@wazo/sdk 1.0.0-alpha.8 → 1.0.0-alpha.9

package/src/domain/Phone/WebRTCPhone.ts

@@ -8,7 +8,7 @@ import { Inviter, Invitation, Session, SessionDescriptionHandlerOptions } from '
 import { SessionDescriptionHandler } from 'sip.js/lib/platform/web';
 import CallSession from '../CallSession';
 import type { Phone, AvailablePhoneOptions } from './Phone';
-import WazoWebRTCClient from '../../web-rtc-client';
+import WebRTCClient from '../../web-rtc-client';
 import Emitter from '../../utils/Emitter';
 import IssueReporter from '../../service/IssueReporter';
 import { PeerConnection, WazoSession } from '../types';
@@ -64,7 +64,7 @@ const logger = IssueReporter.loggerFor('webrtc-phone');
 
 // @deprecated: Use Wazo.Voice.Softphone instead
 export default class WebRTCPhone extends Emitter implements Phone {
-  client: WazoWebRTCClient;
+  client: WebRTCClient;
 
   allowVideo: boolean;
 
@@ -98,7 +98,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
   shouldSendReinvite: boolean;
 
-  constructor(client: WazoWebRTCClient, audioOutputDeviceId: string | undefined, allowVideo = false, audioRingDeviceId?: string) {
+  constructor(client: WebRTCClient, audioOutputDeviceId: string | undefined, allowVideo = false, audioRingDeviceId?: string) {
     super();
     this.client = client;
     this.allowVideo = allowVideo;
@@ -163,7 +163,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   async sendReinvite(callSession?: CallSession, constraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
     const sipSession = this.findSipSession(callSession);
     logger.info('WebRTC phone - send reinvite', {
-      sessionId: sipSession ? sipSession.id : null,
+      sessionId: getSipCallId(sipSession),
       constraints,
       audioOnly,
       conference,
@@ -384,7 +384,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
         const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
         const callSession = this._createCallSession(sipSession);
 
-        delete this.callSessions[sipSessionId];
+        if (sipSessionId) {
+          delete this.callSessions[sipSessionId];
+        }
         this.eventEmitter.emit(ON_CALL_CANCELED, callSession, elsewhere);
       };
     }
@@ -393,7 +395,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       switch (newState) {
         case SessionState.Terminating:
           logger.info('WebRTC phone - call terminating', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
           });
           this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
           break;
@@ -401,7 +403,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         case SessionState.Terminated:
         {
           logger.info('WebRTC phone - call terminated', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
           });
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
@@ -584,7 +586,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
   _onCallAccepted(sipSession: WazoSession, cameraEnabled: boolean): CallSession {
     logger.info('WebRTC phone - on call accepted', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       cameraEnabled,
     });
 
@@ -592,7 +594,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
       logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
-        sipId: sipSession.id,
+        sipId: getSipCallId(sipSession),
         cameraEnabled,
       });
 
@@ -677,7 +679,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSessionId = getSipCallId(sipSession);
 
     logger.info('WebRTC phone - on call terminated', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       sipSessionId,
       ids: Object.keys(this.callSessions),
     });
@@ -701,9 +703,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (sipSessionId) {
       this.removeIncomingSessions(sipSessionId);
-    }
 
-    delete this.callSessions[sipSessionId];
+      delete this.callSessions[sipSessionId];
+    }
 
     if (isCurrentSession) {
       this.currentSipSession = undefined;
@@ -713,7 +715,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const hasIncomingCallSession = this.hasIncomingCallSession();
     // Re-trigger incoming call event for remaining incoming calls
     logger.info('WebRTC phone - check to re-trigger incoming call', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       shouldRetrigger,
       hasIncomingCallSession,
     });
@@ -857,7 +859,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSession = this.client.getSipSession(callSession.getId()) as Invitation;
 
     if (sipSession) {
-      logger.info('accept call, session found', { sipId: sipSession.id });
+      logger.info('accept call, session found', { sipId: getSipCallId(sipSession) });
 
       return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
         return callSession.sipCallId;
@@ -902,7 +904,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (sipSession) {
       logger.info('WebRTC rejecting', {
-        sipId: sipSession.id,
+        sipId: getSipCallId(sipSession),
       });
       this.client.hangup(sipSession);
     }
@@ -923,7 +925,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (sipSession) {
       logger.info('WebRTC atxfer', {
-        sipId: sipSession.id,
+        sipId: getSipCallId(sipSession),
       });
       return this.client.atxfer(sipSession);
     }
@@ -954,13 +956,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC resuming', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
     });
 
     // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
-    if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
+    if (this.currentSipSession && getSipCallId(this.currentSipSession) !== getSipCallId(sipSession)) {
       logger.info('WebRTC hold call after resume', {
-        id: this.currentSipSession.id,
+        id: getSipCallId(this.currentSipSession),
       });
       this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
     }
@@ -983,7 +985,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sessionId = getSipCallId(sipSession);
     const hasVideo = this.client.hasLocalVideo(sessionId);
     logger.info('WebRTC hold sip session', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       hasVideo,
     });
 
@@ -1080,7 +1082,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC muting', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
     });
     this.client.mute(sipSession);
 
@@ -1108,7 +1110,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC unmuting', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
     });
     this.client.unmute(sipSession);
 
@@ -1143,7 +1145,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC turn camera on', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
     });
     this.client.toggleCameraOn(sipSession);
     this.eventEmitter.emit(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
@@ -1157,7 +1159,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC turn camera off', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
     });
     this.client.toggleCameraOff(sipSession);
     this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -1171,7 +1173,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC send key', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       tone,
     });
     this.client.sendDTMF(sipSession, tone);
@@ -1249,7 +1251,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC transfer', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       target,
     });
     this.client.transfer(sipSession, target);
@@ -1259,8 +1261,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSession = this.client.getSipSession(source.sipCallId);
     const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
     logger.info('WebRTC indirect transfer', {
-      sipId: sipSession ? sipSession.id : null,
-      target: sipSessionTarget ? sipSessionTarget.id : null,
+      sipId: sipSession ? getSipCallId(sipSession) : null,
+      target: sipSessionTarget ? getSipCallId(sipSessionTarget) : null,
     });
 
     if (!sipSessionTarget) {
@@ -1299,7 +1301,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSessionId = getSipCallId(sipSession);
 
     logger.info('WebRTC hangup', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       sipSessionId,
     });
     await this.client.hangup(sipSession);
@@ -1488,7 +1490,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
       const withVideo = this.allowVideo ? wantsToDoVideo : false;
       logger.info('WebRTC invite received', {
-        sipId: sipSession.id,
+        sipId: getSipCallId(sipSession),
         withVideo,
         autoAnswer,
       });
@@ -1559,7 +1561,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
     this.client.on(this.client.REJECTED, session => {
-      logger.info('WebRTC call rejected', session.id);
+      logger.info('WebRTC call rejected', { id: getSipCallId(session) });
 
       const callSession = this._createCallSession(session);
 

package/src/simple/Auth.ts

@@ -167,6 +167,10 @@ export class Auth {
     return this._onAuthenticated(rawSession as Session);
   }
 
+  async samlLogOut(): Promise<void | { location: string }> {
+    return await this.logout(false, true);
+  }
+
   async initiateIdpAuthentication(domain: string, redirectUrl: string): Promise<{ location: string, saml_session_id: string } | undefined> {
     try {
       return await getApiClient().auth.initiateIdpAuthentication(domain, redirectUrl);
@@ -218,20 +222,26 @@ export class Auth {
     return getApiClient().auth.refreshToken(refreshToken, '', this.expiration);
   }
 
-  async logout(deleteRefreshToken = true): Promise<void> {
+  async logout(deleteRefreshToken = true, saml = false): Promise<void | { location: string }> {
     try {
       Wazo.Websocket.close(true);
 
-      if (this.clientId && deleteRefreshToken) {
+      if (!saml && this.clientId && deleteRefreshToken) {
         await getApiClient().auth.deleteRefreshToken(this.clientId);
       }
     } catch (e: any) {
       // Nothing to
     }
 
+    let response;
+
     try {
       if (this.session?.token) {
-        await getApiClient().auth.logOut(this.session.token);
+        if (saml) {
+          response = await getApiClient().auth.samlLogOut();
+        } else {
+          await getApiClient().auth.logOut(this.session.token);
+        }
       }
     } catch (e: any) {
       // Nothing to
@@ -243,6 +253,8 @@ export class Auth {
     this.authenticated = false;
     this.usingEdgeServer = undefined;
     setFetchOptions({});
+
+    return response;
   }
 
   setOnHostFromHeaders(callback: (hostFromHeaders: string) => void) {

package/src/simple/Phone.ts

@@ -6,7 +6,7 @@ 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 WebRTCClient, { events as clientEvents, transportEvents } from '../web-rtc-client';
 import IssueReporter from '../service/IssueReporter';
 import Emitter from '../utils/Emitter';
 import Wazo from './index';
@@ -21,7 +21,7 @@ const protocolDebugMessages = ['Received WebSocket text message:', 'Sending WebS
 
 // @deprecated: Use Wazo.Voice.Softphone instead
 export class Phone extends Emitter {
-  client: WazoWebRTCClient;
+  client: WebRTCClient;
 
   phone: WebRTCPhone | null | undefined;
 
@@ -188,7 +188,7 @@ export class Phone extends Emitter {
       options.log.connector = this._logConnector;
     }
 
-    this.client = new WazoWebRTCClient({
+    this.client = new WebRTCClient({
       host,
       port: typeof port === 'string' ? parseInt(port, 10) : port,
       displayName,

package/src/utils/sdp.ts

@@ -1,7 +1,7 @@
 /* eslint-disable no-underscore-dangle */
 import sdpParser from 'sdp-transform';
 import { UserAgent } from 'sip.js/lib/api/user-agent';
-import { Inviter } from 'sip.js/lib/api';
+import { Inviter, Invitation } from 'sip.js/lib/api';
 import { URI } from 'sip.js/lib/grammar/uri';
 
 import type { SipCall } from '../domain/types';
@@ -152,20 +152,21 @@ export const getSipCallId = (sipSession: SipCall | null | undefined): string =>
     return '';
   }
 
-  const message = sipSession.message as Partial<{ callId?: string }>;
-  if (message?.callId) {
-    return message.callId.substring(0, SIP_ID_MIN_LENGTH);
-  }
-
   // For Inviter
   // @ts-ignore: private
   if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage) {
     // @ts-ignore: private
-    return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_MIN_LENGTH);
+    return sipSession.outgoingRequestMessage.callId;
   }
 
   // For Invitation
-  return (sipSession.id || '').substring(0, SIP_ID_MIN_LENGTH);
+  // @ts-ignore: private
+  if (sipSession instanceof Invitation && sipSession.incomingInviteRequest.message) {
+    // @ts-ignore: private
+    return sipSession.incomingInviteRequest.message.callId;
+  }
+
+  return sipSession.id;
 };
 
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.

package/src/voice/call.ts

@@ -116,7 +116,6 @@ class Call extends EventEmitter {
     } as any;
 
     const call = new Call(sipCall, softphone);
-    // call._sendAction(Actions.ES);
     call.apiId = apiCall.id;
     call.answerTime = new Date(+apiCall.startingTime);
 
@@ -158,8 +157,6 @@ class Call extends EventEmitter {
 
     this.number = getCallNumber(sipCall);
     this.displayName = getCallDisplayName(sipCall);
-
-    this._bindEvents();
   }
 
   async accept(options: AcceptOptions = {}): Promise<string | null> {
@@ -551,6 +548,8 @@ class Call extends EventEmitter {
   }
 
   onMakeCall() {
+    this._bindEvents();
+
     this._sendAction(Actions.MAKE_CALL);
   }
 

package/src/web-rtc-client.ts

@@ -245,7 +245,7 @@ export default class WebRTCClient extends Emitter {
         logger.info('sdk webrtc on invite', {
           method: 'delegate.onInvite',
           clientId: this.clientId,
-          id: invitation.id,
+          id: getSipCallId(invitation),
           remoteURI: (invitation as any).remoteURI,
         });
 
@@ -583,7 +583,7 @@ export default class WebRTCClient extends Emitter {
         },
         onReject: (response: IncomingResponse) => {
           logger.info('on call rejected', {
-            id: session?.id,
+            id: getSipCallId(session),
             // @ts-ignore: fromTag does not exist
             fromTag: session?.fromTag,
           });
@@ -620,7 +620,7 @@ export default class WebRTCClient extends Emitter {
   answer(session: Invitation, enableVideo?: boolean): Promise<void> {
     logger.info('sdk webrtc answer call', {
       clientId: this.clientId,
-      id: session.id,
+      id: getSipCallId(session),
       enableVideo,
     });
 
@@ -638,7 +638,7 @@ export default class WebRTCClient extends Emitter {
       if (session.isCanceled) {
         const message = 'accepted a canceled session (or was canceled during the accept phase).';
         logger.error(message, {
-          id: session.id,
+          id: getSipCallId(session),
         });
         this.onCallEnded(session);
         throw new CanceledCallError(message);
@@ -648,7 +648,7 @@ export default class WebRTCClient extends Emitter {
 
       this._onAccepted(session);
     }).catch(e => {
-      logger.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
+      logger.error(`answer call error for ${getSipCallId(session)}`, e);
       throw e;
     });
   }
@@ -716,7 +716,7 @@ export default class WebRTCClient extends Emitter {
   async reject(session: WazoSession): Promise<void> {
     logger.info('sdk webrtc reject call', {
       clientId: this.clientId,
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     try {
@@ -779,7 +779,7 @@ export default class WebRTCClient extends Emitter {
 
   mute(session: WazoSession): void {
     logger.info('sdk webrtc mute', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleAudio(session, true);
@@ -787,7 +787,7 @@ export default class WebRTCClient extends Emitter {
 
   unmute(session: WazoSession): void {
     logger.info('sdk webrtc unmute', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleAudio(session, false);
@@ -832,7 +832,7 @@ export default class WebRTCClient extends Emitter {
 
   toggleCameraOn(session: WazoSession): void {
     logger.info('sdk webrtc toggle camera on', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleVideo(session, false);
@@ -840,7 +840,7 @@ export default class WebRTCClient extends Emitter {
 
   toggleCameraOff(session: WazoSession): void {
     logger.info('sdk webrtc toggle camera off', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleVideo(session, true);
@@ -1041,7 +1041,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     logger.info('send WebRTC message', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       contentType,
     });
 
@@ -1498,7 +1498,7 @@ export default class WebRTCClient extends Emitter {
   }
 
   getSipSession(id: string): WazoSession | null | undefined {
-    return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_MIN_LENGTH)] : null;
+    return id in this.sipSessions ? this.sipSessions[id] : null;
   }
 
   getSipCallIds(): string[] {
@@ -1769,18 +1769,17 @@ export default class WebRTCClient extends Emitter {
   async _onProgress(session: WazoSession, early = false): Promise<void> {
     await this._setupMedias(session);
 
-    const sessionId = getSipCallId(session).substr(0, 20);
+    const sessionId = getSipCallId(session);
+
     updateRemoteStream(session);
-    logger.info('progress received', {
-      sessionId,
-      early,
-    });
+
+    logger.info('progress received', { sessionId, early });
     this.eventEmitter.emit(early ? ON_EARLY_MEDIA : ON_PROGRESS, session);
   }
 
   async _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): Promise<void> {
     logger.info('on call accepted', {
-      id: session.id,
+      id: getSipCallId(session),
       clientId: this.clientId,
       remoteTag: session.remoteTag,
     });
@@ -2018,20 +2017,20 @@ export default class WebRTCClient extends Emitter {
   async _accept(session: Invitation, options: InvitationAcceptOptions = {}) {
     if (session.state === SessionState.Terminated || session.state === SessionState.Terminating) {
       const error = 'Trying to accept a terminated sipSession.';
-      logger.warn(error, { state: session.state, sessionId: session.id });
+      logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
       throw new Error(error);
     }
 
     if (session.state !== SessionState.Initial) {
       const error = 'Trying to accept a non Initial sipSession.';
-      logger.warn(error, { state: session.state, sessionId: session.id });
+      logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
       throw new Error(error);
     }
 
     // @ts-ignore: private
     if (!session.incomingInviteRequest.acceptable) {
       const error = 'Trying to reject a non `acceptable` session';
-      logger.warn(error, { state: session.state, sessionId: session.id });
+      logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
       throw new Error(error);
     }
 
@@ -2044,13 +2043,13 @@ export default class WebRTCClient extends Emitter {
 
   async _reject(session: Invitation, options: InvitationRejectOptions = {}) {
     if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
-      logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: session.id });
+      logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
       return;
     }
 
     // @ts-ignore: private
     if (!session.incomingInviteRequest.rejectable) {
-      logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: session.id });
+      logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: getSipCallId(session) });
       return;
     }
 
@@ -2063,7 +2062,7 @@ export default class WebRTCClient extends Emitter {
 
   async _cancel(session: Inviter, options: InviterCancelOptions = {}) {
     if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
-      logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: session.id });
+      logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
       return;
     }
 
@@ -2076,7 +2075,7 @@ export default class WebRTCClient extends Emitter {
 
   async _bye(session: WazoSession, options: SessionByeOptions = {}) {
     if (session.state !== SessionState.Established) {
-      logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: session.id });
+      logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
       return null;
     }