@ale-rainbow/web-sdk 5.1.0

package/lib/services/webrtc/webrtcP2P.service.d.ts

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+/**
+ * @module Webrtc
+ */
+import type { Subscription } from "rxjs";
+import type { Call } from "../../models/call.model";
+import { CallRB, CallStatus } from "../../models/call.model";
+import type { VirtualBackgroundService } from "../../services/virtualBackground/virtualBackground.service";
+import { WebrtcP2PEventHandler } from "../../services/webrtc/webrtcP2PEventHandler";
+import type { Contact } from "../../models/contact.model";
+import { Service } from "../../services/service";
+import type { JingleSession } from "../../libs/strophe/strophe.jingle.session";
+import type { ConversationRB } from "../../models/conversation.model";
+/**
+ * @internal
+ */
+export declare const WEBRTC_P2P_SVC = "WebrtcP2PService";
+/**
+ * @eventProperty
+ * WebrtcP2PService Events.
+ */
+export declare enum WebrtcP2PServiceEvents {
+    /**
+     * @eventProperty
+     * This RB event is send when P2P called is busy (in order to manage a busy tone).
+     */
+    RAINBOW_ON_WEBRTC_BUSY = "RAINBOW_ON_WEBRTC_BUSY",
+    /**
+     * @eventProperty
+     * This RB event is send when we need to ask for snapshot from other ressources to update the list of calls
+     */
+    RAINBOW_SUPERVISION_SNAPSHOT_NEEDED = "RAINBOW_SUPERVISION_SNAPSHOT_NEEDED",
+    /**
+     * @eventProperty
+     * This RB event is send when a call is updated; This event can be used to manage all calls in single place;
+     */
+    RAINBOW_ON_CALL_UPDATED_EVENT = "RAINBOW_ON_CALL_UPDATED_EVENT",
+    /**
+     * @eventProperty
+     * This RB event is send when a MP call is updated
+     */
+    RAINBOW_ON_MEDIA_PILLAR_CALL_UPDATED_EVENT = "RAINBOW_ON_MEDIA_PILLAR_CALL_UPDATED_EVENT"
+}
+/**
+ * @public
+ * Webrtc P2P Service interface
+ */
+export interface IWebrtcP2PService {
+    /**
+     * Make a standard P2P call toward a contact
+     * @param contact - The Call object that we want to remove
+     * @param mediaType - The media type of call (could be 'audio', 'video' (i.e audio and video), 'videoOnly' (no audio, only video), 'sharing' (audio and sharing) and 'sharingOnly' (only sharing, no audio))
+     * @param message - [optional] additional message send to the distant party for the reason of the call
+     * @returns Returns the call
+     */
+    makeCall: (contact: Contact, mediaType: "audio" | "video" | "videoOnly" | "sharing" | "sharingOnly", message?: string) => Promise<Call | undefined>;
+    /**
+     * Answer an incoming webrtc call
+     * @param call - The Call object that we want to remove
+     * @param mediaType - (optional) The media type to answer the call (by default, it's only audio anwser). In case of audio and video incoming call
+     *   the user can use type = "video" to answer the call with audio and video;
+     */
+    answerCall: (call: CallRB, mediaType: string) => void;
+    /**
+     *Terminate a webrtc call
+     * @param call - The Call object that we want to remove
+     * @param cause - (optional) cause to reject the call
+     * @param notifyOtherParticipant -(optional) Indicates wether the other participant should be notified by a xmpp message or not
+     */
+    terminateCall: (call: CallRB, cause: string, notifyOtherParticipant: boolean) => void;
+    /**
+     * Allow to mute / unmute a call
+     * @param call - The call
+     * @param state - if true, the audio in the call will be muted
+     * @returns The updated call object
+     */
+    muteCall: (call: CallRB, state: boolean) => CallRB;
+    /**
+     * Put the call on Hold
+     * @param call - The call to be put on hold
+     * @param hold - When true the call is hold
+     * @returns the updated call object
+     */
+    holdCall: (call: CallRB, state: boolean) => CallRB;
+    /**
+     * Remove a media from call
+     * @param call - The call object
+     * @param mediaType - the mediaType to remove from the call
+     * @returns Resolved when the action is taken into accout
+     */
+    removeMediaFromCall: (call: CallRB, mediaType: "video" | "sharing") => Promise<CallRB>;
+    /**
+     *Allow to send DTMF
+     * @param call - if true, the audio in the call will be muted
+     * @param dialString - The conversation that is related to the call
+     * @param skipHandle - [optional] Should skip the ontonechange event -- false by default
+     */
+    sendDTMF: (call: CallRB, dialString: string, skipHandle?: boolean) => Promise<void>;
+    /**
+     * Get a call by it's ID
+     * @param callId - The id of the call to find
+     * @returns The call
+     */
+    getCallById: (callId: string) => CallRB;
+    /**
+     * Find and return the current active call (the last one found), or null otherwise
+     * @returns Return the current active call (the last one found), or null otherwise
+     */
+    getCurrentActiveCall: () => CallRB | null;
+    /**
+     * If the current user has any kind of webrtc call (could also be with WebRTC GW / Media Pillar)
+     * @returns True if user has a call
+     */
+    isUserContactInCall: () => boolean;
+    /**
+     * Subscribe to updates on the current calls sessions (all events are of RBEvent type `{ name: string, data:any }`);
+     * All public events that are fired by this service are described in the documentation.
+     * NB There might be some events that are not documented here, they should be ignored
+     * @param handler - The call-back function that will be subscribed to the RxJS subject
+     * @param eventName - The name of the event to filter
+     * @returns Returns RxJS Subscription
+     */
+    subscribe: (handler: any, eventName?: string) => Subscription;
+}
+/**
+ * @internal
+ */
+export declare class WebrtcP2PService extends Service implements IWebrtcP2PService {
+    started: boolean;
+    connected: boolean;
+    RTC: any;
+    calls: Record<string, CallRB>;
+    autoreleaseTimeout: any;
+    makingCall: boolean;
+    reconnectCall: boolean;
+    alreadyAttaching: boolean;
+    firefoxPreferedSharing: string;
+    webrtcP2PEventHandler: WebrtcP2PEventHandler;
+    private isMixPstnWebRtcAllowed;
+    private audioProfileChanging;
+    private subscriptions;
+    private xmppConnectionSubscription;
+    private replaceBackgroundSubscription;
+    private deviceManagementSubscription;
+    private listeners;
+    virtualBackgroundService?: VirtualBackgroundService;
+    private logger;
+    private mainService;
+    private rxSubject;
+    private eventService;
+    private webrtcConfigAndUtils;
+    private contactService;
+    private xmppService;
+    private profileService;
+    private deviceManagementService;
+    private settingsService;
+    private conversationService;
+    getDesktopSharingSource(): any;
+    askToAddSharing(__conversationId: any, __shouldAskForAudio?: boolean): boolean;
+    askToStartSharing(__contactId: any, __media: any, __shouldAskForAudio?: boolean): boolean;
+    isNativeApplication(): boolean;
+    static getInstance(): WebrtcP2PService;
+    static build(): WebrtcP2PService;
+    private constructor();
+    start(): void;
+    stop(): void;
+    reconnect(): void;
+    private attachHandlers;
+    private removeHandlers;
+    private reconnectSessionsIfNeeded;
+    private webrtcConfigAndUtilsEvent;
+    /**
+     * Send an event to the RX Subject (sends always a RBEvent type)
+     * @param name - The name of the event
+     * @param data - The data of the event
+     */
+    sendEvent(name: string, data?: any): void;
+    /**
+     * Subscribe to updates on the current calls sessions (all events are of RBEvent type {name: string, data:any});
+     * All public events that are fired by this service are described in the documentation.
+     * NB There might be some events that are not documented here, they should be ignored
+     * @param handler - The call-back function that will be subscribed to the RxJS subject
+     * @param eventName - The name of the event to filter
+     * @returns Returns RxJS Subscription
+     */
+    subscribe(handler: any, eventName?: string): Subscription;
+    /**
+     * Make a standard P2P call toward a contact
+     * @param contact - The Call object that we want to remove
+     * @param mediaType - The media type of call (could be 'audio', 'video' (i.e audio and video), 'videoOnly' (no audio, only video), 'sharing' (audio and sharing) and 'sharingOnly' (only sharing, no audio))
+     * @param message - [optional] additional message send to the distant party for the reason of the call
+     * @param message - [optional] TRUE in case we want to propose add System audio on top of sharing
+     * @returns Returns the call
+     */
+    makeCall(contact: Contact, mediaType: string, message?: string, addAudioToSharingCall?: boolean): Promise<CallRB>;
+    /**
+     * Answer an incoming webrtc call
+     * @param call - The Call object that we want to remove
+     * @param mediaType - (optional) The media type to answer the call (by default, it's only audio anwser). In case of audio and video incoming call
+     *   the user can use type = "video" to answer the call with audio and video;
+     */
+    answerCall(call: CallRB, mediaType?: string): void;
+    /**
+     * Terminate a webrtc call
+     * @param call - The Call object that we want to remove
+     * @param cause - (optional) cause to reject the call
+     * @param notifyOtherParticipant - (optional) Indicates wether the other participant should be notified by a xmpp message or not
+     */
+    terminateCall(call: CallRB, cause?: string, notifyOtherParticipant?: boolean): void;
+    /**
+     * Allow to mute / unmute acall
+     * @param call - The call
+     * @param state - if true, the audio in the call will be muted
+     * @returns The updated call object
+     */
+    muteCall(call: CallRB, state: boolean): CallRB;
+    /**
+     * Put the call on Hold
+     * @param call - The call to be put on hold
+     * @param  hold - When true the call is hold
+     * @returns the updated call object
+     */
+    holdCall(call: CallRB, hold: boolean): CallRB;
+    /**
+     * Remove a media from call
+     * @param call - The call object
+     * @param mediaType - the mediaType to remove from the call
+     */
+    removeMediaFromCall(call: CallRB, mediaType: "video" | "sharing"): Promise<CallRB>;
+    /**
+     * Allow to send DTMF
+     * @param call - if true, the audio in the call will be muted
+     * @param dialString - The conversation that is related to the call
+     * @param skipHandle - [optional] Should skip the ontonechange event -- false by default
+     */
+    sendDTMF(call: CallRB, dialString: string, skipHandle?: boolean): Promise<void>;
+    /**
+     * Remove The Call object and terminates the webrtc session related to this call.
+     * @param call - The Call object that we want to remove
+     * @param reason - The reason to end the call that is send to the distant
+     * @param text - Additional text send to the distant when the webrtc session is released
+     */
+    removeCallObject(call: CallRB, reason?: string, text?: string): void;
+    private onReplaceBackgroundChangeEvent;
+    private putActiveWebrtcCallOnHold;
+    private onConnectionStateChangeEvent;
+    isUserContactInCall(): boolean;
+    isUserContactInOtherCall(call: CallRB): boolean;
+    /**
+     * Find and return the current active call (the last one found), or null otherwise
+     * @returns Return the current active call (the last one found), or null otherwise
+     */
+    getCurrentActiveCall(): CallRB | null;
+    /**
+     * Retrieve the list of calls with a status equal to a given status (or a list of given status) (if some exist)
+     * @public
+     * @param {any} [searchedCallStatus] Status (or array of status) of calls to search. If not passed, it returns all calls except incoming/queued ringing calls
+     * @returns {CallRB[]} List of calls matching the searched status (or empty array if no calls found)
+     * @instance
+     * @function
+     */
+    getCalls(searchedCallStatus?: CallStatus[] | CallStatus): CallRB[];
+    /**
+     * Return an array with all the WebRTC Jingle active sessions
+     * @returns Return an array with all the WebRTC Jingle active sessions
+     */
+    getCurrentActiveSessions(): any[];
+    /**
+     * Return an array with all the Call objects that are not ended and not related to MP
+     * @returns Return an array with all the Call objects that are not ended and not related to MP
+     */
+    getCurrentCallsNotMediaPillar(): CallRB[];
+    /**
+     * Return an array with all the Call objects that are not ended and not related to MP
+     * @returns Return an array with all the Call objects that are not ended and not related to MP
+     */
+    getCurrentIncomingCallsNotMediaPillar(): CallRB[];
+    /**
+     * Check if the user has an active webrtc session that is not related to Media Pillar / WebRTC GW
+     * @returns Return true if the user has an active webrtc session that is not related to Media Pillar / WebRTC GW
+     */
+    existWebrtcActiveSessionNotMP(): boolean;
+    /**
+     * Check if the user has an active webrtc session that is related to Media Pillar / WebRTC GW
+     * @returns Return true if the user has an active webrtc session that is related to Media Pillar / WebRTC GW
+     */
+    existMediaPillarAudioCall(): boolean;
+    /**
+     * Check if the user has an answering webrtc session that is related to Media Pillar / WebRTC GW
+     * @returns Return true if the user has an answering webrtc session that is related to Media Pillar / WebRTC GW
+     */
+    existMediaPillarAnsweringCall(isSipWise?: boolean): boolean;
+    /**
+     * Return an array of Call that are related to Media Pillar / Webrtc GW
+     * @returns Return an array of Call that are related to Media Pillar / Webrtc GW
+     */
+    getMediaPillarAudioCalls(): CallRB[];
+    /**
+     * Return an array of Call that are related to Media Pillar / Webrtc GW
+     * @returns Return an array of Call that are related to Media Pillar / Webrtc GW
+     */
+    getActiveMediaPillarAudioCall(): CallRB;
+    isInMediaPillarAudioCallWithContactJid(jid: any): boolean;
+    hasMediaPillarCallWithContactJid(jid: string): boolean;
+    hasOtherWebrtcCallWithContactJid(callId: string, contactJid: string): boolean;
+    hasIncomingWebrtcCallWithContactJid(contactJid: string): boolean;
+    /**
+     * Return the Call that is on hold (put on hold by the user)
+     * @returns Return the Call that is on hold (put on hold by the user)
+     */
+    getCallOnHold(): CallRB;
+    /**
+     * Return the Call that is on hold (put on hold by the distant party)
+     * @returns Return the Call that is put on hold
+     */
+    getCallPutOnHold(): CallRB;
+    /**
+     * getWebrtcAudioCall
+     * @returns the audio webrtc call
+     */
+    getWebrtcAudioCall(): CallRB;
+    getWebrtcAudioCalls(): CallRB[];
+    getSecondSharingCallForPbxCall(pbxCall: CallRB): CallRB;
+    /**
+     * getActiveWebrtcAudioCall
+     * @returns the audio webrtc call that is in ACTIVE STATE
+     */
+    getActiveWebrtcAudioCall(): CallRB;
+    /**
+     * GetWebrtcCallsOnHold
+     * @returns the audio webrtc call that is in HOLD STATE
+     */
+    getWebrtcCallsOnHold(): CallRB[];
+    getMPWebrtcAudioCallByState(soughtCallState: any): CallRB;
+    getCurrentSharingCall(): CallRB;
+    /**
+     * Ends all sharing calls related to a given contact
+     * @param contact - The contact related to all calls
+     */
+    endAllCallsForContact(contact: Contact): void;
+    private onAudioVideoInputDeviceChange;
+    getCallById(callId: string): CallRB;
+    /**
+     * Find and return a call by the distant contact's jid or full jid
+     * @param contactJid - The jid of the contact that we want to find
+     * @param fullJid - [optional] The full JID of the contact related to the call
+     * @returns The call
+     */
+    getCallByJid(contactJid: string, fullJid?: string): CallRB;
+    /**
+     * Send to remote (others clients) a dummy reject for an incoming webrtc call
+     * then the others remote ressources manage this call as rejected even if locally it isn't
+     * (mainly for RQRAINB-9057)
+     * @param callId - The tel Call id that we want to remotly manage as rejected
+     */
+    dummyRejectCallForRemote(callId: string): void;
+    /**
+     * Release a standard call
+     * @param call - The Call object that we want to release
+     * @param reason - (optional) reason to release the call
+     * @param text - (optional) additional text
+     */
+    releaseCall(call: CallRB, reason?: string, text?: string): void;
+    callShiftPBX(cstaCallId: string, mpJid: string, otherSid: string): Promise<CallRB>;
+    callShift(otherSid: string, referTo: string, media?: string): Promise<void>;
+    /**
+     * Make a Jingle call toward the destionation (could be a MP call)
+     * @param call - The Call object
+     * @param mediaPillarJid - (optional) the media Pillar Jid only in case of webrtc gateway media Pillar
+     * @param phoneNumber - (optional) the number to join by mediaPillar
+     * @param otherSid - parameter only used in case of transfer webrtc to webrtc call
+     * @param callShift - in case of call shift
+     */
+    makeJingleCall(call: CallRB, mediaPillarJid?: string, phoneNumber?: string, otherSid?: string, callShift?: boolean): Promise<void>;
+    /**
+     * Make the jingle webtc call with sharing and audio towards the destination in the call
+     * @param call - The Call object
+     */
+    makeJingleSharingCall(call: CallRB): Promise<void>;
+    /**
+     * UCAAS Method -- to be used to start the desktop sharing in case of Desktop App
+     * @param contact - The Call object that we want to remove
+     * @param media - The media type of call (could be 'sharing' (audio and sharing) and 'sharingOnly' (only sharing, no audio))
+     */
+    startDesktopSharing(contact: Contact, media: string): void;
+    /**
+     * UCAAS Method -- Desktop app has annulated start of sharing
+     */
+    startSharingCancelled(): void;
+    sendUpdateCallEvent(call: CallRB): void;
+    /**
+     * Answer an incoming jingle call
+     * @param call - the call
+     */
+    answerJingleCall(call: CallRB): Promise<void>;
+    /**
+     * UCAAS Method -- P2P call is transferred to a SFU conference
+     * @param call - The P2P call that should be removed
+     */
+    onCallTransferred(call: CallRB): void;
+    /**
+     * UCAAS Method -- Desktop app to add a sharing to conversation
+     * @param conversation - The conversation related
+     */
+    askToAddDesktopSharing(conversation: ConversationRB): void;
+    /**
+     * Add audio to call (or change the current audio input device)
+     * @param call - The call object
+     * @param mediaModify - if True, the action is to "change" the current input audio device,
+     *  and not add audio to the current call
+     * @returns Resolved when the action is taken into accout
+     */
+    addAudioToCall(call: CallRB, mediaModify?: boolean): Promise<CallRB>;
+    /**
+     * Add Sharing to a existing active call
+     * @param call - The call object
+     * @returns Resolved when the action is taken into accout
+     */
+    addSharingToCall(call: CallRB, withAudio?: boolean): Promise<CallRB>;
+    /**
+     * Add video to active call (or modify the video input)
+     * @param call - The call object
+     * @param mediaModify - if True, the action is to "change" the current video audio device, and not add video to the current call
+     * @returns Resolved when the action is taken into accout
+     */
+    addVideoToCall(call: CallRB, mediaModify?: boolean): Promise<CallRB>;
+    private changeBackgroundForCall;
+    /**
+     * Remove the sharing from the call
+     * @param call - The call object
+     */
+    removeSharingFromCall(call: CallRB): Promise<CallRB>;
+    /**
+     * Remove the sharing and video from the call
+     * @param call - The call object
+     * @returns Resolves when the action is taken into account
+     */
+    private removeVideoAndSharingFromCall;
+    /**
+     * Remove video from the current call
+     * @param call - The call object
+     * @returns Resolved when the action is taken into account
+     */
+    removeVideoFromCall(call: CallRB): Promise<CallRB>;
+    /** ***********************************************************/
+    /** ***********************************************************/
+    /**
+     * Allow to mute / unmute the current call UCAAS ONLY FONCTION
+     * @param conversation - The conversation that is related to the call
+     * @param state - if true, the audio in the call will be muted
+     * @returns The updated call object
+     */
+    muteAudio(conversation: ConversationRB, state: boolean): CallRB;
+    /**
+     * Put the call on Hold
+     * NB this will stop all video and sharing coming from this user before the hold action
+     * @param call - The call to be put on hold
+     * @returns the updated call object
+     */
+    private putCallOnHold;
+    /**
+     * Retrieve the call on Hold
+     * @param call - The call to be put on hold
+     * @returns the updated call object
+     */
+    private retrieveCallOnHold;
+    /**
+     * Functions related to the refer action - update from P2P call to Conference call
+     * @returns the updated call object
+     */
+    referTo(session: any, confEndpointId: string, referTo: string): void;
+    /**
+     * Transfer a webrtc call on hold to another active call (only works for 2 webrtc P2P calls)
+     * @param session - The session related to the call on hold
+     * @param otherSession - The active call session, to whom we want to transfer the call
+     */
+    transferCall(session: JingleSession, otherSession: JingleSession): void;
+    sendReferStatus(callId: string, status: string): void;
+    sendReferTransfer(callId: string, method: string, referTo: string): void;
+    /**
+     * Stops and removes all Tracks from the Media Stream array
+     * @param streams - Array of Media Streams
+     */
+    disableStreams(streams: MediaStream[]): void;
+    pushInLocalStreams(stream: MediaStream, callId: string): void;
+    openErrorPopup(msg?: string): void;
+    private removeSharingStreamFromCall;
+    private removeAudioVideoStreamFromCall;
+    private removeAllStreamsFromCall;
+    resetToSafeState(call?: CallRB): void;
+    dropAudio(): Promise<void>;
+    /** ***********************************************************/
+    /**
+     * Method calculatePresenceMessage
+     * @private
+     * @memberof WebrtcP2PService
+     * UCAAS Method -- Calculate the call message to be shown on the UI
+     * @param {Call} call call object
+     * @returns {string} The new message to be shown for our current presence
+     */
+    calculatePresenceMessage(call: CallRB): string;
+    /**
+     * Method updateRemoteTypeMedia
+     * @private
+     * @memberof WebrtcP2PService
+     * UCAAS Method -- Calculate the remoteMedia object value
+     * @param {string} remoteType The new remote type string
+     * @param {Call} call call object
+     */
+    updateRemoteTypeMedia(remoteType: string, call: CallRB): void;
+    private updateLocalAudioStreamForCall;
+    private updateLocalMediaType;
+    updateCurrentCall(call: CallRB, session: JingleSession, type: any, action: any): Promise<void>;
+    private handleToneChangeEvent;
+    private silentToneChangeEventHandle;
+    clearSrcObjectsFromElements(): void;
+    attachDistantMediaStreamsUnifiedPlan(call: any): void;
+    attachDistantMediaStreams(attach: any, call: any): void;
+    attachLocalVideoOnlyStreamToElement(call: any, attach: any, element: any): void;
+    playStartRecordSoundInCall(): Promise<void>;
+}
+//# sourceMappingURL=webrtcP2P.service.d.ts.map

package/lib/services/webrtc/webrtcP2P.service.d.ts.map

@@ -0,0 +1 @@
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package/lib/services/webrtc/webrtcP2PEventHandler.d.ts

@@ -0,0 +1,55 @@
+import { CallRB } from "../../models/call.model";
+import type { WebrtcP2PService } from "../../services/webrtc/webrtcP2P.service";
+import type { Contact } from "../../models/contact.model";
+import type { JingleSession } from "../../libs/strophe/strophe.jingle.session";
+export declare class WebrtcP2PEventHandler {
+    private webrtcP2PService;
+    private xmppService;
+    private logger;
+    private contactService;
+    private eventService;
+    private webrtcConfigAndUtils;
+    private settingsService;
+    private conversationService;
+    private nameSpace;
+    private messageHandlerRef;
+    private xmppServiceSubscription;
+    constructor(webrtcP2PService: WebrtcP2PService);
+    destroy(): void;
+    private onJingleMessage;
+    private onJingleSessionInitiate;
+    private onJingleTransportReplace;
+    private onJingleWebrtcStatsEvent;
+    private onJingleHoldEvent;
+    private onJingleUnholdEvent;
+    private onJingleCallAccepted;
+    setRemoteTypeMedia(remoteType: string[], call: CallRB): void;
+    setLocalTypeMedia(localType: string[], call: CallRB): void;
+    sendProposition(contact: Contact, media: string, message?: string, addAudioToSharingCall?: boolean): Promise<CallRB>;
+    retractProposition(call: any): void;
+    acceptProposition(call: any, type: any): void;
+    rejectProposition(call: CallRB, __cause?: string, notifyOtherParticipant?: boolean): void;
+    notifyRemoteOfDummyRejectProposition(callId: string, __cause?: string): void;
+    onMessageReceived(stanza: any): boolean;
+    private onMessageReceivedError;
+    private getOrCreateContact;
+    private onPropositionReceived;
+    private onRetractReceived;
+    private onRejectReceived;
+    private onAcceptReceived;
+    private onProceedReceived;
+    /** ***********************************************************/
+    /** ***********************************************************/
+    incomingCall(callId: string): Promise<void>;
+    activeCall(callId: any): void;
+    terminatedCall(callId: any, callReason?: string, doNotUpdatePresence?: boolean): void;
+    onJingleError(sid: any, error: any): void;
+    onSessionConnectionStateUpdate(sid: string, sess: JingleSession): void;
+    onReconnectNeeded(session: JingleSession, reason: string): void;
+    onCallConnected(session: JingleSession): void;
+    mediaModified(sid: any): void;
+    onTrasferCallEvent(referTo: string, otherSid: string): Promise<void>;
+    onIncomingTrasferredCall(sid: string): Promise<void>;
+    onIncomingShiftCall(oldCallId: string, newCallId: string): Promise<void>;
+}
+//# sourceMappingURL=webrtcP2PEventHandler.d.ts.map

package/lib/services/webrtc/webrtcP2PEventHandler.d.ts.map

@@ -0,0 +1 @@
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