keyframe requests allowed within a single burst window. */
+const DEFAULT_KEYFRAME_MAX_BURST$1 = 3;
+/** Duration of the keyframe burst window in milliseconds. */
+const DEFAULT_KEYFRAME_BURST_WINDOW_MS$1 = 3e3;
+/** Cooldown period in ms after burst limit is reached before allowing more keyframes. */
+const DEFAULT_KEYFRAME_COOLDOWN_MS$1 = 1e4;
+/** Minimum time between re-INVITE attempts in milliseconds. */
+const DEFAULT_REINVITE_DEBOUNCE_TIME_MS = 1e4;
+/** Maximum number of re-INVITE attempts per call. */
+const DEFAULT_REINVITE_MAX_ATTEMPTS = 3;
+/** Timeout for a single re-INVITE attempt in milliseconds. */
+const DEFAULT_REINVITE_TIMEOUT_MS = 5e3;
+/** Debounce window in ms to collapse multiple detection signals into one trigger. */
+const DEFAULT_RECOVERY_DEBOUNCE_TIME_MS = 2e3;
+/** Cooldown period in ms between recovery attempts. */
+const DEFAULT_RECOVERY_COOLDOWN_MS = 1e4;
+/** Grace period in ms before treating ICE 'disconnected' as a failure. */
+const DEFAULT_ICE_DISCONNECTED_GRACE_PERIOD_MS = 3e3;
+/** Timeout for a single ICE restart attempt in milliseconds. */
+const DEFAULT_ICE_RESTART_TIMEOUT_MS$1 = 5e3;
+/** Maximum recovery attempts before emitting 'max_attempts_reached'. */
+const DEFAULT_MAX_RECOVERY_ATTEMPTS = 3;
+/** Upper bound in ms for waiting on iceGatheringState === 'complete' after an ICE restart. */
+const ICE_GATHERING_COMPLETE_TIMEOUT_MS = 1e4;
+/** Upper bound in ms for waiting on RTCPeerConnection.connectionState === 'connected' after a recovery ICE restart. */
+const PEER_CONNECTION_RECOVERY_WAIT_MS = 5e3;
+/** Polling interval in ms while waiting for RTCPeerConnection.connectionState to transition. */
+const PEER_CONNECTION_RECOVERY_POLL_MS = 100;
+/** Polling interval for LocalAudioPipeline.level$ (ms). ~30fps is smooth for meters. */
+const AUDIO_LEVEL_POLL_INTERVAL_MS = 33;
+/** RMS level threshold (0..1) above which the local participant is considered speaking. */
+const VAD_THRESHOLD = .03;
+/** Hold window in ms below the threshold before speaking$ flips back to false. */
+const VAD_HOLD_MS = 250;
+/** Whether to persist device selections to storage by default. */
+const DEFAULT_PERSIST_DEVICE_SELECTION = true;
+/** Whether to auto-apply device changes to active calls by default. */
+const DEFAULT_SYNC_DEVICES_TO_ACTIVE_CALLS = true;
+/** Storage keys for persisted device selections. */
+const DEVICE_STORAGE_KEY_AUDIO_INPUT = "sw:device:audioinput";
+const DEVICE_STORAGE_KEY_AUDIO_OUTPUT = "sw:device:audiooutput";
+const DEVICE_STORAGE_KEY_VIDEO_INPUT = "sw:device:videoinput";
+/** Whether to auto-mute video when the tab becomes hidden. */
+const DEFAULT_AUTO_MUTE_VIDEO_ON_HIDDEN = false;
+/** Whether to re-enumerate devices when the page becomes visible. */
+const DEFAULT_REFRESH_DEVICES_ON_VISIBLE = true;
+/** Whether to check peer connection health when the page becomes visible. */
+const DEFAULT_CHECK_CONNECTION_ON_VISIBLE = true;
+/** Whether automatic video degradation on low bandwidth is enabled. */
+const DEFAULT_ENABLE_AUTO_DEGRADATION = true;
+/** Bitrate in kbps below which video is automatically disabled. */
+const DEFAULT_DEGRADATION_BITRATE_THRESHOLD_KBPS = 150;
+/** Bitrate in kbps above which video is automatically re-enabled (hysteresis). */
+const DEFAULT_DEGRADATION_RECOVERY_THRESHOLD_KBPS = 300;
+/** Whether relay-only escalation is enabled as a last-resort recovery tier. */
+const DEFAULT_ENABLE_RELAY_FALLBACK = true;
+/** Whether to listen for browser online/offline/connection events. */
+const DEFAULT_ENABLE_NETWORK_CHANGE_DETECTION = true;
+/** Whether to intercept server-sent media-timeout hangups and attempt recovery. */
+const DEFAULT_ENABLE_SERVER_HANGUP_INTERCEPTION = true;
+/** Default video track constraints applied when video is enabled without explicit constraints. */
+const DEFAULT_VIDEO_CONSTRAINTS = {
+	width: { ideal: 1280 },
+	height: { ideal: 720 },
+	aspectRatio: 16 / 9
+};
+/** Whether stereo Opus is enabled by default. */
+const DEFAULT_STEREO_AUDIO = false;
+/** Max average bitrate for stereo Opus in bits per second. */
+const DEFAULT_STEREO_MAX_AVERAGE_BITRATE = 51e4;
+
 //#endregion
 //#region src/core/errors.ts
 var UnexpectedError = class extends Error {
@@ -9224,6 +9361,33 @@ var MediaTrackError = class extends Error {
 		this.name = "MediaTrackError";
 	}
 };
+/** True when a `getUserMedia`/`getDisplayMedia` rejection is a permission denial. */
+function isMediaAccessDenial(originalError) {
+	return originalError instanceof Error && MEDIA_ACCESS_DENIAL_NAMES.includes(originalError.name);
+}
+/**
+* Failure to acquire local media (camera, microphone, or screen capture)
+* via `getUserMedia`/`getDisplayMedia`.
+*
+* Non-fatal by default: screenshare and additional-device failures never
+* end the call, and main-connection failures degrade to receive-only when
+* possible. The wrapping site sets `fatal` to `true` only when the call
+* cannot continue (receive-only fallback disabled or no receive intent).
+*/
+var MediaAccessError = class extends Error {
+	constructor(operation, media, originalError, fatal = false) {
+		super(`Media access ${isMediaAccessDenial(originalError) ? "denied" : "failed"} for ${operation} (${media})`, { cause: originalError instanceof Error ? originalError : void 0 });
+		this.operation = operation;
+		this.media = media;
+		this.originalError = originalError;
+		this.fatal = fatal;
+		this.name = "MediaAccessError";
+	}
+	/** True when the underlying failure is a permission denial (user or policy). */
+	get denied() {
+		return isMediaAccessDenial(this.originalError);
+	}
+};
 var DPoPInitError = class extends Error {
 	constructor(originalError, message = "Failed to initialize DPoP key pair") {
 		super(message, { cause: originalError instanceof Error ? originalError : void 0 });
@@ -9873,137 +10037,6 @@ var DeviceHistoryManager = class {
 	}
 };
 
-//#endregion
-//#region src/core/constants.ts
-const INVITE_VERSION = 1e3;
-const DEFAULT_ICE_CANDIDATE_TIMEOUT_MS = 600;
-const DEFAULT_ICE_GATHERING_TIMEOUT_MS = 6e3;
-const DEFAULT_RECONNECT_CALLS_TIMEOUT_MS = 300 * 1e3;
-const DEFAULT_CONNECTION_TIMEOUT_MS = 1e4;
-const DEFAULT_RECONNECT_DELAY_MIN_MS = 100;
-const DEFAULT_RECONNECT_DELAY_MAX_MS = 3e3;
-const DEFAULT_DEVICE_DEBOUNCE_TIME_MS = 1500;
-const DEFAULT_DEVICE_POLLING_INTERVAL_MS = 0;
-const PREFERENCES_STORAGE_KEY = "sw:preferences";
-/** Scope value that enables automatic token refresh. */
-const SAT_REFRESH_SCOPE = "sat:refresh";
-/** API endpoints for device token operations. */
-const DEVICE_TOKEN_ENDPOINT = "/api/fabric/subscriber/devices/token";
-const DEVICE_REFRESH_ENDPOINT = "/api/fabric/subscriber/devices/refresh";
-/** Default device token TTL in seconds (15 minutes). */
-const DEVICE_TOKEN_DEFAULT_EXPIRE_IN = 900;
-/** Buffer time in milliseconds before expiry to trigger refresh. */
-const DEVICE_TOKEN_REFRESH_BUFFER_MS = 3e4;
-/** Maximum retry attempts for device token refresh on transient failure. */
-const DEVICE_TOKEN_REFRESH_MAX_RETRIES = 3;
-/** Base delay in milliseconds for exponential backoff on refresh retry. */
-const DEVICE_TOKEN_REFRESH_RETRY_BASE_MS = 1e3;
-/** Maximum retry attempts for developer credential refresh on transient failure. */
-const CREDENTIAL_REFRESH_MAX_RETRIES = 5;
-/** Base delay in milliseconds for exponential backoff on credential refresh retry. */
-const CREDENTIAL_REFRESH_RETRY_BASE_MS = 1e3;
-/** Maximum delay in milliseconds for credential refresh backoff. */
-const CREDENTIAL_REFRESH_MAX_DELAY_MS = 3e4;
-/** Buffer in milliseconds before token expiry to trigger refresh. */
-const CREDENTIAL_REFRESH_BUFFER_MS = 5e3;
-/**
-* Maximum time the coordinator will wait for `DeviceTokenManager.activate()`
-* to resolve before treating the activation as failed and falling back to
-* the developer-provided refresh path. Prevents a wedged HTTP layer from
-* leaving the session with no active refresh mechanism.
-*/
-const CREDENTIAL_ACTIVATE_TIMEOUT_MS = 3e4;
-/** JSON-RPC error code for requester validation failure (corrupted auth state). */
-const RPC_ERROR_REQUESTER_VALIDATION_FAILED = -32003;
-/** JSON-RPC error code for invalid params (e.g., missing authentication block). */
-const RPC_ERROR_INVALID_PARAMS = -32602;
-/** JSON-RPC error code for authentication failure (invalid token, missing DPoP, etc.). */
-const RPC_ERROR_AUTHENTICATION_FAILED = -32002;
-/** Default polling interval for RTCPeerConnection.getStats() in milliseconds. */
-const DEFAULT_STATS_POLLING_INTERVAL_MS = 1e3;
-/** Number of initial samples used to build a baseline for spike detection. */
-const DEFAULT_STATS_BASELINE_SAMPLES = 10;
-/** Duration in ms with no inbound audio packets before emitting a critical issue. */
-const DEFAULT_STATS_NO_PACKET_THRESHOLD_MS = 2e3;
-/** Multiplier applied to baseline RTT to detect a warning-level RTT spike. */
-const DEFAULT_STATS_RTT_SPIKE_MULTIPLIER = 3;
-/** Packet loss fraction (0-1) above which a warning is emitted. */
-const DEFAULT_STATS_PACKET_LOSS_THRESHOLD = .05;
-/** Multiplier applied to baseline jitter to detect a jitter spike. */
-const DEFAULT_STATS_JITTER_SPIKE_MULTIPLIER = 4;
-/** Number of seconds of metrics history to retain. */
-const DEFAULT_STATS_HISTORY_SIZE = 30;
-/** Maximum keyframe requests allowed within a single burst window. */
-const DEFAULT_KEYFRAME_MAX_BURST$1 = 3;
-/** Duration of the keyframe burst window in milliseconds. */
-const DEFAULT_KEYFRAME_BURST_WINDOW_MS$1 = 3e3;
-/** Cooldown period in ms after burst limit is reached before allowing more keyframes. */
-const DEFAULT_KEYFRAME_COOLDOWN_MS$1 = 1e4;
-/** Minimum time between re-INVITE attempts in milliseconds. */
-const DEFAULT_REINVITE_DEBOUNCE_TIME_MS = 1e4;
-/** Maximum number of re-INVITE attempts per call. */
-const DEFAULT_REINVITE_MAX_ATTEMPTS = 3;
-/** Timeout for a single re-INVITE attempt in milliseconds. */
-const DEFAULT_REINVITE_TIMEOUT_MS = 5e3;
-/** Debounce window in ms to collapse multiple detection signals into one trigger. */
-const DEFAULT_RECOVERY_DEBOUNCE_TIME_MS = 2e3;
-/** Cooldown period in ms between recovery attempts. */
-const DEFAULT_RECOVERY_COOLDOWN_MS = 1e4;
-/** Grace period in ms before treating ICE 'disconnected' as a failure. */
-const DEFAULT_ICE_DISCONNECTED_GRACE_PERIOD_MS = 3e3;
-/** Timeout for a single ICE restart attempt in milliseconds. */
-const DEFAULT_ICE_RESTART_TIMEOUT_MS$1 = 5e3;
-/** Maximum recovery attempts before emitting 'max_attempts_reached'. */
-const DEFAULT_MAX_RECOVERY_ATTEMPTS = 3;
-/** Upper bound in ms for waiting on iceGatheringState === 'complete' after an ICE restart. */
-const ICE_GATHERING_COMPLETE_TIMEOUT_MS = 1e4;
-/** Upper bound in ms for waiting on RTCPeerConnection.connectionState === 'connected' after a recovery ICE restart. */
-const PEER_CONNECTION_RECOVERY_WAIT_MS = 5e3;
-/** Polling interval in ms while waiting for RTCPeerConnection.connectionState to transition. */
-const PEER_CONNECTION_RECOVERY_POLL_MS = 100;
-/** Polling interval for LocalAudioPipeline.level$ (ms). ~30fps is smooth for meters. */
-const AUDIO_LEVEL_POLL_INTERVAL_MS = 33;
-/** RMS level threshold (0..1) above which the local participant is considered speaking. */
-const VAD_THRESHOLD = .03;
-/** Hold window in ms below the threshold before speaking$ flips back to false. */
-const VAD_HOLD_MS = 250;
-/** Whether to persist device selections to storage by default. */
-const DEFAULT_PERSIST_DEVICE_SELECTION = true;
-/** Whether to auto-apply device changes to active calls by default. */
-const DEFAULT_SYNC_DEVICES_TO_ACTIVE_CALLS = true;
-/** Storage keys for persisted device selections. */
-const DEVICE_STORAGE_KEY_AUDIO_INPUT = "sw:device:audioinput";
-const DEVICE_STORAGE_KEY_AUDIO_OUTPUT = "sw:device:audiooutput";
-const DEVICE_STORAGE_KEY_VIDEO_INPUT = "sw:device:videoinput";
-/** Whether to auto-mute video when the tab becomes hidden. */
-const DEFAULT_AUTO_MUTE_VIDEO_ON_HIDDEN = false;
-/** Whether to re-enumerate devices when the page becomes visible. */
-const DEFAULT_REFRESH_DEVICES_ON_VISIBLE = true;
-/** Whether to check peer connection health when the page becomes visible. */
-const DEFAULT_CHECK_CONNECTION_ON_VISIBLE = true;
-/** Whether automatic video degradation on low bandwidth is enabled. */
-const DEFAULT_ENABLE_AUTO_DEGRADATION = true;
-/** Bitrate in kbps below which video is automatically disabled. */
-const DEFAULT_DEGRADATION_BITRATE_THRESHOLD_KBPS = 150;
-/** Bitrate in kbps above which video is automatically re-enabled (hysteresis). */
-const DEFAULT_DEGRADATION_RECOVERY_THRESHOLD_KBPS = 300;
-/** Whether relay-only escalation is enabled as a last-resort recovery tier. */
-const DEFAULT_ENABLE_RELAY_FALLBACK = true;
-/** Whether to listen for browser online/offline/connection events. */
-const DEFAULT_ENABLE_NETWORK_CHANGE_DETECTION = true;
-/** Whether to intercept server-sent media-timeout hangups and attempt recovery. */
-const DEFAULT_ENABLE_SERVER_HANGUP_INTERCEPTION = true;
-/** Default video track constraints applied when video is enabled without explicit constraints. */
-const DEFAULT_VIDEO_CONSTRAINTS = {
-	width: { ideal: 1280 },
-	height: { ideal: 720 },
-	aspectRatio: 16 / 9
-};
-/** Whether stereo Opus is enabled by default. */
-const DEFAULT_STEREO_AUDIO = false;
-/** Max average bitrate for stereo Opus in bits per second. */
-const DEFAULT_STEREO_MAX_AVERAGE_BITRATE = 51e4;
-
 //#endregion
 //#region src/utils/time.ts
 function fromSecToMs(seconds) {
@@ -13367,7 +13400,9 @@ const DEFAULT_MEMBER_CAPABILITIES = {
 	position: false,
 	meta: false,
 	remove: false,
-	audioFlags: false
+	audioFlags: false,
+	denoise: false,
+	lowbitrate: false
 };
 /**
 * Default call capabilities with no permissions
@@ -13440,7 +13475,9 @@ function computeMemberCapabilities(flags, memberType) {
 		position: hasBooleanCapability(flags, memberType, null, "position"),
 		meta: hasBooleanCapability(flags, memberType, null, "meta"),
 		remove: hasBooleanCapability(flags, memberType, null, "remove"),
-		audioFlags: hasBooleanCapability(flags, memberType, null, "audioflags")
+		audioFlags: hasBooleanCapability(flags, memberType, null, "audioflags"),
+		denoise: hasBooleanCapability(flags, memberType, null, "denoise"),
+		lowbitrate: hasBooleanCapability(flags, memberType, null, "lowbitrate")
 	};
 }
 /**
@@ -13823,6 +13860,10 @@ var Participant = class extends Destroyable {
 	get nodeId() {
 		return this._state$.value.node_id;
 	}
+	/** Call ID for this participant's leg, or `undefined` if not available. */
+	get callId() {
+		return this._state$.value.call_id;
+	}
 	/** @internal */
 	get value() {
 		return this._state$.value;
@@ -13884,8 +13925,9 @@ var Participant = class extends Destroyable {
 			noise_suppression: !this.noiseSuppression
 		});
 	}
+	/** Toggles low-bitrate mode for this participant's media. */
 	async toggleLowbitrate() {
-		throw new UnimplementedError();
+		await this.executeMethod(this.id, "call.lowbitrate.set", { lowbitrate: !this.lowbitrate });
 	}
 	/**
 	* Adjusts the **conference-only** microphone energy gate / sensitivity level
@@ -13932,10 +13974,27 @@ var Participant = class extends Destroyable {
 	}
 	/**
 	* Sets the participant's position in the video layout.
+	*
+	* Requires the `member.position` capability. The gateway keys positions by the
+	* **target member's own** `call_id`/`node_id` (see issue #19400 and the legacy
+	* `setPositions` implementation), so this sends the participant's own call
+	* context — matching {@link Participant.remove}. A resolved promise does not
+	* guarantee a visible change: the backend silently returns `200` (no-op) for
+	* non-conference targets.
+	*
 	* @param value - The {@link VideoPosition} to assign (e.g. `'auto'`, `'reserved-0'`).
 	*/
 	async setPosition(value) {
-		await this.executeMethod(this.id, "call.member.position.set", { position: value });
+		const state = this._state$.value;
+		const target = {
+			member_id: this.id,
+			call_id: state.call_id ?? "",
+			node_id: state.node_id ?? ""
+		};
+		await this.executeMethod(target, "call.member.position.set", { targets: [{
+			target,
+			position: value
+		}] });
 	}
 	/** Removes this participant from the call. */
 	async remove() {
@@ -14025,12 +14084,21 @@ var SelfParticipant = class extends Participant {
 			noise_suppression: true
 		});
 	}
-	/** Starts sharing the local screen. */
+	/**
+	* Starts sharing the local screen.
+	*
+	* The call is unaffected when acquisition fails.
+	*
+	* @throws The raw `getDisplayMedia` error. A dismissed picker or a
+	* permission denial rejects with a `NotAllowedError` `DOMException` —
+	* inspect `error.name` to tell benign cancels apart from real failures.
+	*/
 	async startScreenShare() {
 		try {
 			await this.vertoManager.addScreenMedia();
 		} catch (error) {
 			logger$24.error("[Participant.startScreenShare] Screen share error:", error);
+			throw error;
 		}
 	}
 	/** Observable of the current screen share status. */
@@ -14045,12 +14113,20 @@ var SelfParticipant = class extends Participant {
 	async stopScreenShare() {
 		return this.vertoManager.removeScreenMedia();
 	}
-	/** Adds an additional media input device to the call. */
+	/**
+	* Adds an additional media input device to the call.
+	*
+	* The call is unaffected when acquisition fails.
+	*
+	* @throws The raw `getUserMedia` error (e.g. `NotAllowedError` on
+	* permission denial) — inspect `error.name` to decide how to react.
+	*/
 	async addAdditionalDevice(options) {
 		try {
 			await this.vertoManager.addInputDevice(options);
 		} catch (error) {
-			logger$24.error("[Participant.startScreenShare] Screen share error:", error);
+			logger$24.error("[Participant.addAdditionalDevice] Additional device error:", error);
+			throw error;
 		}
 	}
 	/** Removes an additional media input device by ID. */
@@ -15682,7 +15758,7 @@ var RTCPeerConnectionController = class extends Destroyable {
 				logger$17.debug(`[RTCPeerConnectionController] ${kind} input device selected:`, deviceInfo?.label);
 			} catch (error) {
 				logger$17.error(`[RTCPeerConnectionController] Failed to select ${kind} input device:`, error);
-				this._errors$.next(toError(error));
+				this._errors$.next(new MediaTrackError("updateSelectedInputDevice", kind, error));
 				throw error;
 			}
 		};
@@ -15904,7 +15980,7 @@ var RTCPeerConnectionController = class extends Destroyable {
 			case "main":
 			default: return {
 				...options,
-				offerToReceiveAudio: true,
+				offerToReceiveAudio: this.options.receiveAudio ?? true,
 				offerToReceiveVideo: this.options.receiveVideo ?? Boolean(this.inputVideoDeviceConstraints)
 			};
 		}
@@ -16058,13 +16134,14 @@ var RTCPeerConnectionController = class extends Destroyable {
 	*/
 	async acceptInbound(mediaOverrides) {
 		if (mediaOverrides) {
-			const { audio, video, receiveAudio, receiveVideo } = mediaOverrides;
+			const { audio, video, receiveAudio, receiveVideo, fallbackToReceiveOnly } = mediaOverrides;
 			this.options = {
 				...this.options,
 				...audio !== void 0 ? { audio } : {},
 				...video !== void 0 ? { video } : {},
 				...receiveAudio !== void 0 ? { receiveAudio } : {},
-				...receiveVideo !== void 0 ? { receiveVideo } : {}
+				...receiveVideo !== void 0 ? { receiveVideo } : {},
+				...fallbackToReceiveOnly !== void 0 ? { fallbackToReceiveOnly } : {}
 			};
 			this.transceiverController?.updateOptions({
 				receiveAudio: this.receiveAudio,
@@ -16221,7 +16298,19 @@ var RTCPeerConnectionController = class extends Destroyable {
 	}
 	async setupLocalTracks() {
 		logger$17.debug("[RTCPeerConnectionController] Setting up local tracks/transceivers.");
-		const localStream = this.localStream ?? await this.localStreamController.buildLocalStream();
+		if (this.hasNoLocalMediaToSend()) {
+			if (!this.receiveAudio && !this.receiveVideo) throw new InvalidParams("Call requests no media: enable audio/video or receiveAudio/receiveVideo");
+			logger$17.debug("[RTCPeerConnectionController] No local media requested; negotiating receive-only.");
+			this.setupReceiveOnlyTransceivers();
+			return;
+		}
+		let localStream;
+		try {
+			localStream = this.localStream ?? await this.localStreamController.buildLocalStream();
+		} catch (error) {
+			this.handleLocalMediaFailure(error);
+			return;
+		}
 		if (this.transceiverController?.useAddStream ?? false) {
 			logger$17.warn("[RTCPeerConnectionController] Using deprecated addStream API to add local stream.");
 			this.peerConnection?.addStream(localStream);
@@ -16248,6 +16337,48 @@ var RTCPeerConnectionController = class extends Destroyable {
 			}
 		}
 	}
+	/** True for a main connection with no local media to send. */
+	hasNoLocalMediaToSend() {
+		const hasInputStreams = Boolean(this.options.inputAudioStream ?? this.options.inputVideoStream);
+		return this.propose === "main" && !this.localStream && !hasInputStreams && !this.inputAudioDeviceConstraints && !this.inputVideoDeviceConstraints;
+	}
+	/** The media kinds this connection wants to send: 'audiovideo' | 'video' | 'audio'. */
+	get requestedMediaKinds() {
+		const wantsAudio = Boolean(this.inputAudioDeviceConstraints);
+		const wantsVideo = Boolean(this.inputVideoDeviceConstraints);
+		if (wantsAudio && wantsVideo) return "audiovideo";
+		return wantsVideo ? "video" : "audio";
+	}
+	/**
+	* Handle a local media acquisition failure with a typed, semantically
+	* accurate MediaAccessError created at the acquisition site:
+	* - Auxiliary connections (screenshare / additional-device) throw a
+	*   non-fatal error — VertoManager surfaces it and the call is unaffected.
+	* - The main connection degrades to receive-only when allowed (default),
+	*   otherwise fails with a fatal error.
+	*/
+	handleLocalMediaFailure(error) {
+		if (this.propose === "screenshare") throw new MediaAccessError("startScreenShare", "screen", error, false);
+		if (this.propose === "additional-device") throw new MediaAccessError("addInputDevice", this.requestedMediaKinds, error, false);
+		const canReceive = this.receiveAudio || this.receiveVideo;
+		if (!((this.options.fallbackToReceiveOnly ?? true) && canReceive)) throw new MediaAccessError("acquireLocalMedia", this.requestedMediaKinds, error, true);
+		logger$17.warn("[RTCPeerConnectionController] Local media unavailable; continuing receive-only:", error);
+		this._errors$.next(new MediaAccessError("acquireLocalMedia", this.requestedMediaKinds, error, false));
+		this.setupReceiveOnlyTransceivers();
+	}
+	/**
+	* Negotiate receive-only m-lines when there are no local tracks to send.
+	* Only offer-type connections add transceivers — answer-type connections
+	* reuse the transceivers created from the remote offer.
+	*/
+	setupReceiveOnlyTransceivers() {
+		if (this.type !== "offer") return;
+		if (this.transceiverController?.useAddTransceivers ?? false) {
+			this.peerConnection?.addTransceiver("audio", { direction: this.receiveAudio ? "recvonly" : "inactive" });
+			this.peerConnection?.addTransceiver("video", { direction: this.receiveVideo ? "recvonly" : "inactive" });
+		}
+		if (!this.isNegotiating) this.negotiationNeeded$.next();
+	}
 	async getUserMedia(constraints) {
 		return (this.options.webRTCApiProvider?.mediaDevices ?? navigator.mediaDevices).getUserMedia(constraints);
 	}
@@ -16284,7 +16415,7 @@ var RTCPeerConnectionController = class extends Destroyable {
 			stream = await this.getUserMedia({ audio: constraints });
 		} catch (error) {
 			logger$17.error("[RTCPeerConnectionController] Failed to re-acquire mic for pipeline restore:", error);
-			this._errors$.next(toError(error));
+			this._errors$.next(new MediaTrackError("restoreAudioPipelineInput", "audio", error));
 			return;
 		}
 		const newTrack = stream.getAudioTracks().at(0);
@@ -16346,7 +16477,7 @@ var RTCPeerConnectionController = class extends Destroyable {
 			logger$17.debug(`[RTCPeerConnectionController] ${track.kind} track added:`, track.id);
 		} catch (error) {
 			logger$17.error(`[RTCPeerConnectionController] Failed to add ${track.kind} track:`, error);
-			this._errors$.next(toError(error));
+			this._errors$.next(new MediaTrackError("addLocalTrack", track.kind, error));
 			throw error;
 		}
 	}
@@ -16371,7 +16502,7 @@ var RTCPeerConnectionController = class extends Destroyable {
 			logger$17.debug(`[RTCPeerConnectionController] ${sender.track?.kind} track removed:`, trackId);
 		} catch (error) {
 			logger$17.error(`[RTCPeerConnectionController] Failed to remove ${sender.track?.kind} track:`, error);
-			this._errors$.next(toError(error));
+			this._errors$.next(new MediaTrackError("removeLocalTrack", sender.track?.kind ?? "unknown", error));
 			throw error;
 		}
 	}
@@ -16481,7 +16612,11 @@ function isVertoInviteMessage(value) {
 	const msg = value;
 	return msg.method === "verto.invite" && isObject(msg.params) && hasProperty(msg.params, "sdp") && hasProperty(msg.params, "callID");
 }
-function isVertoByeMessage(value) {
+/**
+* Guards inbound `verto.bye` frames (server → client). Outbound bye messages are
+* built locally and never type-guarded, so only the inbound shape is asserted.
+*/
+function isVertoByeInboundMessage(value) {
 	if (!isVertoMethodMessage(value)) return false;
 	return value.method === "verto.bye";
 }
@@ -16501,6 +16636,14 @@ function isVertoMediaParamsInnerParams(value) {
 function isVertoPingInnerParams(value) {
 	return isObject(value) && hasProperty(value, "jsonrpc") && value.jsonrpc === "2.0" && hasProperty(value, "method") && value.method === "verto.ping";
 }
+/**
+* Guards the params of an inbound `verto.bye` frame (top-level `callID`). Use
+* this to recognise the bye payload extracted by `vertoBye$`, e.g. when racing
+* it against a boolean answer/reject signal.
+*/
+function isVertoByeInboundParamsGuard(value) {
+	return isObject(value) && hasProperty(value, "callID") && typeof value.callID === "string";
+}
 
 //#endregion
 //#region src/managers/VertoManager.ts
@@ -16839,7 +16982,7 @@ var WebRTCVertoManager = class extends VertoManager {
 		return this.cachedObservable("vertoMediaParams$", () => this.webRtcCallSession.webrtcMessages$.pipe(filterAs(isVertoMediaParamsInnerParams, "params"), (0, import_cjs$15.takeUntil)(this.destroyed$)));
 	}
 	get vertoBye$() {
-		return this.cachedObservable("vertoBye$", () => this.webRtcCallSession.webrtcMessages$.pipe(filterAs(isVertoByeMessage, "params"), (0, import_cjs$15.takeUntil)(this.destroyed$)));
+		return this.cachedObservable("vertoBye$", () => this.webRtcCallSession.webrtcMessages$.pipe(filterAs(isVertoByeInboundMessage, "params"), (0, import_cjs$15.takeUntil)(this.destroyed$)));
 	}
 	get vertoAttach$() {
 		return this.cachedObservable("vertoAttach$", () => this.webRtcCallSession.webrtcMessages$.pipe(filterAs(isVertoAttachMessage, "params"), (0, import_cjs$15.takeUntil)(this.destroyed$)));
@@ -16968,6 +17111,7 @@ var WebRTCVertoManager = class extends VertoManager {
 			inputVideoStream: options.inputVideoStream,
 			receiveAudio: options.receiveAudio,
 			receiveVideo: options.receiveVideo,
+			fallbackToReceiveOnly: options.fallbackToReceiveOnly,
 			webRTCApiProvider: this.webRTCApiProvider,
 			preferredVideoCodecs: options.preferredVideoCodecs,
 			preferredAudioCodecs: options.preferredAudioCodecs,
@@ -16992,7 +17136,7 @@ var WebRTCVertoManager = class extends VertoManager {
 			logger$16.debug("[WebRTCManager] Inbound answer handler aborted (destroyed).");
 			return;
 		}
-		if (isVertoByeMessage(vertoByeOrAccepted)) {
+		if (isVertoByeInboundParamsGuard(vertoByeOrAccepted)) {
 			logger$16.info("[WebRTCManager] Inbound call ended by remote before answer.");
 			this.callSession?.destroy();
 		} else if (!vertoByeOrAccepted) {
@@ -17094,7 +17238,7 @@ var WebRTCVertoManager = class extends VertoManager {
 			logger$16.debug("[WebRTCManager] Destroyed while waiting for call acceptance");
 			return;
 		}
-		if (isVertoByeMessage(vertoByeOrAccepted)) {
+		if (isVertoByeInboundParamsGuard(vertoByeOrAccepted)) {
 			logger$16.info("[WebRTCManager] Call ended before answer was sent.");
 			this.callSession?.destroy();
 		} else if (!vertoByeOrAccepted) {
@@ -17194,9 +17338,11 @@ var WebRTCVertoManager = class extends VertoManager {
 		await this.initAdditionalPeerConnection("screenshare", options);
 	}
 	async initAdditionalPeerConnection(propose, options) {
+		const isScreenShare = propose === "screenshare";
+		let firstPeerConnectionError;
 		let rtcPeerConnController = null;
 		try {
-			this._screenShareStatus$.next("starting");
+			if (isScreenShare) this._screenShareStatus$.next("starting");
 			rtcPeerConnController = new RTCPeerConnectionController({
 				...options,
 				...this.RTCPeerConnectionConfig,
@@ -17204,21 +17350,27 @@ var WebRTCVertoManager = class extends VertoManager {
 				webRTCApiProvider: this.webRTCApiProvider
 			}, void 0, this.deviceController);
 			this.setupLocalDescriptionHandler(rtcPeerConnController);
-			if (propose === "screenshare") this._screenShareId = rtcPeerConnController.id;
+			if (isScreenShare) this._screenShareId = rtcPeerConnController.id;
 			this._rtcPeerConnectionsMap.set(rtcPeerConnController.id, rtcPeerConnController);
 			this._rtcPeerConnections$.next(Array.from(this._rtcPeerConnectionsMap.values()));
 			this.subscribeTo(rtcPeerConnController.errors$, (error) => {
-				this.onError?.(error);
+				firstPeerConnectionError ??= error;
+				this.onError?.(error, { fatal: false });
 			});
 			await (0, import_cjs$15.firstValueFrom)(rtcPeerConnController.connectionState$.pipe((0, import_cjs$15.filter)((state) => state === "connected"), (0, import_cjs$15.take)(1), (0, import_cjs$15.timeout)(this._screenShareTimeoutMs), (0, import_cjs$15.takeUntil)(this.destroyed$)));
-			this._screenShareStatus$.next("started");
-			logger$16.info("[WebRTCManager] Screen share started successfully.");
+			if (isScreenShare) this._screenShareStatus$.next("started");
+			logger$16.info(`[WebRTCManager] Additional peer connection connected (${propose}).`);
 			return rtcPeerConnController.id;
 		} catch (error) {
 			logger$16.warn("[WebRTCManager] Error initializing additional peer connection:", error);
-			this.onError?.(error instanceof Error ? error : new Error(String(error), { cause: error }));
 			if (rtcPeerConnController) rtcPeerConnController.destroy();
-			this._screenShareStatus$.next("none");
+			if (isScreenShare) this._screenShareStatus$.next("none");
+			if (firstPeerConnectionError) throw firstPeerConnectionError instanceof MediaAccessError && firstPeerConnectionError.originalError instanceof Error ? firstPeerConnectionError.originalError : firstPeerConnectionError;
+			if (error instanceof import_cjs$15.EmptyError) {
+				logger$16.debug("[WebRTCManager] Additional peer connection aborted before connecting.");
+				return;
+			}
+			throw error instanceof Error ? error : new Error(String(error), { cause: error });
 		}
 	}
 	async removeInputDevices(id) {
@@ -18147,6 +18299,12 @@ function mosToQualityLevel(mos) {
 var import_cjs$11 = require_cjs();
 const logger$12 = getLogger();
 /**
+* Verto method for setting member layout positions. Its gateway DTO requires a
+* `targets` array whose entries are `{ target, position }` (NOT bare targets),
+* so {@link WebRTCCall.buildMethodParams} special-cases it. See issue #19400.
+*/
+const POSITION_SET_METHOD = "call.member.position.set";
+/**
 * Ratio between the critical and warning RTT spike multipliers.
 * Warning threshold = baseline * warningMultiplier (default 3x)
 * Critical threshold = baseline * warningMultiplier * RTT_CRITICAL_TO_WARNING_RATIO
@@ -18353,7 +18511,7 @@ var WebRTCCall = class extends Destroyable {
 	* @throws {JSONRPCError} If the RPC call returns an error.
 	*/
 	async executeMethod(target, method, args) {
-		const params = this.buildMethodParams(target, args);
+		const params = this.buildMethodParams(target, args, method);
 		const request = buildRPCRequest({
 			method,
 			params
@@ -18367,12 +18525,16 @@ var WebRTCCall = class extends Destroyable {
 			throw error;
 		}
 	}
-	buildMethodParams(target, args) {
+	buildMethodParams(target, args, method) {
 		const self = {
 			node_id: this.nodeId ?? "",
 			call_id: this.id,
 			member_id: this.vertoManager.selfId ?? ""
 		};
+		if (method === POSITION_SET_METHOD) return {
+			...args,
+			self
+		};
 		if (typeof target === "object") return {
 			...args,
 			self,
@@ -18938,10 +19100,21 @@ var WebRTCCall = class extends Destroyable {
 		return this.deferEmission(this._answered$.asObservable());
 	}
 	/**
-	* Sets the call layout and participant positions.
+	* Sets the call layout and, optionally, individual participant positions.
+	*
+	* The gateway `call.layout.set` DTO has **no** `positions` member, so when
+	* `positions` is provided this method issues a `call.member.position.set`
+	* request per member (via {@link Participant.setPosition}, which keys each
+	* position by that member's own call context) alongside `call.layout.set`
+	* (issue #19400, Flag #6).
+	*
+	* **These operations are NOT atomic.** The layout is applied first, then each
+	* member position sequentially, so members may briefly flash into their
+	* default slots before being moved to the requested positions.
 	*
 	* @param layout - Layout name (must be one of {@link layouts}).
-	* @param positions - Map of member IDs to {@link VideoPosition} values.
+	* @param positions - Optional map of member IDs to {@link VideoPosition} values.
+	*   When omitted or empty, only the layout is changed.
 	* @throws {InvalidParams} If the layout is not in the available {@link layouts}.
 	*
 	* @example
@@ -18954,10 +19127,17 @@ var WebRTCCall = class extends Destroyable {
 	async setLayout(layout, positions) {
 		if (!this.layouts.includes(layout)) throw new InvalidParams(`Layout ${layout} is not available in the current call layouts: ${this.layouts.join(", ")}`);
 		const selfId = await (0, import_cjs$11.firstValueFrom)(this.selfId$.pipe((0, import_cjs$11.filter)((id) => id !== null)));
-		await this.executeMethod(selfId, "call.layout.set", {
-			layout,
-			positions
-		});
+		await this.executeMethod(selfId, "call.layout.set", { layout });
+		const positionEntries = Object.entries(positions ?? {});
+		if (positionEntries.length === 0) return;
+		for (const [memberId, position] of positionEntries) {
+			const participant = this.participants.find((p) => p.id === memberId);
+			if (!participant) {
+				logger$12.warn(`[Call] setLayout: member ${memberId} not found in participants; skipping position ${position}`);
+				continue;
+			}
+			await participant.setPosition(position);
+		}
 	}
 	/**
 	* Transfers the call to another destination.
@@ -19131,6 +19311,7 @@ function inferCallErrorKind(error) {
 	if (error instanceof RPCTimeoutError) return "timeout";
 	if (error instanceof JSONRPCError) return "signaling";
 	if (error instanceof MediaTrackError) return "media";
+	if (error instanceof MediaAccessError) return "media";
 	if (error instanceof WebSocketConnectionError || error instanceof TransportConnectionError) return "network";
 	return "internal";
 }
@@ -19147,6 +19328,7 @@ const RECOVERABLE_RPC_CODES = new Set([
 function isFatalError(error) {
 	if (error instanceof VertoPongError) return false;
 	if (error instanceof MediaTrackError) return false;
+	if (error instanceof MediaAccessError) return error.fatal;
 	if (error instanceof RPCTimeoutError) return false;
 	if (error instanceof JSONRPCError && RECOVERABLE_RPC_CODES.has(error.code)) return false;
 	return true;
@@ -19172,10 +19354,10 @@ var CallFactory = class {
 				return {
 					vertoManager: new WebRTCVertoManager(callInstance, this.attachManager, this.deviceController, this.webRTCApiProvider, {
 						nodeId: options.nodeId,
-						onError: (error) => {
+						onError: (error, options$1) => {
 							const callError = {
 								kind: inferCallErrorKind(error),
-								fatal: isFatalError(error),
+								fatal: options$1?.fatal ?? isFatalError(error),
 								error,
 								callId: callInstance.id
 							};
@@ -19638,6 +19820,17 @@ var PendingRPC = class PendingRPC {
 //#region src/managers/ClientSessionManager.ts
 var import_cjs$7 = require_cjs();
 const logger$9 = getLogger();
+/**
+* Decide whether an error emitted on `call.errors$` during dial should
+* abort the dial. A non-fatal MediaAccessError means the call degraded to
+* receive-only and still connects — everything else rejects `dial()` with
+* the real cause.
+*
+* Pure function — exported for unit testing.
+*/
+function shouldAbortDial(callError) {
+	return callError.fatal || !(callError.error instanceof MediaAccessError);
+}
 const getAddressSearchURI = (options) => {
 	const to = options.to?.split("?")[0];
 	const from$9 = options.from?.startsWith("subscriber://") ? options.from.replace("subscriber://", "") : options.from;
@@ -19876,11 +20069,41 @@ var ClientSessionManager = class extends Destroyable {
 			}
 		} else this._errors$.next(error);
 	}
+	/**
+	* Clear the resume state (authorization_state + protocol) only.
+	*
+	* This is the stale-auth-state recovery helper used by handleAuthError:
+	* the server rejected a reconnect, so the resume state is discarded and a
+	* fresh connect follows. Attach records are deliberately preserved — the
+	* session lives on through the reconnect and reattachCalls() needs the
+	* stored call references afterwards. Do NOT add detachAll() here.
+	*
+	* For public teardown (disconnect/destroy), use {@link teardownSessionState}
+	* instead, which clears the attach records as well.
+	*/
 	async cleanupStoredConnectionParams() {
 		await this.transport.setProtocol(void 0);
 		await this.updateAuthorizationStateInStorage(void 0);
 		this._authorization$.next(void 0);
 	}
+	/**
+	* Public-teardown helper for disconnect()/destroy(). Clears the resume
+	* state (authorization_state + protocol) AND the attach records as one
+	* atomic unit.
+	*
+	* The two stores are coupled: the backend only honors attach records
+	* within the session identified by the resume state, so ending the
+	* session must clear both. Clearing one without the other strands records
+	* no future session can honor (disconnect) or revives a session the
+	* developer explicitly ended (destroy).
+	*
+	* Distinct from {@link cleanupStoredConnectionParams}, which keeps the
+	* attach records for the stale-auth-state recovery path.
+	*/
+	async teardownSessionState() {
+		await this.cleanupStoredConnectionParams();
+		await this.attachManager.detachAll();
+	}
 	async updateAuthState(authorization_state) {
 		try {
 			await this.storage.setItem(this.authorizationStateKey, authorization_state);
@@ -19975,7 +20198,7 @@ var ClientSessionManager = class extends Destroyable {
 	async disconnect() {
 		this.transport.disconnect();
 		this._authState$.next({ kind: "unauthenticated" });
-		await this.cleanupStoredConnectionParams();
+		await this.teardownSessionState();
 	}
 	async createInboundCall(invite) {
 		const callSession = await this.createCall({
@@ -20034,7 +20257,7 @@ var ClientSessionManager = class extends Destroyable {
 				to: destinationURI,
 				...options
 			});
-			await (0, import_cjs$7.firstValueFrom)((0, import_cjs$7.race)(callSession.selfId$.pipe((0, import_cjs$7.filter)((id) => Boolean(id)), (0, import_cjs$7.take)(1), (0, import_cjs$7.timeout)(this.callCreateTimeout)), callSession.errors$.pipe((0, import_cjs$7.take)(1), (0, import_cjs$7.switchMap)((callError) => (0, import_cjs$7.throwError)(() => callError.error)))));
+			await (0, import_cjs$7.firstValueFrom)((0, import_cjs$7.race)(callSession.selfId$.pipe((0, import_cjs$7.filter)((id) => Boolean(id)), (0, import_cjs$7.take)(1), (0, import_cjs$7.timeout)(this.callCreateTimeout)), callSession.errors$.pipe((0, import_cjs$7.filter)(shouldAbortDial), (0, import_cjs$7.take)(1), (0, import_cjs$7.switchMap)((callError) => (0, import_cjs$7.throwError)(() => callError.error)))));
 			this._calls$.next({
 				[`${callSession.id}`]: callSession,
 				...this._calls$.value
@@ -21142,6 +21365,10 @@ var TransportManager = class extends Destroyable {
 				if (!isSignalwireRequest(message)) return true;
 				const eventChannel = message.params.event_channel;
 				if (!eventChannel) return true;
+				if (message.params.event_type.startsWith("conversation.")) {
+					logger$2.debug(`[Transport] Received conversation event: ${message.params.event_type} (event_channel: ${eventChannel})`);
+					return true;
+				}
 				const currentProtocol = this._currentProtocol;
 				if (!currentProtocol) return true;
 				if (!eventChannel.includes(currentProtocol)) {
@@ -21742,7 +21969,7 @@ var SignalWire = class extends Destroyable {
 			logger$1.warn("[SignalWire] Failed to initialize VisibilityController:", error);
 		}
 		try {
-			this._diagnosticsCollector = new DiagnosticsCollector({ sdkVersion: "4.0.0-rc.0" });
+			this._diagnosticsCollector = new DiagnosticsCollector({ sdkVersion: "4.0.0-rc.2" });
 		} catch (error) {
 			logger$1.warn("[SignalWire] Failed to initialize DiagnosticsCollector:", error);
 		}
@@ -21750,6 +21977,14 @@ var SignalWire = class extends Destroyable {
 	/**
 	* Disconnects the WebSocket and tears down the current session.
 	*
+	* Ends the session identified by the protocol and clears its persisted
+	* resume state (`authorization_state` + protocol) and attach records
+	* together — a later {@link connect} with the same credentials starts a
+	* fresh session and cannot reattach to the ended session's calls.
+	* Credentials and device preferences are preserved. To temporarily stop
+	* receiving inbound calls while keeping the session alive, use
+	* `unregister()` instead.
+	*
 	* The client can be reconnected by calling {@link connect} again,
 	* which creates a fresh transport and session.
 	*/
@@ -22141,12 +22376,20 @@ var SignalWire = class extends Destroyable {
 		prefs.preferredVideoInput = null;
 		await this._deviceController.clearDeviceState();
 	}
-	/** Destroys the client, clearing timers and releasing all resources. */
+	/**
+	* Destroys the client, clearing timers and releasing all resources.
+	*
+	* Intentionally destroying the client ends its session: the resume state
+	* (`authorization_state` + protocol) and the attach records are both
+	* cleared. Credentials and device preferences are preserved — use
+	* {@link resetToDefaults} for a full wipe. To temporarily stop receiving
+	* inbound calls while keeping the session alive, use `unregister()`.
+	*/
 	destroy() {
 		this._refreshCoordinator?.destroy();
 		this._refreshCoordinator = void 0;
 		this._dpopManager?.destroy();
-		if (this._attachManager) this._attachManager.detachAll();
+		this._clientSession.teardownSessionState();
 		this._transport.destroy();
 		this._clientSession.destroy();
 		try {
@@ -22261,7 +22504,7 @@ var StaticCredentialProvider = class {
 /**
 * Library version from package.json, injected at build time.
 */
-const version = "4.0.0-rc.0";
+const version = "4.0.0-rc.2";
 /**
 * Flag indicating the library has been loaded and is ready to use.
 * For UMD builds: `window.SignalWire.ready`
@@ -22283,7 +22526,7 @@ const ready = true;
 */
 const emitReadyEvent = () => {
 	if (typeof window !== "undefined") {
-		const event = new CustomEvent("signalwire:js:ready", { detail: { version: "4.0.0-rc.0" } });
+		const event = new CustomEvent("signalwire:js:ready", { detail: { version: "4.0.0-rc.2" } });
 		window.dispatchEvent(event);
 	}
 };
@@ -22307,5 +22550,5 @@ emitReadyEvent();
 if (typeof process === "undefined") globalThis.process = { env: { NODE_ENV: "production" } };
 
 //#endregion
-export { Address, CallCreateError, ClientPreferences, CollectionFetchError, DPoPInitError, DeviceTokenError, EmbedTokenCredentialProvider, InvalidCredentialsError, MediaTrackError, MessageParseError, OverconstrainedFallbackError, Participant, PreflightError, RecoveryError, SelfCapabilities, SelfParticipant, SignalWire, StaticCredentialProvider, TokenRefreshError, UnexpectedError, User, VertoPongError, WebRTCCall, embeddableCall, getLogger, isSelfParticipant, ready, setDebugOptions, setLogLevel, setLogger, version };
+export { Address, CallCreateError, ClientPreferences, CollectionFetchError, DPoPInitError, DeviceTokenError, EmbedTokenCredentialProvider, InvalidCredentialsError, MediaAccessError, MediaTrackError, MessageParseError, OverconstrainedFallbackError, Participant, PreflightError, RecoveryError, SelfCapabilities, SelfParticipant, SignalWire, StaticCredentialProvider, TokenRefreshError, UnexpectedError, User, VertoPongError, WebRTCCall, embeddableCall, getLogger, isSelfParticipant, ready, setDebugOptions, setLogLevel, setLogger, version };
 //# sourceMappingURL=browser.mjs.map