@wazo/sdk 0.42.0 → 0.43.0

package/src/domain/Phone/Phone.ts

@@ -31,7 +31,7 @@ export type AvailablePhoneOptions = {
 };
 export interface Phone extends IEmitter {
   accept(callSession: CallSession, enableVideo?: boolean): Promise<string | null>;
-  changeAudioDevice(id: string): PhoneVoid;
+  changeAudioDevice(id: string): Promise<void>;
   changeRingDevice(id: string): PhoneVoid;
   changeAudioVolume(volume: number): PhoneVoid;
   changeRingVolume(volume: number): PhoneVoid;

package/src/domain/Phone/WebRTCPhone.ts

@@ -612,12 +612,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return callSession;
   }
 
-  changeAudioDevice(id: string) {
+  async changeAudioDevice(id: string) {
     logger.info('WebRTC phone - change audio device', {
       deviceId: id,
     });
     this.audioOutputDeviceId = id;
-    this.client.changeAudioOutputDevice(id);
+    return this.client.changeAudioOutputDevice(id);
   }
 
   createAudioElementFor(sessionId: string) {
@@ -1528,7 +1528,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
       }, 2000);
     });
-    this.client.on(this.client.ACCEPTED, (sipSession: WazoSession) => {
+    this.client.on(this.client.ACCEPTED, async (sipSession: WazoSession) => {
       const sessionId = this.getSipSessionId(sipSession);
       const hasSession = (sessionId in this.callSessions);
       logger.info('WebRTC call accepted', {
@@ -1546,7 +1546,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
 
       if (this.audioOutputDeviceId) {
-        this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
+        await this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
       }
     });
     this.client.on('ended', () => {});

package/src/domain/Profile.ts

@@ -2,6 +2,7 @@ import Line, { LineResponse } from './Line';
 import ForwardOption, { FORWARD_KEYS } from './ForwardOption';
 import newFrom from '../utils/new-from';
 import SipLine from './SipLine';
+import Incall, { GenericIncall } from './Incall';
 
 export const STATE = {
   AVAILABLE: 'available',
@@ -74,6 +75,7 @@ type ProfileResponse = {
     uuid: string;
   }>;
   online_call_record_enabled: boolean | null | undefined;
+  incalls: GenericIncall[];
 };
 
 type ProfileArguments = {
@@ -108,6 +110,7 @@ type ProfileArguments = {
     uuid: string;
   }>;
   onlineCallRecordEnabled?: boolean | null | undefined;
+  incalls?: GenericIncall[];
 };
 export default class Profile {
   id: string;
@@ -122,6 +125,8 @@ export default class Profile {
 
   sipLines: Array<SipLine>;
 
+  incalls: Array<Incall>;
+
   username: string;
 
   mobileNumber: string;
@@ -167,6 +172,7 @@ export default class Profile {
       lastName: plain.lastName || plain.lastname || '',
       email: plain.email,
       lines: plain.lines.map(line => Line.parse(line)),
+      incalls: plain.incalls,
       username: plain.username,
       mobileNumber: plain.mobile_phone_number || '',
       ringSeconds: plain.ring_seconds,
@@ -206,6 +212,7 @@ export default class Profile {
     sipLines,
     callPickupTargetUsers,
     onlineCallRecordEnabled,
+    incalls,
   }: ProfileArguments) {
     this.id = id;
     this.firstName = firstName;
@@ -226,6 +233,7 @@ export default class Profile {
     this.onlineCallRecordEnabled = onlineCallRecordEnabled;
     this.ringSeconds = ringSeconds;
     this.sipLines = sipLines || [];
+    this.incalls = incalls?.map(Incall.parse) || [];
   }
 
   static getLinesState(lines: Array<Record<string, any>>) {

package/src/domain/__tests__/Profile.test.ts

@@ -54,6 +54,21 @@ describe('Profile domain', () => {
         }],
         endpoint_custom: null,
       }],
+      incalls: [{
+        id: 1,
+        extensions: [{
+          id: 3,
+          exten: '8001112222',
+          context: 'incall',
+        }],
+      }, {
+        id: 2,
+        extensions: [{
+          id: 4,
+          exten: '8001113333',
+          context: 'incall',
+        }],
+      }],
       username: 'john.doe',
       services: {
         dnd: {
@@ -121,6 +136,21 @@ describe('Profile domain', () => {
           context: 'internal',
         }],
       })],
+      incalls: [{
+        id: 1,
+        extensions: [{
+          id: 3,
+          exten: '8001112222',
+          context: 'incall',
+        }],
+      }, {
+        id: 2,
+        extensions: [{
+          id: 4,
+          exten: '8001113333',
+          context: 'incall',
+        }],
+      }],
       mobileNumber: '',
       username: 'john.doe',
       doNotDisturb: false,
@@ -134,5 +164,7 @@ describe('Profile domain', () => {
       }],
       onlineCallRecordEnabled: true,
     }));
+    expect(profile.incalls[0].extension).toBe(plain.incalls[0].extensions[0].exten);
+    expect(profile.incalls[1].extension).toBe(plain.incalls[1].extensions[0].exten);
   });
 });

package/src/domain/types.ts

@@ -1,4 +1,9 @@
-import type { Invitation, Inviter, Session as SipSession, UserAgentOptions as sipJsUserAgentOptions } from 'sip.js';
+import type {
+  Invitation,
+  Inviter,
+  Session as SipSession,
+  UserAgentOptions as sipJsUserAgentOptions,
+} from 'sip.js';
 import { SessionDescriptionHandlerFactoryOptions } from 'sip.js/lib/platform/web';
 import type { IncomingResponse as SipIncomingResponse } from 'sip.js/lib/core/messages/incoming-response';
 import { Transport } from 'sip.js/lib/api';
@@ -6,10 +11,7 @@ import WazoSessionDescriptionHandler from '../lib/WazoSessionDescriptionHandler'
 import { Websocket } from '../simple/Websocket';
 import * as WebSocketClient from '../websocket-client';
 
-const {
-  SOCKET_EVENTS,
-  ...OTHER_EVENTS
-} = WebSocketClient;
+const { SOCKET_EVENTS, ...OTHER_EVENTS } = WebSocketClient;
 
 type GenericObject = Record<string, any>;
 
@@ -79,6 +81,12 @@ export type Link = {
   href: string;
   rel: string;
 };
+export type ExtensionRelation = {
+  id: number;
+  exten: string;
+  context: string;
+  links?: Link[];
+};
 type LocalLine = {
   id: number;
   endpoint_sip: {
@@ -88,12 +96,7 @@ type LocalLine = {
   };
   endpoint_sccp: string | null | undefined;
   endpoint_custom: string | null | undefined;
-  extensions: Array<{
-    id: number;
-    exten: string;
-    context: string;
-    links: Array<Link>;
-  }>;
+  extensions: ExtensionRelation[];
 };
 export type ConfdUser = {
   id: number;
@@ -169,7 +172,6 @@ export type ConfdUser = {
   // @TODO
   voicemail: string | null | undefined;
   queues: Array<Record<string, any>>; // @TODO
-
 };
 export type ListConfdUsersResponse = {
   total: number;
@@ -193,7 +195,6 @@ export type ListApplicationsResponse = {
 export type Node = {
   uuid: UUID;
   calls: Array<Record<string, any>>; // @TODO
-
 };
 export type CallNode = GenericObject;
 export type ListNodesResponse = {
@@ -206,21 +207,26 @@ export type ListCallNodesResponse = {
 export type GetTenantResponse = GenericObject;
 export type GetUserResponse = GenericObject;
 export type CTITransfer = {
-  'flow': string;
-  'id': string;
-  'initiator_call': string;
-  'initiator_uuid': string;
-  'recipient_call': string;
-  'status': string;
-  'transferred_call': string;
+  flow: string;
+  id: string;
+  initiator_call: string;
+  initiator_uuid: string;
+  recipient_call: string;
+  status: string;
+  transferred_call: string;
 };
 export type UserAgentOptions = sipJsUserAgentOptions & {
-  peerConnectionOptions?: Record<string, any>,
-  sessionDescriptionHandlerFactory: (session: SipSession, options: SessionDescriptionHandlerFactoryOptions) => WazoSessionDescriptionHandler,
+  peerConnectionOptions?: Record<string, any>;
+  sessionDescriptionHandlerFactory: (
+    session: SipSession,
+    options: SessionDescriptionHandlerFactoryOptions
+  ) => WazoSessionDescriptionHandler;
 };
-export type UserAgentConfigOverrides = Partial<UserAgentOptions & {
-  traceSip: any,
-}>;
+export type UserAgentConfigOverrides = Partial<
+UserAgentOptions & {
+  traceSip: any;
+}
+>;
 export type MediaConfig = {
   audio: Record<string, any> | boolean;
   video?: Record<string, any> | boolean;
@@ -237,7 +243,6 @@ export type WebRtcConfig = {
   media?: MediaConfig;
   iceCheckingTimeout?: number;
   log?: Record<string, any>;
-  audioOutputDeviceId?: string;
   audioOutputVolume?: number;
   userAgentString?: string;
   heartbeatDelay?: number;
@@ -245,31 +250,32 @@ export type WebRtcConfig = {
   maxHeartbeats?: number;
   skipRegister?: boolean;
   userUuid?: string,
+  uaConfigOverrides?: UserAgentConfigOverrides,
+  audioDeviceOutput?: string,
+  audioDeviceRing?: string
 }; // @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
 
-export type ConnectionOptions = WebRtcConfig & {
-  uaConfigOverrides: UserAgentConfigOverrides,
-  audioDeviceOutput: string,
-  audioDeviceRing: string
-};
 export type IncomingResponse = SipIncomingResponse & { session: any };
 
 export type PeerConnection = RTCPeerConnection & {
-  getRemoteStreams: () => MediaStream[],
-  getLocalStreams: () => MediaStream[],
-  onremovestream: (func: any) => void,
-  addStream: (stream: MediaStream) => void,
-  sfu: any
+  getRemoteStreams: () => MediaStream[];
+  getLocalStreams: () => MediaStream[];
+  onremovestream: (func: any) => void;
+  addStream: (stream: MediaStream) => void;
+  sfu: any;
 };
 
-export type WazoSession = (Invitation | Inviter) & { remoteTag?: any, callId?: string };
+export type WazoSession = (Invitation | Inviter) & {
+  remoteTag?: any;
+  callId?: string;
+};
 
 export type WazoTransport = Transport & {
-  configuration: Record<string, any>,
-  connectPromise: Promise<any>,
-  disconnectPromise: Promise<any>,
-  disconnectResolve: any,
-  transitionState: any,
+  configuration: Record<string, any>;
+  connectPromise: Promise<any>;
+  disconnectPromise: Promise<any>;
+  disconnectResolve: any;
+  transitionState: any;
   transitioningState: any;
   _ws: any;
 };
@@ -302,7 +308,7 @@ export type PresenceResponse = {
   state: string;
   status: string;
   user_uuid: string;
-  last_activity?: string,
+  last_activity?: string;
 };
 
 export type QueryParams = {
@@ -320,4 +326,6 @@ export type UuidSearchableQueryParams = SearchableQueryParams & {
   uuid?: string;
 };
 
-export type WebsocketType = Websocket & typeof SOCKET_EVENTS & typeof OTHER_EVENTS;
+export type WebsocketType = Websocket &
+  typeof SOCKET_EVENTS &
+  typeof OTHER_EVENTS;

package/src/index.ts

@@ -14,6 +14,7 @@ export { default as Contact } from './domain/Contact';
 export { default as COUNTRIES } from './domain/Country';
 export { default as Features } from './domain/Features';
 export { default as ForwardOption, FORWARD_KEYS } from './domain/ForwardOption';
+export { default as Incall } from './domain/Incall';
 export { default as Line } from './domain/Line';
 export { default as NotificationOptions } from './domain/NotificationOptions';
 export { default as Profile, STATE as PROFILE_STATE, LINE_STATE } from './domain/Profile';

package/src/simple/Auth.ts

@@ -160,6 +160,11 @@ export class Auth {
     return this._onAuthenticated(rawSession as Session);
   }
 
+  async samlLogIn(samlSessionId: string): Promise<Session | null> {
+    const rawSession = await getApiClient().auth.samlLogIn(samlSessionId);
+    return this._onAuthenticated(rawSession as Session);
+  }
+
   async logInViaRefreshToken(refreshToken: string): Promise<Session | null> {
     const rawSession = await getApiClient().auth.refreshToken(refreshToken, '', this.expiration, this.mobile);
     return this._onAuthenticated(rawSession as Session);

package/src/simple/Phone.ts

@@ -11,7 +11,7 @@ import IssueReporter from '../service/IssueReporter';
 import Emitter from '../utils/Emitter';
 import Wazo from './index';
 import SFUNotAvailableError from '../domain/SFUNotAvailableError';
-import { ConnectionOptions, WazoSession } from '../domain/types';
+import { WazoSession, WebRtcConfig } from '../domain/types';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -127,7 +127,7 @@ export class Phone extends Emitter {
     this.SessionState = SessionState;
   }
 
-  async connect(rawOptions: Partial<ConnectionOptions> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
+  async connect(rawOptions: Partial<WebRtcConfig> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
     const options = rawOptions;
     if (this.phone) {
       // Already connected
@@ -162,7 +162,7 @@ export class Phone extends Emitter {
     server: string,
     sipLine: SipLine,
     displayName: string,
-    rawOptions: Partial<ConnectionOptions> = {},
+    rawOptions: Partial<WebRtcConfig> = {},
   ): void {
     if (this.phone) {
       // Already connected
@@ -194,7 +194,7 @@ export class Phone extends Emitter {
       password: sipLine.secret,
       uri: `${sipLine.username}@${server}`,
       ...options,
-    }, null, options.uaConfigOverrides);
+    }, undefined, options.uaConfigOverrides);
     this.phone = new WebRTCPhone(this.client, options.audioDeviceOutput, true, options.audioDeviceRing);
 
     this._transferEvents();

package/src/simple/room/Room.ts

@@ -492,13 +492,13 @@ class Room extends Emitter {
     // Listen to REINVITE to ba able to map msid after upgrading to video in a  audio only conference
     // This allow to map msid with the non parsed (eg without the `stripVideo` modifier) SDP
     Wazo.Phone.phone.on(Wazo.Phone.phone.client.ON_REINVITE, this._boundOnReinvite);
-    this.on(this.ON_AUDIO_STREAM, stream => {
+    this.on(this.ON_AUDIO_STREAM, async stream => {
       logger.info('on room audio stream');
       this.audioStream = stream;
 
       if (!this.roomAudioElement) {
         const sessionId = Wazo.Phone.phone?.getSipSessionId(Wazo.Phone.phone?.currentSipSession as WazoSession);
-        this.roomAudioElement = Wazo.Phone.phone?.createAudioElementFor(sessionId as string) as HTMLAudioElement;
+        this.roomAudioElement = await Wazo.Phone.phone?.createAudioElementFor(sessionId as string) as HTMLAudioElement;
         this.roomAudioElement.srcObject = stream;
       } else {
         this.roomAudioElement.srcObject = stream;

package/src/web-rtc-client.ts

@@ -90,7 +90,7 @@ export default class WebRTCClient extends Emitter {
 
   audio: MediaTrackConstraintSet | boolean | undefined;
 
-  audioElements: Record<string, HTMLAudioElement & { setSinkId?: (id: string) => void }>;
+  audioElements: Record<string, HTMLAudioElement>;
 
   video: MediaTrackConstraintSet | boolean | undefined;
 
@@ -182,7 +182,7 @@ export default class WebRTCClient extends Emitter {
     return [];
   }
 
-  constructor(config: WebRtcConfig, session: WazoSession | null | undefined, uaConfigOverrides?: UserAgentConfigOverrides) {
+  constructor(config: WebRtcConfig, session?: WazoSession, uaConfigOverrides?: UserAgentConfigOverrides) {
     super();
 
     // For debug purpose
@@ -198,7 +198,7 @@ export default class WebRTCClient extends Emitter {
       this.userAgent = this.createUserAgent(uaConfigOverrides);
     });
 
-    this.audioOutputDeviceId = config.audioOutputDeviceId;
+    this.audioOutputDeviceId = config.audioDeviceOutput;
     this.audioOutputVolume = config.audioOutputVolume || 1;
 
     if (config.media) {
@@ -1246,20 +1246,14 @@ export default class WebRTCClient extends Emitter {
     this.audioOutputVolume = volume;
   }
 
-  changeAudioOutputDevice(id: string) {
+  async changeAudioOutputDevice(id: string) {
     logger.info('Changing audio output device', {
       id,
     });
     this.audioOutputDeviceId = id;
-    Object.values(this.audioElements).forEach(audioElement => {
-      // `setSinkId` method is not included in any flow type definitions for HTMLAudioElements but is a valid method
-      // audioElement is an array of HTMLAudioElements, and HTMLAudioElement inherits the method from HTMLMediaElement
-      // https://developer.mozilla.org/en-US/docs/Web/API/HTMLAudioElement
-      // https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/setSinkId
-      if (audioElement.setSinkId) {
-        audioElement.setSinkId(id);
-      }
-    });
+    const promises: Promise<void>[] = [];
+    Object.values(this.audioElements).forEach(audioElement => promises.push(audioElement.setSinkId(id)));
+    await Promise.allSettled(promises);
   }
 
   async changeAudioInputDevice(id: string, session: WazoSession | null | undefined, force?: boolean): Promise<MediaStream | null> {
@@ -1735,7 +1729,7 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  updateLocalStream(sipSessionId: string, newStream: MediaStream): void {
+  async updateLocalStream(sipSessionId: string, newStream: MediaStream): Promise<void> {
     const sipSession = this.sipSessions[sipSessionId];
 
     if (!sipSession) {
@@ -1756,7 +1750,7 @@ export default class WebRTCClient extends Emitter {
     this.setLocalMediaStream(sipSessionId, newStream);
 
     // Update the local video element
-    this._setupMedias(sipSession, newStream);
+    await this._setupMedias(sipSession, newStream);
   }
 
   updateRemoteStream(sessionId: string, allTracks = true): void {
@@ -1832,14 +1826,14 @@ export default class WebRTCClient extends Emitter {
     return sessionId in this.conferences;
   }
 
-  createAudioElementFor(sessionId: string): HTMLAudioElement {
+  async createAudioElementFor(sessionId: string): Promise<HTMLAudioElement> {
     const audio: HTMLAudioElement = document.createElement('audio');
     audio.setAttribute('id', `audio-${sessionId}`);
 
     logger.info('creating audio element', { sessionId, audioOutputDeviceId: this.audioOutputDeviceId });
 
-    if ((audio as any).setSinkId && this.audioOutputDeviceId) {
-      (audio as any).setSinkId(this.audioOutputDeviceId);
+    if (this.audioOutputDeviceId) {
+      await audio.setSinkId(this.audioOutputDeviceId);
     }
 
     if (document.body) {
@@ -2135,7 +2129,7 @@ export default class WebRTCClient extends Emitter {
       oldInviteRequest(request);
     };
 
-    session.delegate.onInvite = (request: IncomingRequestMessage) => {
+    session.delegate.onInvite = async (request: IncomingRequestMessage) => {
       let updatedCalleeName: string | null = null;
       let updatedNumber = null;
 
@@ -2165,14 +2159,14 @@ export default class WebRTCClient extends Emitter {
 
       this.updateRemoteStream(sipSessionId, false);
 
-      this._setupMedias(session);
+      await this._setupMedias(session);
 
       return this.eventEmitter.emit(ON_REINVITE, session, request, updatedCalleeName, updatedNumber, hadRemoteVideo);
     };
   }
 
-  _onProgress(session: WazoSession, early = false): void {
-    this._setupMedias(session);
+  async _onProgress(session: WazoSession, early = false): Promise<void> {
+    await this._setupMedias(session);
 
     const sessionId = this.getSipSessionId(session).substr(0, 20);
     this.updateRemoteStream(sessionId);
@@ -2183,7 +2177,7 @@ export default class WebRTCClient extends Emitter {
     this.eventEmitter.emit(early ? ON_EARLY_MEDIA : ON_PROGRESS, session);
   }
 
-  _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): void {
+  async _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): Promise<void> {
     logger.info('on call accepted', {
       id: session.id,
       clientId: this.clientId,
@@ -2191,7 +2185,7 @@ export default class WebRTCClient extends Emitter {
     });
     this.storeSipSession(session);
 
-    this._setupMedias(session);
+    await this._setupMedias(session);
 
     this.updateRemoteStream(this.getSipSessionId(session), initAllTracks);
     const pc = (session.sessionDescriptionHandler as SessionDescriptionHandler)?.peerConnection;
@@ -2253,7 +2247,7 @@ export default class WebRTCClient extends Emitter {
     return Boolean(session.sessionDescriptionHandlerModifiersReInvite.find(modifier => modifier === stripVideo));
   }
 
-  _setupMedias(session: WazoSession, newStream: MediaStream | null | undefined = null): void {
+  async _setupMedias(session: WazoSession, newStream: MediaStream | null | undefined = null): Promise<void> {
     if (!this._isWeb()) {
       logger.info('Setup media on mobile, no need to setup html element, bailing');
       return;
@@ -2271,7 +2265,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
-      this.createAudioElementFor(sessionId);
+      await this.createAudioElementFor(sessionId);
     }
 
     const audioElement = this.audioElements[sessionId];