@wazo/sdk 0.38.22 → 0.39.0-beta.1

package/src/simple/Phone.ts

@@ -0,0 +1,623 @@
+import { SessionState } from 'sip.js/lib/api/session-state';
+import { Inviter, Invitation } from 'sip.js/lib/api';
+import { OutgoingInviteRequest } from 'sip.js/lib/core';
+import type SipLine from '../domain/SipLine';
+import type Session from '../domain/Session';
+import type CallSession from '../domain/CallSession';
+import AdHocAPIConference from '../domain/AdHocAPIConference';
+import WebRTCPhone, * as PHONE_EVENTS from '../domain/Phone/WebRTCPhone';
+import { MESSAGE_TYPE_CHAT, MESSAGE_TYPE_SIGNAL } from '../domain/Phone/WebRTCPhone';
+import WazoWebRTCClient, { events as clientEvents, transportEvents } from '../web-rtc-client';
+import IssueReporter from '../service/IssueReporter';
+import Emitter, { IEmitter } from '../utils/Emitter';
+import Wazo from './index';
+import SFUNotAvailableError from '../domain/SFUNotAvailableError';
+
+const logger = IssueReporter.loggerFor('simple-phone');
+const sipLogger = IssueReporter.loggerFor('sip.js');
+const protocolLogger = IssueReporter.loggerFor('sip');
+const protocolDebugMessages = ['Received WebSocket text message:', 'Sending WebSocket message:'];
+
+export interface IPhone extends IEmitter {
+  client: WazoWebRTCClient;
+  phone: WebRTCPhone | null | undefined;
+  session: Session;
+  sipLine: SipLine | null | undefined;
+  SessionState: Record<string, any>;
+
+  ON_USER_AGENT: string;
+  ON_REGISTERED: string;
+  ON_UNREGISTERED: string;
+  ON_PROGRESS: string;
+  ON_CALL_ACCEPTED: string;
+  ON_CALL_ANSWERED: string;
+  ON_CALL_INCOMING: string;
+  ON_CALL_OUTGOING: string;
+  ON_CALL_MUTED: string;
+  ON_CALL_UNMUTED: string;
+  ON_CALL_RESUMED: string;
+  ON_CALL_HELD: string;
+  ON_CALL_UNHELD: string;
+  ON_CAMERA_DISABLED: string;
+  ON_CAMERA_RESUMED: string;
+  ON_CALL_CANCELED: string;
+  ON_CALL_FAILED: string;
+  ON_CALL_REJECTED: string;
+  ON_CALL_ENDED: string;
+  ON_CALL_ENDING: string;
+  ON_MESSAGE: string;
+  ON_REINVITE: string;
+  ON_TRACK: string;
+  ON_AUDIO_STREAM: string;
+  ON_VIDEO_STREAM: string;
+  ON_REMOVE_STREAM: string;
+  ON_SHARE_SCREEN_STARTED: string;
+  ON_SHARE_SCREEN_ENDING: string;
+  ON_SHARE_SCREEN_ENDED: string;
+  ON_TERMINATE_SOUND: string;
+  ON_PLAY_RING_SOUND: string;
+  ON_PLAY_INBOUND_CALL_SIGNAL_SOUND: string;
+  ON_PLAY_HANGUP_SOUND: string;
+  ON_PLAY_PROGRESS_SOUND: string;
+  ON_VIDEO_INPUT_CHANGE: string;
+  ON_CALL_ERROR: string;
+  ON_MESSAGE_TRACK_UPDATED: string;
+  ON_NETWORK_STATS: string;
+  ON_CHAT: string;
+  ON_SIGNAL: string;
+  ON_DISCONNECTED: string;
+  ON_EARLY_MEDIA: string;
+  MESSAGE_TYPE_CHAT: string;
+  MESSAGE_TYPE_SIGNAL: string;
+
+  connect: (options: Record<string, any>, sipLine?: SipLine | null | undefined) => Promise<void>;
+  connectWithCredentials: (server: string, sipLine: SipLine, displayName: string, rawOptions: Record<string, any>) => void;
+  disconnect: () => Promise<void>;
+  call: (extension: string, withCamera: boolean, rawSipLine: SipLine | null | undefined, audioOnly: boolean, conference: boolean) => Promise<CallSession | null | undefined>;
+  hangup: (callSession: CallSession) => Promise<boolean> ;
+  startConference: (host: CallSession, otherCalls: CallSession[]) => Promise<AdHocAPIConference>;
+  mute: (callSession: CallSession, withApi?: boolean) => void;
+  unmute: (callSession: CallSession, withApi?: boolean) => void;
+  muteViaAPI: (callSession: CallSession) => void;
+  unmuteViaAPI: (callSession: CallSession) => void;
+  hold: (callSession: CallSession) => Promise<void | OutgoingInviteRequest> | null | undefined;
+  unhold: (callSession: CallSession) => Promise<MediaStream | void | null | undefined>;
+  resume: (callSession: CallSession) => Promise<MediaStream | null | void>;
+  reject: (callSession: CallSession) => Promise<void> | undefined;
+  transfer: (callSession: CallSession, target: string) => void;
+  atxfer: (callSession: CallSession) => Record<string, any> | null;
+  reinvite: (callSession: CallSession, constraints: (Record<string, any> | null), conference: boolean) => Promise<OutgoingInviteRequest | void | null>;
+  getStats: (callSession: CallSession) => Promise<RTCStatsReport | null | undefined>;
+  startNetworkMonitoring: (callSession: CallSession, interval: number) => void;
+  stopNetworkMonitoring: (callSession: CallSession) => void;
+  getSipSessionId: (sipSession: Invitation | Inviter) => string | null | undefined;
+  sendMessage: (body: string, sipSession?: Inviter | Invitation, contentType?: string) => void;
+  sendChat: (content: string, sipSession?: Inviter | Invitation) => void;
+  sendSignal: (content: any, sipSession?: Inviter | Invitation) => void ;
+  turnCameraOff: (callSession: CallSession) => void;
+  turnCameraOn: (callSession: CallSession) => void;
+  startScreenSharing: (constraints: Record<string, any>, callSession?: CallSession) => Promise<MediaStream | null>;
+  stopScreenSharing: (callSession?: CallSession, restoreLocalStream?: boolean) => Promise<OutgoingInviteRequest | void>;
+  sendDTMF: (tone: string, callSession: CallSession) => void;
+  getLocalStream: (callSession: CallSession) => MediaStream | null | undefined ;
+  hasLocalVideo: (callSession: CallSession) => boolean;
+  hasALocalVideoTrack: (callSession: CallSession) => boolean;
+  getLocalMediaStream: (callSession: CallSession) => MediaStream | null | undefined;
+  getLocalVideoStream: (callSession: CallSession) => MediaStream | null;
+  getRemoteStream: (callSession: CallSession) => MediaStream | null;
+  getRemoteVideoStream: (callSession: CallSession) => MediaStream | null ;
+  isVideoRemotelyHeld: (callSession: CallSession) => boolean;
+  getRemoteStreamForCall: (callSession: CallSession) => MediaStream | null;
+  getRemoteStreamsForCall: (callSession: CallSession) => MediaStream | null;
+  getRemoteVideoStreamForCall: (callSession: CallSession) => MediaStream | null;
+  getRemoteVideoStreamFromPc: (callSession: CallSession) => MediaStream | null;
+  hasVideo: (callSession: CallSession) => boolean;
+  hasAVideoTrack: (callSession: CallSession) => boolean;
+  getCurrentSipSession: () => Invitation | Inviter | null;
+  getPrimaryWebRtcLine: () => SipLine | null;
+  getOutputDevice: () => string | null ;
+  getPrimaryLine: () => SipLine | null;
+  getLineById: (lineId: string) => SipLine | null;
+  getSipLines: () => SipLine[];
+  hasSfu: () => boolean;
+  checkSfu: () => void;
+  _transferEvents: () => void;
+}
+
+class Phone extends Emitter implements IPhone {
+  client: WazoWebRTCClient;
+
+  phone: WebRTCPhone | null | undefined;
+
+  session: Session;
+
+  sipLine: SipLine | null | undefined;
+
+  SessionState: Record<string, any>;
+
+  ON_USER_AGENT: string;
+
+  ON_REGISTERED: string;
+
+  ON_UNREGISTERED: string;
+
+  ON_PROGRESS: string;
+
+  ON_CALL_ACCEPTED: string;
+
+  ON_CALL_ANSWERED: string;
+
+  ON_CALL_INCOMING: string;
+
+  ON_CALL_OUTGOING: string;
+
+  ON_CALL_MUTED: string;
+
+  ON_CALL_UNMUTED: string;
+
+  ON_CALL_RESUMED: string;
+
+  ON_CALL_HELD: string;
+
+  ON_CALL_UNHELD: string;
+
+  ON_CAMERA_DISABLED: string;
+
+  ON_CAMERA_RESUMED: string;
+
+  ON_CALL_CANCELED: string;
+
+  ON_CALL_FAILED: string;
+
+  ON_CALL_REJECTED: string;
+
+  ON_CALL_ENDED: string;
+
+  ON_CALL_ENDING: string;
+
+  ON_MESSAGE: string;
+
+  ON_REINVITE: string;
+
+  ON_TRACK: string;
+
+  ON_AUDIO_STREAM: string;
+
+  ON_VIDEO_STREAM: string;
+
+  ON_REMOVE_STREAM: string;
+
+  ON_SHARE_SCREEN_STARTED: string;
+
+  ON_SHARE_SCREEN_ENDING: string;
+
+  ON_SHARE_SCREEN_ENDED: string;
+
+  ON_TERMINATE_SOUND: string;
+
+  ON_PLAY_RING_SOUND: string;
+
+  ON_PLAY_INBOUND_CALL_SIGNAL_SOUND: string;
+
+  ON_PLAY_HANGUP_SOUND: string;
+
+  ON_PLAY_PROGRESS_SOUND: string;
+
+  ON_VIDEO_INPUT_CHANGE: string;
+
+  ON_CALL_ERROR: string;
+
+  ON_MESSAGE_TRACK_UPDATED: string;
+
+  ON_NETWORK_STATS: string;
+
+  ON_CHAT: string;
+
+  ON_SIGNAL: string;
+
+  ON_DISCONNECTED: string;
+
+  ON_EARLY_MEDIA: string;
+
+  MESSAGE_TYPE_CHAT: string;
+
+  MESSAGE_TYPE_SIGNAL: string;
+
+  constructor() {
+    super();
+    // Sugar syntax for `Wazo.Phone.EVENT_NAME`
+    Object.keys(PHONE_EVENTS).forEach((key: string) => {
+      // @ts-ignore
+      this[key] = PHONE_EVENTS[key];
+    });
+    this.SessionState = SessionState;
+  }
+
+  async connect(options: Record<string, any> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
+    if (this.phone) {
+      // Already connected
+      // let's update media constraints if they're being fed
+      if (options.media) {
+        this.phone.setMediaConstraints(options.media);
+      }
+
+      return;
+    }
+
+    const server = Wazo.Auth.getHost();
+    const session = Wazo.Auth.getSession();
+
+    if (!server || !session) {
+      throw new Error('Please connect to the server using `Wazo.Auth.logIn` or `Wazo.Auth.authenticate` before using Room.connect().');
+    }
+
+    this.session = session;
+    this.sipLine = sipLine || this.getPrimaryWebRtcLine();
+
+    if (!this.sipLine) {
+      throw new Error('Sorry, no sip lines found for this user');
+    }
+
+    this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
+  }
+
+  connectWithCredentials(server: string, sipLine: SipLine, displayName: string, rawOptions: Record<string, any> = {}): void {
+    if (this.phone) {
+      // Already connected
+      return;
+    }
+
+    const [host, port = 443] = server.split(':');
+    const options = rawOptions;
+    options.media = options.media || {
+      audio: true,
+      video: false,
+    };
+    options.uaConfigOverrides = options.uaConfigOverrides || {};
+
+    if (IssueReporter.enabled) {
+      options.uaConfigOverrides.traceSip = true;
+      options.log = options.log || {};
+      options.log.builtinEnabled = false;
+      options.log.logLevel = 'debug';
+
+      options.log.connector = (level: any, className: string, label: any, content: string) => {
+        const protocolIndex = content && content.indexOf ? protocolDebugMessages.findIndex(prefix => content.indexOf(prefix) !== -1) : -1;
+
+        if (className === 'sip.Transport' && protocolIndex !== -1) {
+          const direction = protocolIndex === 0 ? 'receiving' : 'sending';
+          const message = content.replace(`${protocolDebugMessages[protocolIndex]}\n\n`, '').replace('\r\n', '\n');
+          protocolLogger.trace(message, {
+            className,
+            direction,
+          });
+        } else {
+          sipLogger.trace(content, {
+            className,
+          });
+        }
+      };
+    }
+
+    this.client = new WazoWebRTCClient({
+      host,
+      port: typeof port === 'string' ? parseInt(port, 10) : port,
+      displayName,
+      authorizationUser: sipLine.username,
+      password: sipLine.secret,
+      uri: `${sipLine.username}@${server}`,
+      ...options,
+    }, null, options.uaConfigOverrides);
+    this.phone = new WebRTCPhone(this.client, options.audioDeviceOutput, true, options.audioDeviceRing);
+
+    this._transferEvents();
+  }
+
+  async disconnect(): Promise<void> {
+    if (this.phone) {
+      if (this.phone.hasAnActiveCall()) {
+        logger.info('hangup call on disconnect');
+        await this.phone.hangup(null);
+      }
+
+      await this.phone.close();
+    }
+
+    this.phone = null;
+  }
+
+  // If audioOnly is set to true, all video stream will be deactivated, even remotes ones.
+  async call(extension: string, withCamera = false, rawSipLine: SipLine | null | undefined = null, audioOnly = false, conference = false): Promise<CallSession | null | undefined> {
+    if (!this.phone) {
+      return;
+    }
+
+    const sipLine = rawSipLine || this.getPrimaryWebRtcLine();
+    return this.phone.makeCall(extension, sipLine, withCamera, audioOnly, conference);
+  }
+
+  async hangup(callSession: CallSession): Promise<boolean> {
+    logger.info('hangup via simple phone', {
+      callId: callSession.getId(),
+    });
+    return this.phone ? this.phone.hangup(callSession) : false;
+  }
+
+  async accept(callSession: CallSession, cameraEnabled?: boolean): Promise<string | null> {
+    logger.info('accept via simple phone', {
+      callId: callSession.getId(),
+      cameraEnabled,
+    });
+    return this.phone ? this.phone.accept(callSession, cameraEnabled) : null;
+  }
+
+  async startConference(host: CallSession, otherCalls: CallSession[]): Promise<AdHocAPIConference> {
+    const participants = [host, ...otherCalls].reduce((acc: Record<string, any>, participant: CallSession) => {
+      acc[participant.getTalkingToIds()[0]] = participant;
+      return acc;
+    }, {});
+
+    if (!this.phone) {
+      return Promise.reject();
+    }
+
+    const adHocConference = new AdHocAPIConference({
+      phone: this.phone,
+      host,
+      participants,
+    });
+    return adHocConference.start();
+  }
+
+  mute(callSession: CallSession, withApi = true): void {
+    if (withApi) {
+      this.muteViaAPI(callSession);
+    }
+
+    this.phone?.mute(callSession);
+  }
+
+  unmute(callSession: CallSession, withApi = true): void {
+    if (withApi) {
+      this.unmuteViaAPI(callSession);
+    }
+
+    this.phone?.unmute(callSession);
+  }
+
+  muteViaAPI(callSession: CallSession): void {
+    if (callSession && callSession.callId) {
+      Wazo.getApiClient().calld.mute(callSession.callId).catch(e => {
+        logger.error('Mute via API, error', e);
+      });
+    }
+  }
+
+  unmuteViaAPI(callSession: CallSession): void {
+    if (callSession && callSession.callId) {
+      Wazo.getApiClient().calld.unmute(callSession.callId).catch(e => {
+        logger.error('Unmute via API, error', e);
+      });
+    }
+  }
+
+  hold(callSession: CallSession): Promise<void | OutgoingInviteRequest> | null | undefined {
+    return this.phone?.hold(callSession, true);
+  }
+
+  async unhold(callSession: CallSession): Promise<MediaStream | void | null | undefined> {
+    return this.phone?.unhold(callSession, true);
+  }
+
+  async resume(callSession: CallSession): Promise<MediaStream | null | void> {
+    return this.phone?.resume(callSession) || null;
+  }
+
+  reject(callSession: CallSession): Promise<void> | undefined {
+    return this.phone?.reject(callSession);
+  }
+
+  transfer(callSession: CallSession, target: string): void {
+    this.phone?.transfer(callSession, target);
+  }
+
+  atxfer(callSession: CallSession): Record<string, any> | null {
+    return this.phone?.atxfer(callSession) || null;
+  }
+
+  async reinvite(callSession: CallSession, constraints: (Record<string, any> | null) = null, conference = false): Promise<OutgoingInviteRequest | void | null> {
+    return this.phone ? this.phone.sendReinvite(callSession, constraints, conference) : null;
+  }
+
+  async getStats(callSession: CallSession): Promise<RTCStatsReport | null | undefined> {
+    return this.phone ? this.phone.getStats(callSession) : null;
+  }
+
+  startNetworkMonitoring(callSession: CallSession, interval = 1000) {
+    return this.phone ? this.phone.startNetworkMonitoring(callSession, interval) : null;
+  }
+
+  stopNetworkMonitoring(callSession: CallSession) {
+    return this.phone ? this.phone.stopNetworkMonitoring(callSession) : null;
+  }
+
+  getSipSessionId(sipSession: Invitation | Inviter): string | null | undefined {
+    if (!sipSession || !this.phone) {
+      return null;
+    }
+    return this.phone.getSipSessionId(sipSession);
+  }
+
+  sendMessage(body: string, sipSession?: Inviter | Invitation, contentType = 'text/plain'): void {
+    const toSipSession = sipSession || this.getCurrentSipSession();
+
+    if (!toSipSession || !this.phone) {
+      return;
+    }
+
+    this.phone.sendMessage(toSipSession, body, contentType);
+  }
+
+  sendChat(content: string, sipSession?: Inviter | Invitation): void {
+    this.sendMessage(JSON.stringify({
+      type: MESSAGE_TYPE_CHAT,
+      content,
+    }), sipSession, 'application/json');
+  }
+
+  sendSignal(content: any, sipSession?: Inviter | Invitation): void {
+    this.sendMessage(JSON.stringify({
+      type: MESSAGE_TYPE_SIGNAL,
+      content,
+    }), sipSession, 'application/json');
+  }
+
+  turnCameraOff(callSession: CallSession): void {
+    this.phone?.turnCameraOff(callSession);
+  }
+
+  turnCameraOn(callSession: CallSession): void {
+    this.phone?.turnCameraOn(callSession);
+  }
+
+  async startScreenSharing(constraints: Record<string, any>, callSession?: CallSession): Promise<MediaStream | null> {
+    return this.phone?.startScreenSharing(constraints, callSession) || null;
+  }
+
+  stopScreenSharing(callSession?: CallSession, restoreLocalStream = true): Promise<OutgoingInviteRequest | void> {
+    return this.phone ? this.phone.stopScreenSharing(restoreLocalStream, callSession) : Promise.resolve();
+  }
+
+  sendDTMF(tone: string, callSession: CallSession): void {
+    this.phone?.sendKey(callSession, tone);
+  }
+
+  getLocalStream(callSession: CallSession): MediaStream | null | undefined {
+    return this.phone?.getLocalStream(callSession);
+  }
+
+  hasLocalVideo(callSession: CallSession): boolean {
+    return this.phone?.hasLocalVideo(callSession) || false;
+  }
+
+  hasALocalVideoTrack(callSession: CallSession): boolean {
+    return this.phone?.hasALocalVideoTrack(callSession) || false;
+  }
+
+  // @Deprecated
+  getLocalMediaStream(callSession: CallSession): MediaStream | null | undefined {
+    logger.warn('Phone.getLocalMediaStream is deprecated, use Phone.getLocalStream instead');
+    return this.phone?.getLocalStream(callSession);
+  }
+
+  getLocalVideoStream(callSession: CallSession): MediaStream | null {
+    return this.phone?.getLocalVideoStream(callSession) || null;
+  }
+
+  getRemoteStream(callSession: CallSession): MediaStream | null {
+    return this.phone?.getRemoteStream(callSession) || null;
+  }
+
+  getRemoteVideoStream(callSession: CallSession): MediaStream | null {
+    return this.phone?.getRemoteVideoStream(callSession) || null;
+  }
+
+  isVideoRemotelyHeld(callSession: CallSession): boolean {
+    return this.phone?.isVideoRemotelyHeld(callSession) || false;
+  }
+
+  // @Deprecated
+  getRemoteStreamForCall(callSession: CallSession): MediaStream | null {
+    logger.warn('Phone.getRemoteStreamForCall is deprecated, use Phone.getRemoteStream instead');
+    return this.getRemoteStream(callSession) || null;
+  }
+
+  // Returns remote streams directly from the peerConnection
+  // @Deprecated
+  getRemoteStreamsForCall(callSession: CallSession): MediaStream | null {
+    logger.warn('Phone.getRemoteStreamsForCall is deprecated, use Phone.getLocalStream instead');
+    return this.getLocalStream(callSession) || null;
+  }
+
+  // @Deprecated
+  getRemoteVideoStreamForCall(callSession: CallSession): MediaStream | null {
+    logger.warn('Phone.getRemoteVideoStreamForCall is deprecated, use Phone.getRemoteVideoStream instead');
+    return this.getRemoteVideoStream(callSession);
+  }
+
+  //  Useful in a react-native environment when remoteMediaStream is not updated
+  getRemoteVideoStreamFromPc(callSession: CallSession) : MediaStream | null {
+    return this.phone?.getRemoteVideoStreamFromPc(callSession) || null;
+  }
+
+  hasVideo(callSession: CallSession): boolean {
+    return this.phone ? this.phone.hasVideo(callSession) : false;
+  }
+
+  hasAVideoTrack(callSession: CallSession): boolean {
+    return this.phone ? this.phone.hasAVideoTrack(callSession) : false;
+  }
+
+  getCurrentSipSession(): Invitation | Inviter | null {
+    return this.phone?.currentSipSession || null;
+  }
+
+  getPrimaryWebRtcLine(): SipLine | null {
+    const session = Wazo.Auth.getSession();
+    return session?.primaryWebRtcLine() || null;
+  }
+
+  getOutputDevice(): string | null {
+    return this.phone?.audioOutputDeviceId || null;
+  }
+
+  getPrimaryLine(): SipLine | null {
+    const session = Wazo.Auth.getSession();
+    return session?.primarySipLine() || null;
+  }
+
+  getLineById(lineId: string): SipLine | null {
+    return this.getSipLines().find(line => line && line.id === lineId) || null;
+  }
+
+  getSipLines(): SipLine[] {
+    const session = Wazo.Auth.getSession();
+
+    if (!session) {
+      return [];
+    }
+
+    return session.profile?.sipLines || [];
+  }
+
+  hasSfu(): boolean {
+    return this.sipLine?.hasVideoConference() || false;
+  }
+
+  checkSfu(): void {
+    if (!this.hasSfu()) {
+      throw new SFUNotAvailableError();
+    }
+  }
+
+  _transferEvents() {
+    this.unbind();
+    [...clientEvents, ...transportEvents].forEach(event => {
+      this.client.on(event, (...args) => this.eventEmitter.emit.apply(this.eventEmitter.emit, [`client-${event}`, ...args]));
+    });
+    Object.values(PHONE_EVENTS).forEach(event => {
+      if (typeof event !== 'string' || !this.phone) {
+        return;
+      }
+
+      this.phone.on(event, (...args) => this.eventEmitter.emit.apply(this.eventEmitter, [event, ...args]));
+    });
+  }
+
+}
+
+if (!global.wazoTelephonyInstance) {
+  global.wazoTelephonyInstance = new Phone();
+}
+
+export default global.wazoTelephonyInstance;