@ale-rainbow/web-sdk 5.1.0

package/lib/libs/strophe/strophe.jingle.session.d.ts

@@ -0,0 +1,624 @@
+import type { ServiceEventsToUnion } from "../../services/event-mixin";
+import type { JingleSessionMetricsEventData, JingleStreamUpdatedEventData, WebrtcStatisticsL2SsrcState } from "./strophe.jingle.session.stats";
+/**
+ * @internal
+ * Call service events send to other services
+ * Can listen to it via `CallService.subscribe()` API.
+ */
+export declare enum StropheJingleEvents {
+    /**
+     * Fired after each WebRTC stats collection cycle (L1 every 5 s).
+     *
+     * The event is emitted on `jingle.rxSubject` by {@link WebrtcSessionStatsHandler} once all
+     * calculations (MOS, RTT, packet-loss deltas) have been applied to the raw `RTCPeerConnection`
+     * stats report.  Consumers such as `WebrtcConfigAndUtils` listen to this event to add
+     * `clientVersion` / `desktopVersion` metadata and persist the record to `callsStats`.
+     *
+     * **Event data**: {@link JingleSessionMetricsEventData}
+     */
+    ON_SESSION_METRICS = "ON_SESSION_METRICS",
+    /**
+     * Fired after each WebRTC stats collection cycle (L2 every 1 s for 15 s).
+     *
+     * The event is emitted on `jingle.rxSubject` by {@link WebrtcSessionStatsHandler} once all
+     * calculations (MOS, RTT, packet-loss deltas) have been applied to the raw `RTCPeerConnection`
+     * stats report.  Consumers such as `WebrtcConfigAndUtils` listen to this event to add
+     * `clientVersion` / `desktopVersion` metadata and persist the record to `callsStats`.
+     *
+     * **Event data**: {@link JingleSessionMetricsEventData}
+     */
+    ON_SESSION_METRICS_L2 = "ON_SESSION_METRICS_L2",
+    /**
+     * Fired from the `ontrack` WebRTC handler whenever a remote media stream is
+     * created or updated, for both P2P and conference/webinar sessions.
+     *
+     * For **conference** sessions `medias` and `confId` are populated.
+     * For **P2P** sessions both fields are absent — listeners identify the call via `sid`.
+     *
+     * **Event data**: {@link JingleStreamUpdatedEventData}
+     */
+    ON_STREAM_UPDATED = "ON_STREAM_UPDATED",
+    /**
+     * Fired whenever the ICE, signalling, or DTLS connection state changes on the peer connection.
+     *
+     * Emitted from `oniceconnectionstatechange`, `onsignalingstatechange`, and
+     * `onconnectionstatechange` handlers. Consumers use this to track whether a session is
+     * connecting, connected, disconnected, or failed.
+     *
+     * **Event data**: `{ sid: string; session: JingleSession }`
+     */
+    ON_CONNECTION_STATE_CHANGED = "ON_CONNECTION_STATE_CHANGED",
+    /**
+     * Fired when a WebRTC operation on the peer connection fails (SDP offer/answer creation,
+     * transport-info application, or remote description setting).
+     *
+     * The `error.source` property identifies the failing operation (`'offer'`, `'answer'`,
+     * `'transportinfo'`, etc.).
+     *
+     * **Event data**: `{ sid: string; error: Error & { source?: string } }`
+     */
+    ON_ERROR = "ON_ERROR",
+    /**
+     * Fired when an incoming transport-replace (ICE restart) has been applied and the remote
+     * description is ready. The receiver should respond with a `transport-accept` stanza.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_TRANSPORT_REPLACE = "ON_TRANSPORT_REPLACE",
+    /**
+     * Fired after `RTCPeerConnection.setLocalDescription()` completes successfully (offer or answer).
+     *
+     * Consumers use this to send the resulting SDP to the remote peer via a Jingle stanza.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_LOCAL_DESCRIPTION_SET = "ON_LOCAL_DESCRIPTION_SET",
+    /**
+     * Fired after `RTCPeerConnection.createOffer()` succeeds, before `setLocalDescription` is called.
+     *
+     * Carries the Jingle `action` that triggered the offer (e.g. `'session-initiate'`,
+     * `'content-add'`), allowing consumers to send the appropriate Jingle stanza.
+     *
+     * **Event data**: `{ sid: string; action: string }`
+     */
+    ON_OFFER_CREATED = "ON_OFFER_CREATED",
+    /**
+     * Fired when `RTCPeerConnection.onicecandidateerror` reports a TURN/STUN or server-reachability
+     * failure (error codes 300–799).
+     *
+     * **Event data**: `{ sid: string; type: "TURN/STUN" | "SERVER_NOT_REACHABLE"; info: any }`
+     */
+    ON_ICE_CANDIDATE_ERROR = "ON_ICE_CANDIDATE_ERROR",
+    /**
+     * Fired when a TURN credential expiry is detected (ICE candidate error code 401).
+     *
+     * Consumers should refresh the ICE configuration and restart ICE.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_TURN_CREDENTIAL_EXPIRE = "ON_TURN_CREDENTIAL_EXPIRE",
+    /**
+     * Fired after `RTCPeerConnection.setRemoteDescription()` completes successfully.
+     *
+     * Consumers use this to trigger any post-remote-description logic such as sending
+     * a Jingle answer or updating transceiver state.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_REMOTE_DESCRIPTION_SET = "ON_REMOTE_DESCRIPTION_SET",
+    /**
+     * Fired after a `content-modify` renegotiation completes — the remote SDP has been applied
+     * and the local description updated with the new media parameters.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_MEDIA_MODIFIED = "ON_MEDIA_MODIFIED",
+    /**
+     * Fired when the remote peer accepts the session (`session-accept` Jingle action received
+     * or a reconnection accept is processed).
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_SESSION_ACCEPT = "ON_SESSION_ACCEPT",
+    /**
+     * Fired when a `session-info` stanza with a `<ringing>` element is received from the remote peer.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_SESSION_INFO_RINGING = "ON_SESSION_INFO_RINGING",
+    /**
+     * Fired when a `session-info` stanza with a `<hold>` element is received — the remote peer
+     * has put the call on hold.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_SESSION_HOLD = "ON_SESSION_HOLD",
+    /**
+     * Fired when a `session-info` stanza with an `<unhold>` element is received — the remote peer
+     * has resumed the call from hold.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_SESSION_UNHOLD = "ON_SESSION_UNHOLD",
+    /**
+     * Fired when a `session-info` stanza with a `<refer>` element is received — the remote peer
+     * is requesting a call transfer or conference escalation.
+     *
+     * **Event data**: `{ sid: string; status: string; confEndpointId: string; referTo: string; method: string; otherSid: string }`
+     */
+    ON_SESSION_INFO_REFER = "ON_SESSION_INFO_REFER",
+    /**
+     * Fired when a `session-info` stanza with a `<slowlink>` element is received — the remote
+     * peer is reporting packet loss on a media stream.
+     *
+     * **Event data**: `{ sid: string; direction: string; media: string; lost: number }`
+     */
+    ON_SESSION_INFO_SLOWLINK = "ON_SESSION_INFO_SLOWLINK",
+    /**
+     * Fired when a `session-terminate` Jingle stanza is received or sent.
+     *
+     * **Event data**: `{ sid: string; confId?: number; reason?: string; reasonText?: string }`
+     */
+    ON_SESSION_TERMINATE = "ON_SESSION_TERMINATE",
+    /**
+     * Fired when a `content-add` stanza is received and the session has a BUNDLE group
+     * (conference sessions only). Signals that a new media track has been added remotely.
+     *
+     * **Event data**: `{ sid: string; mediaType: string }`
+     */
+    ON_CONTENT_ADD = "ON_CONTENT_ADD",
+    /**
+     * Fired when a `content-modify` stanza is received — the remote peer has changed the
+     * direction or parameters of a media content.
+     *
+     * **Event data**: `{ sid: string; medias: any }`
+     */
+    ON_CONTENT_MODIFY = "ON_CONTENT_MODIFY",
+    /**
+     * Fired when a `content-remove` stanza is received — the remote peer has removed a
+     * media content from the session.
+     *
+     * **Event data**: `{ sid: string; mediaType: string }`
+     */
+    ON_CONTENT_REMOVE = "ON_CONTENT_REMOVE",
+    /**
+     * Fired when an incoming `session-initiate` Jingle stanza has been processed and the session
+     * object is ready. Consumers should respond with `session-accept` or `session-terminate`.
+     *
+     * **Event data**: `{ sid: string }`
+     */
+    ON_SESSION_INITIATE = "ON_SESSION_INITIATE",
+    /**
+     * Fired when the overall connection is fully established
+     * (`connectionState` or `iceConnectionState` is `"connected"` or `"completed"`).
+     *
+     * **Event data**: `{ sid: string; jingleSession: JingleSession; confId: number }`
+     */
+    ON_ESTABLISHED = "ON_ESTABLISHED",
+    /**
+     * Fired when `iceConnectionState` is `"disconnected"` or `"failed"`.
+     * Fired also when the session is in `stalled` state or any other `blocked` state that requests reconnection
+     * (conference and webinar only) Fired when `videoPacketsReceivedDeltaEmptyCount` reaches 5 consecutive empty cycles,
+     * indicating that the video subscription has stalled and the consumer should drop and
+     * re-subscribe to the conference/webinar session.
+     * Consumers should start a reconnection mechanism for this session
+     * We can get information why it happens from the reason field (it's a string)
+     *
+     * **Event data**: `{ sid: string; jingleSession: JingleSession; confId: number, reason: string }`
+     */
+    ON_RECONNECT_NEEDED = "ON_RECONNECT_NEEDED",
+    /**
+     * Fired when the video quality should be adjusted based on network conditions.
+     * Emitted when packet-loss counters cross their respective thresholds.
+     *
+     * **Event data**: `{ sid: string; confId: number; decreaseQuality: boolean }`
+     */
+    ON_VIDEO_QUALITY_CHANGE = "ON_VIDEO_QUALITY_CHANGE",
+    /**
+     * Fired when no audio packets are received in a stats cycle for a conference/webinar
+     * subscriber session, prompting the consumer to verify and potentially re-establish
+     * the publisher subscription.
+     *
+     * **Event data**: `{ sid: string; confId: number }`
+     */
+    ON_CHECK_SUBSCRIPTION_STATUS = "ON_CHECK_SUBSCRIPTION_STATUS"
+}
+/**
+ * Payload map for {@link StropheJingleEvents}, keyed by enum member.
+ * Mirrors the `**Event data**` JSDoc documented on each member above.
+ */
+export interface IJingleEventTypes {
+    ON_SESSION_METRICS: JingleSessionMetricsEventData;
+    ON_SESSION_METRICS_L2: JingleSessionMetricsEventData;
+    ON_STREAM_UPDATED: JingleStreamUpdatedEventData;
+    ON_CONNECTION_STATE_CHANGED: {
+        sid: string;
+        session: JingleSession;
+    };
+    ON_ERROR: {
+        sid: string;
+        error: Error & {
+            source?: string;
+        };
+    };
+    ON_TRANSPORT_REPLACE: {
+        sid: string;
+    };
+    ON_LOCAL_DESCRIPTION_SET: {
+        sid: string;
+    };
+    ON_OFFER_CREATED: {
+        sid: string;
+        action: string;
+    };
+    ON_ICE_CANDIDATE_ERROR: {
+        sid: string;
+        type: "TURN/STUN" | "SERVER_NOT_REACHABLE";
+        info: any;
+    };
+    ON_TURN_CREDENTIAL_EXPIRE: {
+        sid: string;
+    };
+    ON_REMOTE_DESCRIPTION_SET: {
+        sid: string;
+    };
+    ON_MEDIA_MODIFIED: {
+        sid: string;
+    };
+    ON_SESSION_ACCEPT: {
+        sid: string;
+    };
+    ON_SESSION_INFO_RINGING: {
+        sid: string;
+    };
+    ON_SESSION_HOLD: {
+        sid: string;
+    };
+    ON_SESSION_UNHOLD: {
+        sid: string;
+    };
+    ON_SESSION_INFO_REFER: {
+        sid: string;
+        status: string;
+        confEndpointId: string;
+        referTo: string;
+        method: string;
+        otherSid: string;
+    };
+    ON_SESSION_INFO_SLOWLINK: {
+        sid: string;
+        direction: string;
+        media: string;
+        lost: string;
+    };
+    ON_SESSION_TERMINATE: {
+        sid: string;
+        confId?: number;
+        reason?: string;
+        reasonText?: string;
+    };
+    ON_CONTENT_ADD: {
+        sid: string;
+        mediaType: string;
+    };
+    ON_CONTENT_MODIFY: {
+        sid: string;
+        medias: any;
+    };
+    ON_CONTENT_REMOVE: {
+        sid: string;
+        mediaType: string;
+    };
+    ON_SESSION_INITIATE: {
+        sid: string;
+    };
+    ON_ESTABLISHED: {
+        sid: string;
+        jingleSession: JingleSession;
+        confId: number;
+    };
+    ON_RECONNECT_NEEDED: {
+        sid: string;
+        jingleSession: JingleSession;
+        confId: number;
+        reason: string;
+    };
+    ON_VIDEO_QUALITY_CHANGE: {
+        sid: string;
+        confId: number;
+        decreaseQuality: boolean;
+    };
+    ON_CHECK_SUBSCRIPTION_STATUS: {
+        sid: string;
+        confId: number;
+    };
+}
+export type JingleEvent = ServiceEventsToUnion<IJingleEventTypes>;
+export declare enum MediaStreamKind {
+    /**
+     * Audio only
+     */
+    AUDIO = "audio",
+    /**
+     * Video only
+     */
+    VIDEO = "video",
+    /**
+     * Audio and video bundle
+     */
+    AUDIO_VIDEO_BUNDLE = "audiovideo",
+    /**
+     * Screen Sharing only
+     */
+    SHARING = "sharing",
+    /**
+     * Screen Sharing and audio of system
+     */
+    SHARING_BUNDLE = "sharingbundle"
+}
+export declare class JingleSession {
+    me: string;
+    sid: string;
+    connection: any;
+    initiator: string;
+    responder: string;
+    isInitiator: boolean;
+    peerjid: string;
+    state: string;
+    peerconnection: RTCPeerConnection;
+    localStreams: MediaStream[];
+    remoteStreamsObject: Record<string, MediaStream>;
+    remoteStream: any;
+    remoteStreams: any[];
+    application: any;
+    isRenegotiating: boolean;
+    conference_id: any;
+    confId: string;
+    publisherId: string;
+    remoteType: any;
+    localType: any;
+    localSDP: any;
+    remoteSDP: any;
+    localTransceivers: RTCRtpTransceiver[];
+    onHold: boolean;
+    logger: any;
+    isEchoTest: boolean;
+    calledNumber: any;
+    replaceSid: string;
+    callShift: boolean;
+    isWebinarSession: boolean;
+    autoReconnectTimer: any;
+    private reason;
+    /**
+     * SESSION SETTINGS
+     */
+    media_constraints: RTCOfferOptions | RTCAnswerOptions;
+    ice_config: any;
+    company_config: any;
+    private simulcastSettings;
+    private simulcastMCU;
+    private codecMCU;
+    /** Max bitrate in bps for the MCU video sender when simulcast is off; 0 = unconstrained */
+    private maxBitrateMCU;
+    private bandwidthLimitSettings;
+    private DTX;
+    private iceFilterList;
+    enableDSCP: boolean;
+    private jingleIqTimeout;
+    private reconnectAckTimeout;
+    timestamp: number;
+    remoteSDPNumber: number;
+    webinarId?: string;
+    /** Set to `true` once the session has been detected as stalled to avoid triggering multiple times. */
+    private isStalled;
+    /** Set to `true` once the session has been confirmed live (both counters growing) to avoid triggering multiple times. */
+    private isLive;
+    private readonly connectionStateUpdateSubject;
+    private connectionStateUpdateSubscription;
+    private readonly webrtcSessionStatsHandler;
+    /**
+     * WEBRTC STATS INFOS
+     */
+    statsinterval: any;
+    webrtcStatsInterval: any;
+    webrtcStatsIntervalL2: any;
+    webrtcStatsL2StopTimeout: any;
+    webrtcStatsL2: {
+        iteration: number;
+        active: boolean;
+        audio: {
+            totalAudioPacketsReceived: number;
+            totalAudioPacketsLost: number;
+            bytesSent: number;
+            bytesReceived: number;
+            globalRTT: number;
+            lastRoundTripTimeMeasurements: number;
+            lastDeltaAvgRTT: number;
+            oldTotalSamplesDurationMediaPlayout: number;
+            oldTotalSynthesizedSamplesDuration: number;
+        };
+        video: {
+            bytesReceived: number;
+            totalVideoPacketsReceived: number;
+        };
+        sharing: {
+            bytesReceived: number;
+        };
+        ssrc: Record<string, WebrtcStatisticsL2SsrcState>;
+        lastSeenTimestamps: Record<string, number>;
+    };
+    webrtcStats: {
+        iteration: number;
+        intervalDuration: number;
+        audio: {
+            totalAudioPacketsSent: number;
+            totalAudioPacketsReceived: number;
+            totalAudioPacketsLost: number;
+            bytesReceived: number;
+            bytesSent: number;
+            globalMOS: number;
+            meanMOS: number;
+            maxRTT: number;
+            globalJitter: number;
+            maxJitter: number;
+            globalRTT: number;
+            lastDeltaAvgRTT: number;
+            lastRoundTripTimeMeasurements: number;
+            lastAudioOutputLevel: number;
+            rms: number;
+            oldTotalSamplesDurationMediaPlayout: number;
+            oldTotalSynthesizedSamplesDuration: number;
+            synthesizedSamplesPlayed: number;
+        };
+        ice: {
+            networkType: string;
+            local: string;
+            remote: string;
+        };
+        video: {
+            totalVideoPacketsReceived: number;
+            totalVideoPacketsLost: number;
+            videoPacketsReceivedDelta: number;
+            videoPacketsReceivedDeltaEmptyCount: number;
+            videoPacketLostErrorCount: number;
+            videoPacketLostOKCount: number;
+            bytesReceived: number;
+        };
+        sharing: {
+            bytesReceived: number;
+        };
+        ssrc: {};
+        stalled: boolean;
+        codec: string;
+    };
+    /**
+     * RECONNECTION STUFF
+     */
+    isReconnecting: boolean;
+    reconnectTriggered: boolean;
+    private savedIncomingTransportReplace;
+    private createdOfferTimeout;
+    private transportAcceptBackstopTimer;
+    reconnectionOptions: any;
+    private sendIceCandidatesPromise;
+    hasSentTransportReplace: boolean;
+    /**
+     * LYFE CYCLE AND APIs
+     */
+    constructor(me: string, sid: string, connection: any, jingleIqTimeout?: number);
+    initiate(peerjid: any, isInitiator: any): Promise<void>;
+    private addPeerConnectionListeners;
+    private createRemoteStreamForConference;
+    private createOrUpdateRemoteStreams;
+    updateRemoteStreams(): void;
+    replaceVideoTrack(videoTrack: any): void;
+    replaceTransceiverTrack(track: any, id?: number): Promise<void>;
+    updateSessionStreamsAndTransceivers(streams?: MediaStream[]): Promise<void>;
+    private updateConferenceOrWebinarStreams;
+    private addVideoTransceiverForConfOrWebinar;
+    updateP2PSessionWithNewStreams(streams?: MediaStream[]): Promise<void>;
+    private replaceTrackInTransceiver;
+    /**
+     * Tear the session down.
+     *
+     * @param reason - jingle terminate reason.
+     * @param keepLocalTracks - leave the LOCAL capture running instead of stopping it. Terminating
+     * normally destroys everything it can reach, which is right when the media is over -- but a
+     * caller can outlive the session it published through: the speaker PiP unpublishes a screen
+     * share while still compositing that very capture into its camera. Remote tracks are always
+     * stopped; they belong to a peer connection that is closing either way.
+     */
+    terminate(reason?: any, keepLocalTracks?: boolean): void;
+    active(): boolean;
+    private sendIceCandidate;
+    private sendIceCandidates;
+    hackOfferSDP(offer: RTCSessionDescriptionInit): RTCSessionDescriptionInit;
+    hackMsLabelSDP(offer: RTCSessionDescriptionInit): RTCSessionDescriptionInit;
+    sendTransportReplace(): Promise<void>;
+    sendTransportReplaceWithAwait(): Promise<void>;
+    updateIceConfigAndReconnect(iceServers?: []): Promise<void>;
+    reconnectSession(): Promise<void>;
+    saveIncomingTransportReplace(elem: any, desctype: any, ack: any): void;
+    clearTransportAcceptBackstop(): void;
+    acceptSavedIncomingTransportReplace(): void;
+    rejectSavedIncomingTransportReplace(): void;
+    sendCreatedOfferWithAckWait(offer: RTCSessionDescriptionInit, actionName: any): void;
+    sendOffer(iceRestart: boolean, application?: any): Promise<void>;
+    createdOffer(offer: RTCSessionDescriptionInit, actionName?: string, application?: any): Promise<void>;
+    contentAdd(mediaStreamType: string, addMediaLocalType?: boolean): Promise<void>;
+    contentRemove(mediaStreamType: string, addMediaLocalType?: boolean): Promise<void>;
+    contentModify(mediaStreamType: string, options?: any): Promise<void>;
+    contentUpdateOffer(sdp: any, mediaStreamType: string, actionType: any, addLocalType?: boolean): Promise<void>;
+    contentChangeAnswer(sdp: any, updateStreams: any): Promise<void>;
+    private setCodecPreferencesForSession;
+    private sortByMimeTypes;
+    private setPreferredProfileForCodec;
+    private setCodecPreference;
+    getSenders(): Promise<RTCRtpSender[]>;
+    setRemoteDescription(elem: any, desctype: any): Promise<void>;
+    addIceCandidate(elem: any): void;
+    private addIceCandidateWithRetry;
+    sendAnswer(type?: string): Promise<void>;
+    private createdAnswer;
+    sendTerminate(reason: string | null, text?: string): void;
+    mediaModified(updateStreams?: any): void;
+    muteAudio(muted: any): void;
+    sendMute(muted: any, content: any): void;
+    sendRefer(confendpointId: any, referTo: any): void;
+    sendP2PTransferCallRefer(replaceSid: any, referTo: any): void;
+    sendReferStatus(status: any): void;
+    sendReferTransfer(method: any, referTo: any): void;
+    sendHold(hold: boolean): Promise<void>;
+    sendIceFailed(): void;
+    sendRinging(): void;
+    modifyEncodingState(state: any, customMuteAction: any): void;
+    enableHQEncoding(enableHQEncoding: any): Promise<void>;
+    disableSubStreamLevel(levelIndex: any): void;
+    removeSsrcGroupFidForFirefox(sdp: string): string;
+    setDSCPValues(): Promise<void>;
+    webinarGetMedia(): any;
+    /**
+     * Whether a stall verdict is allowed to ask for a reconnection on this session.
+     *
+     * Only the two legs where the verdict proved trustworthy pass:
+     * - a **direct P2P** call (the peer JID is a regular Rainbow user JID);
+     * - a **hybrid PBX** call -- OXO / OXE -- relayed by a MediaPillar gateway (`mp_` peer JID).
+     *
+     * Everything else keeps the CRRAINB-53973 block:
+     * - **HUB** calls -- the MediaPillar of the Rainbow Voice Communication Platform, recognised by
+     *   the `mp_rvcp` peer JID -- the exact leg whose false verdicts caused the audio loss;
+     * - **conferences and webinars** -- SFU / MCU sessions, which always carry a conference or room id.
+     */
+    private isStallReconnectAllowed;
+    /**
+     * Called by {@link WebrtcSessionStatsHandler} when L2 metrics detect that the session is stalled
+     * (no growth in `responsesReceived` or `bytesReceived` across consecutive ticks).
+     * Guards against repeated calls via {@link isStalled}.
+     *
+     * **Reconnection on this verdict is restricted to direct P2P and hybrid PBX (OXO / OXE) calls**
+     * (CRRAINB-53973), see {@link isStallReconnectAllowed}. Chromium freezes the counters of its audio
+     * `inbound-rtp` report when the remote peer switches RTP source mid-session — which the HUB does on
+     * every answered softphone call, when it stops its own ringback tone and resumes relaying the carrier
+     * stream. The verdict then fires on calls whose audio is perfectly fine, and the ICE restart it
+     * triggers is fatal on that leg: the browser re-gathers, the peer rejects the trickled candidates,
+     * and the call loses audio in both directions.
+     *
+     * The flag and the log below are kept so the frequency stays measurable in production on the legs
+     * that stay blocked. Set `localStorage.qaStallReconnectEnabled` to `"true"` to reconnect on every
+     * session type regardless.
+     */
+    onSessionStalled(): void;
+    /**
+     * Called by {@link WebrtcSessionStatsHandler} when L2 metrics confirm the session is live —
+     * both `responsesReceived` and `bytesReceived` are growing while ICE and DTLS are connected.
+     * Guards against repeated calls via {@link isLive}.
+     */
+    onSessionLive(): void;
+    /**
+     * Log DTLS / ICE transport details (selected candidate pair state, candidate types and protocols) so a blocked
+     * session can be attributed to the client network or to the media server. Diagnostic only, never throws.
+     */
+    private logTransportDiagnostics;
+    private onConnectionStateUpdate;
+}
+//# sourceMappingURL=strophe.jingle.session.d.ts.map

package/lib/libs/strophe/strophe.jingle.session.d.ts.map

@@ -0,0 +1 @@
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