@ale-rainbow/web-sdk 5.1.0

package/lib/services/webrtc/webrtcConfigAndUtils.d.ts

@@ -0,0 +1,170 @@
+import type { Subscription } from "rxjs";
+import type { CallRB } from "../../models/call.model";
+import { Service } from "../../services/service";
+export declare const WEBRTC_CONFIG_UTILS_SVC = "WebrtcConfigAndUtils";
+export interface IWebrtcConfigAndUtils {
+    RTC: any;
+}
+/**
+ * This module contains all the webrtc configuration tools and utils that can be used to manages webrtc audio/video calls
+ * This module can be used to set the video quality and frame rate for all P2P calls
+ * The lifecycle of this module is managed by the WebRTC P2P Service
+ */
+export declare class WebrtcConfigAndUtils extends Service implements IWebrtcConfigAndUtils {
+    /**
+     * The current webrtc configuration used (TURN servers and DC)
+     */
+    webrtcConfig: any;
+    /**
+     * The current webrtc company configuration (if set for the company)
+     */
+    companyConfig: any;
+    /** Overrides applied after server config is fetched — set via WebinarService.configure() */
+    companyConfigOverrides: Record<string, any>;
+    /**
+     * The current webrtc video and sharing quality configuration when in call with mobile phone
+     */
+    mobileVideoQualityConfiguration: {
+        video: {
+            fixedFrameRate: number;
+            fixedVideoWidth: number;
+            fixedVideoHeight: number;
+        };
+        sharing: {
+            fixedFrameRate: number;
+            fixedVideoWidth: number;
+            fixedVideoHeight: number;
+        };
+    };
+    /**
+     * The current webrtc video and sharing quality configuration
+     */
+    videoQualityConfiguration: {
+        video: {
+            fixedFrameRate: number;
+            fixedVideoWidth: number;
+            fixedVideoHeight: number;
+        };
+        sharing: {
+            fixedFrameRate: number;
+            fixedVideoWidth: number;
+            fixedVideoHeight: number;
+        };
+    };
+    /**
+     * The RTC object that contains the attach and detach functions to add stream to DOM element based on the platform
+     * The attachSessionMediaToElement is used for session that have only 1 media (for conferences case); We need to send the HTML element ID + the session that we want to "attach" to that element
+     */
+    RTC: any;
+    /**
+     * 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
+     */
+    subscribe(handler: any): Subscription;
+    /**
+     * Send an event to the RX Subject (sends always a RBEvent type)
+     * @param string - name The name of the event
+     * @param data - The data of the event
+     */
+    private sendEvent;
+    /**
+     * The Calls Stats object that contains all the statistics for the current call (that will be uploaded to the server at the end of the call)
+     */
+    callsStats: {};
+    askToAddSharing(__conversationId: string, __shouldAskForAudio?: boolean): boolean;
+    private logger;
+    private settingsService;
+    private authService;
+    private xmppService;
+    private mainService;
+    private contactService;
+    private eventService;
+    private deviceManagementService;
+    private profileService;
+    private rxSubject;
+    private jingleSubscription;
+    private localPeerConnection;
+    private ttl;
+    private ttlTimeout;
+    private started;
+    private statsWaitingQueue;
+    static getInstance(): WebrtcConfigAndUtils;
+    static build(): WebrtcConfigAndUtils;
+    private constructor();
+    start(): Promise<void>;
+    stop(): void;
+    private onConnectionStateChangeEvent;
+    private onJingleMessage;
+    private makeLocalEchoTest;
+    private logPeerConnectionStates;
+    /**
+     * This method builds the local peer connection
+     */
+    private buildLocalPeerConnection;
+    private destroyLocalPeerConnection;
+    getBrowserMethodHandler(): any;
+    private jingleTurnCredentialExpire;
+    setBrowserMethodHandler(): any;
+    getDesktopSharingSource(): any;
+    getDesktopVersion(): string;
+    private getConstraintsForMedia;
+    getBrowserMedia(userMedia: any[], fixedVideoWidth?: number, fixedVideoHeight?: number, fixedFrameRate?: number, referer?: any, getAudioForSharing?: boolean, isMediaPillarCall?: boolean): Promise<MediaStream | undefined>;
+    /**
+     * Calls `getUserMedia` and transparently retries a bounded number of times when it fails with
+     * `NotReadableError` ("Could not start audio source") — a transient failure seen with some USB
+     * headsets (e.g. EPOS/Sennheiser) whose driver briefly locks the audio device while switching
+     * mode right after an off-hook/answer signal.
+     *
+     * Any other error (permission denied, device missing, overconstrained…) is rethrown immediately,
+     * unchanged, on the first attempt — same behaviour as before this retry was added.
+     *
+     * Bounded by {@link NOT_READABLE_ERROR_MAX_RETRIES}: once exhausted, the last error is rethrown
+     * as-is so the call ends up in exactly the same failure state as today (just a bit later), never
+     * left hanging.
+     */
+    private getUserMediaWithRetry;
+    /**
+     * Disables and removes all local streams for this session
+     * @param session - The session (or any object that contains localStreams that will be removed)
+     */
+    disableAllStreamsForSession(session: any): void;
+    /**
+     * Disables MediaStream
+     * @param streams - An array of MediaStream to be stopped and removed
+     */
+    disableMediaStreams(streams: MediaStream[]): void;
+    /**
+     * Disables MediaStream
+     * @param stream - The MediaStream to be stopped and removed
+     */
+    disableMediaStream(stream: MediaStream): void;
+    getMobileMediaConstraints(type: string): any;
+    createOrUpdateStatisticsForCall(call: CallRB, isCallEnding?: boolean): Promise<void>;
+    updateStatisticsForCall(id: string, callId: string, callState: string, metrics: any): Promise<void>;
+    createStatisticsForSession(callId: string, peerId: string, callState: string): Promise<string>;
+    setWebrtcConfiguration(): Promise<void>;
+    /**
+     * Get the Media stream from the browser w/o any constraints (except for the selected current device);
+     * @param userMedia - An array of medias to get (accepted values inside the array are : "audio" and "video");
+     */
+    getBrowserMediaWithNoConstraints(userMedia: string[]): Promise<MediaStream>;
+    private getDefaultConstraintsForMedia;
+    private setIceConfig;
+    private manageIceServers;
+    private getIceConfig;
+    private setCompanyWebrtcConfig;
+    private applyCompanyConfigOverrides;
+    private setVideoQualityConfigurationForCompany;
+    private getCompanyWebrtcConfig;
+    /**
+     * For QA Tests Auto
+     * Set/reset an interval for generating WebRTC statistics that will be used instead of the default 5s interval.
+     * Set by default to 1000ms.
+     * Usage: rainbow.services.WebrtcConfigAndUtils.setWebrtcStatsInterval(intervalMs);
+     */
+    setWebrtcStatsInterval(interval?: number): void;
+    resetWebrtcStatsInterval(): void;
+}
+//# sourceMappingURL=webrtcConfigAndUtils.d.ts.map

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

@@ -0,0 +1 @@
+{"version":3,"file":"webrtcConfigAndUtils.d.ts","sourceRoot":"","sources":["../../../src/services/webrtc/webrtcConfigAndUtils.ts"],"names":[],"mappings":"AACA,OAAO,KAAK,EAAE,YAAY,EAAE,MAAM,MAAM,CAAC;AAEzC,OAAO,KAAK,EAAE,MAAM,EAAE,MAAM,wCAAwC,CAAC;AAWrE,OAAO,EAAE,OAAO,EAAE,MAAM,uCAAuC,CAAC;AAQhE,eAAO,MAAM,uBAAuB,yBAAyB,CAAC;AAa9D,MAAM,WAAW,qBAAqB;IAClC,GAAG,EAAE,GAAG,CAAC;CACZ;AAYD;;;;GAIG;AAEH,qBAAa,oBAAqB,SAAQ,OAAQ,YAAW,qBAAqB;IAC9E;;OAEG;IACI,YAAY,EAAE,GAAG,CAAC;IAEzB;;OAEG;IACI,aAAa,EAAE,GAAG,CAAC;IAE1B,4FAA4F;IACrF,sBAAsB,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAM;IAExD;;OAEG;IACI,+BAA+B;;;;;;;;;;;MAWpC;IAEF;;OAEG;IACI,yBAAyB;;;;;;;;;;;MAW9B;IAEF;;;OAGG;IACI,GAAG,EAAE,GAAG,CAMb;IAEF;;;;;OAKG;IACI,SAAS,CAAC,OAAO,EAAE,GAAG,GAAG,YAAY;IAM5C;;;;OAIG;IACH,OAAO,CAAC,SAAS;IAIjB;;OAEG;IACI,UAAU,KAAM;IAEhB,eAAe,CAAC,gBAAgB,EAAE,MAAM,EAAE,mBAAmB,GAAE,OAAe;IAIrF,OAAO,CAAC,MAAM,CAA8C;IAC5D,OAAO,CAAC,eAAe,CAAkD;IACzE,OAAO,CAAC,WAAW,CAA0C;IAC7D,OAAO,CAAC,WAAW,CAA0C;IAC7D,OAAO,CAAC,WAAW,CAA0C;IAC7D,OAAO,CAAC,cAAc,CAAgD;IACtE,OAAO,CAAC,YAAY,CAA4C;IAChE,OAAO,CAAC,uBAAuB,CAA4B;IAC3D,OAAO,CAAC,cAAc,CAAgD;IACtE,OAAO,CAAC,SAAS,CAAoC;IAErD,OAAO,CAAC,kBAAkB,CAA2B;IACrD,OAAO,CAAC,mBAAmB,CAUzB;IAEF,OAAO,CAAC,GAAG,CAAa;IACxB,OAAO,CAAC,UAAU,CAAa;IAE/B,OAAO,CAAC,OAAO,CAAkB;IAEjC,OAAO,CAAC,iBAAiB,CAA6B;WAExC,WAAW;WAIX,KAAK;IAInB,OAAO;IAMM,KAAK;IAuBX,IAAI;YAYG,4BAA4B;IAyB1C,OAAO,CAAC,eAAe;IAmDvB,OAAO,CAAC,iBAAiB;IAUzB,OAAO,CAAC,uBAAuB;IAc/B;;OAEG;YACW,wBAAwB;IA6ItC,OAAO,CAAC,0BAA0B;IAM3B,uBAAuB;YAIhB,0BAA0B;IAyCjC,uBAAuB;IAqKvB,uBAAuB;IAKvB,iBAAiB;IAIxB,OAAO,CAAC,sBAAsB;IAuMjB,eAAe,CAAC,SAAS,EAAE,GAAG,EAAE,EAAE,eAAe,GAAE,MAAU,EAAE,gBAAgB,GAAE,MAAU,EAAE,cAAc,GAAE,MAAU,EAAE,OAAO,GAAE,GAAU,EAAE,kBAAkB,GAAE,OAAe,EAAE,iBAAiB,GAAE,OAAe,GAAG,OAAO,CAAC,WAAW,GAAG,SAAS,CAAC;IAwGrQ;;;;;;;;;;;;OAYG;YACW,qBAAqB;IAiBnC;;;OAGG;IACI,2BAA2B,CAAC,OAAO,EAAE,GAAG;IA0B/C;;;OAGG;IACI,mBAAmB,CAAC,OAAO,EAAE,WAAW,EAAE,GAAG,IAAI;IAgBxD;;;OAGG;IACI,kBAAkB,CAAC,MAAM,EAAE,WAAW,GAAG,IAAI;IAa7C,yBAAyB,CAAC,IAAI,EAAE,MAAM;IAIhC,+BAA+B,CAAC,IAAI,EAAE,MAAM,EAAE,YAAY,GAAE,OAAe;IAsD3E,uBAAuB,CAAC,EAAE,EAAE,MAAM,EAAE,MAAM,EAAE,MAAM,EAAE,SAAS,EAAE,MAAM,EAAE,OAAO,EAAE,GAAG,GAAG,OAAO,CAAC,IAAI,CAAC;IA2CnG,0BAA0B,CAAC,MAAM,EAAE,MAAM,EAAE,MAAM,EAAE,MAAM,EAAE,SAAS,EAAE,MAAM,GAAG,OAAO,CAAC,MAAM,CAAC;IA+B9F,sBAAsB;IAQnC;;;OAGG;IACU,gCAAgC,CAAC,SAAS,EAAE,MAAM,EAAE,GAAG,OAAO,CAAC,WAAW,CAAC;IA2BxF,OAAO,CAAC,6BAA6B;YAuCvB,YAAY;IAiF1B,OAAO,CAAC,gBAAgB;YAoBV,YAAY;YA+CZ,sBAAsB;IAqBpC,OAAO,CAAC,2BAA2B;IAUnC,OAAO,CAAC,sCAAsC;YAchC,sBAAsB;IAqBpC;;;;;OAKG;IACI,sBAAsB,CAAC,QAAQ,GAAE,MAAa;IAQ9C,wBAAwB;CAGlC"}