@wazo/sdk 0.39.0-beta.13 → 0.39.0-beta.14

package/lib/types/web-rtc-client.d.ts

@@ -11,39 +11,16 @@ import { OutgoingByeRequest, OutgoingInviteRequest, OutgoingRequest } from 'sip.
 import { Inviter, Invitation, Registerer } from 'sip.js/lib/api';
 import Emitter from './utils/Emitter';
 import Heartbeat from './utils/Heartbeat';
+import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, UserAgentOptions } from './domain/types';
 export declare const replaceLocalIpModifier: (description: Record<string, any>) => Promise<any>;
 export declare const events: string[];
 export declare const transportEvents: string[];
 export declare class CanceledCallError extends Error {
 }
-type MediaConfig = {
-    audio: Record<string, any> | boolean;
-    video?: Record<string, any> | boolean;
-    localVideo?: Record<string, any> | boolean;
-};
-type WebRtcConfig = {
-    displayName?: string;
-    host: string;
-    port?: number;
-    websocketSip?: string;
-    authorizationUser?: string;
-    password?: string;
-    uri?: string;
-    media?: MediaConfig;
-    iceCheckingTimeout?: number;
-    log?: Record<string, any>;
-    audioOutputDeviceId?: string;
-    audioOutputVolume?: number;
-    userAgentString?: string;
-    heartbeatDelay?: number;
-    heartbeatTimeout?: number;
-    maxHeartbeats?: number;
-    skipRegister?: boolean;
-};
 export default class WebRTCClient extends Emitter {
     clientId: number;
     config: WebRtcConfig;
-    uaConfigOverrides: Record<string, any> | null | undefined;
+    uaConfigOverrides?: UserAgentConfigOverrides;
     userAgent: UserAgent | null;
     registerer: Registerer | null;
     hasAudio: boolean;
@@ -88,10 +65,10 @@ export default class WebRTCClient extends Emitter {
     static getIceServers(ip: string): Array<{
         urls: Array<string>;
     }>;
-    constructor(config: WebRtcConfig, session: Invitation | Inviter | null | undefined, uaConfigOverrides?: Record<string, any> | null | undefined);
+    constructor(config: WebRtcConfig, session: Invitation | Inviter | null | undefined, uaConfigOverrides?: UserAgentConfigOverrides);
     configureMedia(media: MediaConfig): void;
     setMediaConstraints(media: MediaStreamConstraints): void;
-    createUserAgent(configOverrides: Record<string, any> | null | undefined): UserAgent;
+    createUserAgent(uaConfigOverrides?: UserAgentConfigOverrides): UserAgent;
     isConnected(): boolean;
     isConnecting(): boolean;
     isRegistered(): boolean;
@@ -185,7 +162,7 @@ export default class WebRTCClient extends Emitter {
     _getVideoConstraints(video?: boolean): MediaTrackConstraints | boolean;
     _connectIfNeeded(): Promise<any>;
     _buildConfig(config: WebRtcConfig, session: Invitation | Inviter | null | undefined): Promise<WebRtcConfig>;
-    _createWebRTCConfiguration(configOverrides?: Record<string, any> | null): Record<string, any>;
+    _createUaOptions(uaOptionsOverrides?: UserAgentConfigOverrides): UserAgentOptions;
     _getRtcOptions(): {
         mandatory: {
             OfferToReceiveAudio: boolean;
@@ -216,5 +193,4 @@ export default class WebRTCClient extends Emitter {
     _cancel(session: Inviter, options?: InviterCancelOptions): Promise<void>;
     _bye(session: Invitation | Inviter, options?: SessionByeOptions): Promise<OutgoingByeRequest | null>;
 }
-export {};
 //# sourceMappingURL=web-rtc-client.d.ts.map

package/lib/types/web-rtc-client.d.ts.map

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package/lib/web-rtc-client.js

@@ -103,6 +103,8 @@ class CanceledCallError extends Error {
 }
 exports.CanceledCallError = CanceledCallError;
 const MAX_REGISTER_TRIES = 5;
+// setting a 24hr timeout and letting the backend define the actual value
+const NO_ANSWER_TIMEOUT = 60 * 60 * 24; // in seconds
 class WebRTCClient extends Emitter_1.default {
     static isAPrivateIp(ip) {
         const regex = /^(?:10|127|172\.(?:1[6-9]|2[0-9]|3[01])|192\.168)\..*/;
@@ -116,7 +118,7 @@ class WebRTCClient extends Emitter_1.default {
         }
         return [];
     }
-    constructor(config, session, uaConfigOverrides = undefined) {
+    constructor(config, session, uaConfigOverrides) {
         super();
         // For debug purpose
         this.clientId = Math.ceil(Math.random() * 1000);
@@ -177,10 +179,10 @@ class WebRTCClient extends Emitter_1.default {
         this.video = media.video;
         this.audio = media.audio;
     }
-    createUserAgent(configOverrides) {
-        const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
-        logger.info('sdk webrtc, creating UA', { webRTCConfiguration, clientId: this.clientId });
-        webRTCConfiguration.delegate = {
+    createUserAgent(uaConfigOverrides) {
+        const uaOptions = this._createUaOptions(uaConfigOverrides);
+        logger.info('sdk webrtc, creating UA', { uaOptions, clientId: this.clientId });
+        uaOptions.delegate = {
             onConnect: this.onConnect.bind(this),
             onDisconnect: this.onDisconnect.bind(this),
             onInvite: (invitation) => {
@@ -196,7 +198,7 @@ class WebRTCClient extends Emitter_1.default {
                 this.eventEmitter.emit(INVITE, invitation, this.sessionWantsToDoVideo(invitation), shouldAutoAnswer);
             },
         };
-        const ua = new user_agent_1.UserAgent(webRTCConfiguration);
+        const ua = new user_agent_1.UserAgent(uaOptions);
         if (ua.transport && ua.transport.connectPromise) {
             ua.transport.connectPromise.catch((e) => {
                 logger.warn('Transport connect error', e);
@@ -457,9 +459,7 @@ class WebRTCClient extends Emitter_1.default {
             requestDelegate: {
                 onAccept: (response) => {
                     var _a;
-                    // @ts-ignore
                     if ((_a = session.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.peerConnection) {
-                        // @ts-ignore
                         session.sessionDescriptionHandler.peerConnection.sfu = conference;
                     }
                     // @ts-ignore
@@ -1653,7 +1653,7 @@ class WebRTCClient extends Emitter_1.default {
         // @ts-ignore
         return client.confd.getUserLineSipFromToken(session.uuid).then(sipLine => (Object.assign({ authorizationUser: sipLine.username, password: sipLine.secret, uri: `${sipLine.username}@${config.host}` }, config)));
     }
-    _createWebRTCConfiguration(configOverrides = {}) {
+    _createUaOptions(uaOptionsOverrides = {}) {
         let { host, } = this.config;
         let port = this.config.port || 443;
         if (this.config.websocketSip) {
@@ -1661,7 +1661,8 @@ class WebRTCClient extends Emitter_1.default {
             [host] = webSocketSip;
             port = Number(webSocketSip[1]);
         }
-        const config = {
+        const uaOptions = {
+            noAnswerTimeout: NO_ANSWER_TIMEOUT,
             authorizationUsername: this.config.authorizationUser,
             authorizationPassword: this.config.password,
             displayName: this.config.displayName,
@@ -1686,7 +1687,7 @@ class WebRTCClient extends Emitter_1.default {
                 return new WazoSessionDescriptionHandler_1.default(uaLogger, WazoSessionDescriptionHandler_1.wazoMediaStreamFactory, sdhOptions, isWeb, session);
             },
             transportOptions: {
-                traceSip: (configOverrides === null || configOverrides === void 0 ? void 0 : configOverrides.traceSip) || false,
+                traceSip: (uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.traceSip) || false,
                 wsServers: `wss://${host}:${port}/api/asterisk/ws`,
             },
             sessionDescriptionHandlerFactoryOptions: {
@@ -1699,13 +1700,14 @@ class WebRTCClient extends Emitter_1.default {
                 peerConnectionOptions: {
                     iceCheckingTimeout: this.config.iceCheckingTimeout || 1000,
                     iceGatheringTimeout: this.config.iceCheckingTimeout || 1000,
-                    rtcConfiguration: Object.assign(Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, this._getRtcOptions()), ((configOverrides === null || configOverrides === void 0 ? void 0 : configOverrides.peerConnectionOptions) || {})),
+                    rtcConfiguration: Object.assign(Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, this._getRtcOptions()), ((uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.peerConnectionOptions) || {})),
                 },
                 // Configuration used in SDH to create the PeerConnection
-                peerConnectionConfiguration: Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, ((configOverrides === null || configOverrides === void 0 ? void 0 : configOverrides.peerConnectionOptions) || {})),
+                peerConnectionConfiguration: Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, ((uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.peerConnectionOptions) || {})),
             },
         };
-        return Object.assign(Object.assign({}, config), configOverrides);
+        uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? true : delete uaOptionsOverrides.traceSip;
+        return Object.assign(Object.assign({}, uaOptions), uaOptionsOverrides);
     }
     // eslint-disable-next-line no-unused-vars
     _getRtcOptions() {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.39.0-beta.13",
+  "version": "0.39.0-beta.14",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/domain/Session.ts

@@ -222,6 +222,7 @@ export default class Session {
       state: state || '',
       status,
       backend: 'wazo',
+      source: 'default',
     });
   }
 

package/src/domain/types.ts

@@ -1,3 +1,7 @@
+import type { Session, UserAgentOptions as sipJsUserAgentOptions } from 'sip.js';
+import { SessionDescriptionHandlerFactoryOptions } from 'sip.js/lib/platform/web';
+import WazoSessionDescriptionHandler from '../lib/WazoSessionDescriptionHandler';
+
 // @TODO: stand-in for empty object types. Was `type Something = {};` in JS
 type TodoObject = object;
 
@@ -202,20 +206,6 @@ export type ListCallNodesResponse = {
 };
 export type GetTenantResponse = TodoObject;
 export type GetUserResponse = TodoObject;
-export type WebRtcConfig = {
-  id: number;
-  host: string;
-  line: {
-    id: number;
-    links: Array<Link>;
-  };
-  links: Array<Link>;
-  options: Array<Array<string>>;
-  secret: string;
-  tenant_uuid: string;
-  type: string;
-  username: string;
-};
 export type CTITransfer = {
   'flow': string;
   'id': string;
@@ -225,3 +215,40 @@ export type CTITransfer = {
   'status': string;
   'transferred_call': string;
 };
+export type UserAgentOptions = sipJsUserAgentOptions & {
+  peerConnectionOptions?: Record<string, any>,
+  sessionDescriptionHandlerFactory: (session: Session, options: SessionDescriptionHandlerFactoryOptions) => WazoSessionDescriptionHandler,
+};
+export type UserAgentConfigOverrides = Partial<UserAgentOptions & {
+  traceSip: any,
+}>;
+export type MediaConfig = {
+  audio: Record<string, any> | boolean;
+  video?: Record<string, any> | boolean;
+  localVideo?: Record<string, any> | boolean;
+};
+export type WebRtcConfig = {
+  displayName?: string;
+  host: string;
+  port?: number;
+  websocketSip?: string;
+  authorizationUser?: string;
+  password?: string;
+  uri?: string;
+  media?: MediaConfig;
+  iceCheckingTimeout?: number;
+  log?: Record<string, any>;
+  audioOutputDeviceId?: string;
+  audioOutputVolume?: number;
+  userAgentString?: string;
+  heartbeatDelay?: number;
+  heartbeatTimeout?: number;
+  maxHeartbeats?: number;
+  skipRegister?: boolean;
+}; // @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
+
+export type ConnectionOptions = WebRtcConfig & {
+  uaConfigOverrides: UserAgentConfigOverrides,
+  audioDeviceOutput: string,
+  audioDeviceRing: string
+};

package/src/simple/Phone.ts

@@ -12,6 +12,7 @@ import IssueReporter from '../service/IssueReporter';
 import Emitter, { IEmitter } from '../utils/Emitter';
 import Wazo from './index';
 import SFUNotAvailableError from '../domain/SFUNotAvailableError';
+import { ConnectionOptions } from '../domain/types';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -70,8 +71,8 @@ export interface IPhone extends IEmitter {
   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;
+  connect: (options: Partial<ConnectionOptions>, sipLine?: SipLine | null | undefined) => Promise<void>;
+  connectWithCredentials: (server: string, sipLine: SipLine, displayName: string, rawOptions: Partial<ConnectionOptions>) => 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> ;
@@ -233,7 +234,7 @@ class Phone extends Emitter implements IPhone {
     this.SessionState = SessionState;
   }
 
-  async connect(options: Record<string, any> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
+  async connect(options: Partial<ConnectionOptions> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
     if (this.phone) {
       // Already connected
       // let's update media constraints if they're being fed
@@ -261,7 +262,12 @@ class Phone extends Emitter implements IPhone {
     this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
   }
 
-  connectWithCredentials(server: string, sipLine: SipLine, displayName: string, rawOptions: Record<string, any> = {}): void {
+  connectWithCredentials(
+    server: string,
+    sipLine: SipLine,
+    displayName: string,
+    rawOptions: Partial<ConnectionOptions> = {},
+  ): void {
     if (this.phone) {
       // Already connected
       return;

package/src/web-rtc-client.ts

@@ -8,7 +8,6 @@ import type { InvitationRejectOptions } from 'sip.js/lib/api/invitation-reject-o
 import type { InviterCancelOptions } from 'sip.js/lib/api/inviter-cancel-options';
 import type { SessionByeOptions } from 'sip.js/lib/api/session-bye-options';
 import type { InvitationAcceptOptions } from 'sip.js/lib/api/invitation-accept-options';
-
 import type { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
 import type { IncomingRequestMessage } from 'sip.js/lib/core/messages/incoming-request-message';
 import type { SessionDescriptionHandlerFactoryOptions } from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-factory-options';
@@ -29,7 +28,7 @@ import { defaultPeerConnectionConfiguration } from 'sip.js/lib/platform/web/sess
 import getStats from 'getstats';
 
 import { OutgoingByeRequest, OutgoingInviteRequest, OutgoingRequest } from 'sip.js/lib/core';
-import { Inviter, Invitation, Registerer } from 'sip.js/lib/api';
+import { Inviter, Invitation, Registerer, Session } from 'sip.js/lib/api';
 import WazoSessionDescriptionHandler, { wazoMediaStreamFactory } from './lib/WazoSessionDescriptionHandler';
 import Emitter from './utils/Emitter';
 import ApiClient from './api-client';
@@ -37,6 +36,7 @@ import IssueReporter from './service/IssueReporter';
 import Heartbeat from './utils/Heartbeat';
 import { getVideoDirection, hasAnActiveVideo } from './utils/sdp';
 import { lastIndexOf } from './utils/array';
+import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, UserAgentOptions } from './domain/types';
 
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
 export const replaceLocalIpModifier = (description: Record<string, any>) => Promise.resolve({ // description is immutable... so we have to clone it or the `type` attribute won't be returned.
@@ -70,38 +70,17 @@ const ON_DISCONNECTED = 'onDisconnected';
 export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {}
+
 const MAX_REGISTER_TRIES = 5;
-type MediaConfig = {
-  audio: Record<string, any> | boolean;
-  video?: Record<string, any> | boolean;
-  localVideo?: Record<string, any> | boolean;
-};
-type WebRtcConfig = {
-  displayName?: string;
-  host: string;
-  port?: number;
-  websocketSip?: string;
-  authorizationUser?: string;
-  password?: string;
-  uri?: string;
-  media?: MediaConfig;
-  iceCheckingTimeout?: number;
-  log?: Record<string, any>;
-  audioOutputDeviceId?: string;
-  audioOutputVolume?: number;
-  userAgentString?: string;
-  heartbeatDelay?: number;
-  heartbeatTimeout?: number;
-  maxHeartbeats?: number;
-  skipRegister?: boolean;
-}; // @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
+// setting a 24hr timeout and letting the backend define the actual value
+const NO_ANSWER_TIMEOUT = 60 * 60 * 24; // in seconds
 
 export default class WebRTCClient extends Emitter {
   clientId: number;
 
   config: WebRtcConfig;
 
-  uaConfigOverrides: Record<string, any> | null | undefined;
+  uaConfigOverrides?: UserAgentConfigOverrides;
 
   userAgent: UserAgent | null;
 
@@ -201,7 +180,7 @@ export default class WebRTCClient extends Emitter {
     return [];
   }
 
-  constructor(config: WebRtcConfig, session: Invitation | Inviter | null | undefined, uaConfigOverrides: Record<string, any> | null | undefined = undefined) {
+  constructor(config: WebRtcConfig, session: Invitation | Inviter | null | undefined, uaConfigOverrides?: UserAgentConfigOverrides) {
     super();
 
     // For debug purpose
@@ -271,11 +250,11 @@ export default class WebRTCClient extends Emitter {
     this.audio = media.audio;
   }
 
-  createUserAgent(configOverrides: Record<string, any> | null | undefined): UserAgent {
-    const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
+  createUserAgent(uaConfigOverrides?: UserAgentConfigOverrides): UserAgent {
+    const uaOptions = this._createUaOptions(uaConfigOverrides);
 
-    logger.info('sdk webrtc, creating UA', { webRTCConfiguration, clientId: this.clientId });
-    webRTCConfiguration.delegate = {
+    logger.info('sdk webrtc, creating UA', { uaOptions, clientId: this.clientId });
+    uaOptions.delegate = {
       onConnect: this.onConnect.bind(this),
       onDisconnect: this.onDisconnect.bind(this),
       onInvite: (invitation: Invitation) => {
@@ -293,7 +272,7 @@ export default class WebRTCClient extends Emitter {
         this.eventEmitter.emit(INVITE, invitation, this.sessionWantsToDoVideo(invitation), shouldAutoAnswer);
       },
     };
-    const ua: any = new UserAgent(webRTCConfiguration);
+    const ua: any = new UserAgent(uaOptions);
 
     if (ua.transport && ua.transport.connectPromise) {
       ua.transport.connectPromise.catch((e: any) => {
@@ -593,9 +572,7 @@ export default class WebRTCClient extends Emitter {
     const inviteOptions: InviterInviteOptions = {
       requestDelegate: {
         onAccept: (response: IncomingResponse) => {
-          // @ts-ignore
           if (session.sessionDescriptionHandler?.peerConnection) {
-            // @ts-ignore
             session.sessionDescriptionHandler.peerConnection.sfu = conference;
           }
           // @ts-ignore
@@ -2013,7 +1990,7 @@ export default class WebRTCClient extends Emitter {
     }));
   }
 
-  _createWebRTCConfiguration(configOverrides: Record<string, any> | null = {}): Record<string, any> {
+  _createUaOptions(uaOptionsOverrides: UserAgentConfigOverrides = {}): UserAgentOptions {
     let {
       host,
     } = this.config;
@@ -2025,7 +2002,8 @@ export default class WebRTCClient extends Emitter {
       port = Number(webSocketSip[1]);
     }
 
-    const config: Record<string, any> = {
+    const uaOptions: UserAgentOptions = {
+      noAnswerTimeout: NO_ANSWER_TIMEOUT,
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
       displayName: this.config.displayName,
@@ -2041,7 +2019,8 @@ export default class WebRTCClient extends Emitter {
       userAgentString: this.config.userAgentString || 'wazo-sdk',
       reconnectionAttempts: 10000,
       reconnectionDelay: 5,
-      sessionDescriptionHandlerFactory: (session: Inviter | Invitation, options: SessionDescriptionHandlerFactoryOptions = {}) => {
+
+      sessionDescriptionHandlerFactory: (session: Session, options: SessionDescriptionHandlerFactoryOptions = {}) => {
         const uaLogger = session.userAgent.getLogger('sip.WazoSessionDescriptionHandler');
 
         const isWeb = this._isWeb();
@@ -2054,10 +2033,10 @@ export default class WebRTCClient extends Emitter {
             ...(options.peerConnectionConfiguration || {}),
           },
         };
-        return new WazoSessionDescriptionHandler(uaLogger, wazoMediaStreamFactory, sdhOptions, isWeb, session);
+        return new WazoSessionDescriptionHandler(uaLogger, wazoMediaStreamFactory, sdhOptions, isWeb, session as Inviter | Invitation);
       },
       transportOptions: {
-        traceSip: configOverrides?.traceSip || false,
+        traceSip: uaOptionsOverrides?.traceSip || false,
         wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
@@ -2074,19 +2053,23 @@ export default class WebRTCClient extends Emitter {
             rtcpMuxPolicy: 'require',
             iceServers: WebRTCClient.getIceServers(this.config.host),
             ...this._getRtcOptions(),
-            ...(configOverrides?.peerConnectionOptions || {}),
+            ...(uaOptionsOverrides?.peerConnectionOptions || {}),
           },
         },
         // Configuration used in SDH to create the PeerConnection
         peerConnectionConfiguration: {
           rtcpMuxPolicy: 'require',
           iceServers: WebRTCClient.getIceServers(this.config.host),
-          ...(configOverrides?.peerConnectionOptions || {}),
+          ...(uaOptionsOverrides?.peerConnectionOptions || {}),
         },
       },
     };
-    return { ...config,
-      ...configOverrides,
+
+    delete uaOptionsOverrides?.traceSip;
+
+    return {
+      ...uaOptions,
+      ...uaOptionsOverrides,
     };
   }