@wazo/sdk 0.43.7 → 0.43.9

package/lib/web-rtc-client.js

@@ -73,7 +73,6 @@ const array_1 = require("./utils/array");
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
 const replaceLocalIpModifier = (description) => Promise.resolve(Object.assign(Object.assign({}, JSON.parse(JSON.stringify(description))), { sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1') }));
 exports.replaceLocalIpModifier = replaceLocalIpModifier;
-const SIP_ID_LENGTH = 36;
 const DEFAULT_ICE_TIMEOUT = 3000;
 const SEND_STATS_DELAY = 5000;
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
@@ -195,7 +194,7 @@ class WebRTCClient extends Emitter_1.default {
                 logger.info('sdk webrtc on invite', {
                     method: 'delegate.onInvite',
                     clientId: this.clientId,
-                    id: invitation.id,
+                    id: this.getSipSessionId(invitation),
                     remoteURI: invitation.remoteURI,
                 });
                 this._setupSession(invitation);
@@ -485,7 +484,7 @@ class WebRTCClient extends Emitter_1.default {
                 },
                 onReject: (response) => {
                     logger.info('on call rejected', {
-                        id: session === null || session === void 0 ? void 0 : session.id,
+                        id: this.getSipSessionId(session),
                         // @ts-ignore: fromTag does not exist
                         fromTag: session === null || session === void 0 ? void 0 : session.fromTag,
                     });
@@ -515,7 +514,7 @@ class WebRTCClient extends Emitter_1.default {
     answer(session, enableVideo) {
         logger.info('sdk webrtc answer call', {
             clientId: this.clientId,
-            id: session.id,
+            id: this.getSipSessionId(session),
             enableVideo,
         });
         if (!session || !session.accept) {
@@ -531,7 +530,7 @@ class WebRTCClient extends Emitter_1.default {
             if (session.isCanceled) {
                 const message = 'accepted a canceled session (or was canceled during the accept phase).';
                 logger.error(message, {
-                    id: session.id,
+                    id: this.getSipSessionId(session),
                 });
                 this.onCallEnded(session);
                 throw new CanceledCallError(message);
@@ -539,7 +538,7 @@ class WebRTCClient extends Emitter_1.default {
             logger.info('sdk webrtc answer, accepted.');
             this._onAccepted(session);
         }).catch(e => {
-            logger.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
+            logger.error(`answer call error for ${this.getSipSessionId(session)}`, e);
             throw e;
         });
     }
@@ -609,7 +608,7 @@ class WebRTCClient extends Emitter_1.default {
         return __awaiter(this, void 0, void 0, function* () {
             logger.info('sdk webrtc reject call', {
                 clientId: this.clientId,
-                id: session.id,
+                id: this.getSipSessionId(session),
             });
             try {
                 if (session instanceof api_1.Invitation) {
@@ -673,13 +672,13 @@ class WebRTCClient extends Emitter_1.default {
     }
     mute(session) {
         logger.info('sdk webrtc mute', {
-            id: session.id,
+            id: this.getSipSessionId(session),
         });
         this._toggleAudio(session, true);
     }
     unmute(session) {
         logger.info('sdk webrtc unmute', {
-            id: session.id,
+            id: this.getSipSessionId(session),
         });
         this._toggleAudio(session, false);
     }
@@ -716,13 +715,13 @@ class WebRTCClient extends Emitter_1.default {
     }
     toggleCameraOn(session) {
         logger.info('sdk webrtc toggle camera on', {
-            id: session.id,
+            id: this.getSipSessionId(session),
         });
         this._toggleVideo(session, false);
     }
     toggleCameraOff(session) {
         logger.info('sdk webrtc toggle camera off', {
-            id: session.id,
+            id: this.getSipSessionId(session),
         });
         this._toggleVideo(session, true);
     }
@@ -979,7 +978,7 @@ class WebRTCClient extends Emitter_1.default {
             return;
         }
         logger.info('send WebRTC message', {
-            sipId: sipSession.id,
+            sipId: this.getSipSessionId(sipSession),
             contentType,
         });
         try {
@@ -1344,18 +1343,19 @@ class WebRTCClient extends Emitter_1.default {
         if (!sipSession) {
             return '';
         }
-        const message = sipSession.message;
-        if (message === null || message === void 0 ? void 0 : message.callId) {
-            return message.callId.substring(0, SIP_ID_LENGTH);
-        }
         // For Inviter
         // @ts-ignore: private
         if (sipSession instanceof api_1.Inviter && sipSession.outgoingRequestMessage) {
             // @ts-ignore: private
-            return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_LENGTH);
+            return sipSession.outgoingRequestMessage.callId;
         }
         // For Invitation
-        return (sipSession.id || '').substring(0, SIP_ID_LENGTH);
+        // @ts-ignore: private
+        if (sipSession instanceof api_1.Invitation && sipSession.incomingInviteRequest.message) {
+            // @ts-ignore: private
+            return sipSession.incomingInviteRequest.message.callId;
+        }
+        return '';
     }
     waitForRegister() {
         return __awaiter(this, void 0, void 0, function* () {
@@ -1415,7 +1415,7 @@ class WebRTCClient extends Emitter_1.default {
         this.sipSessions[id] = session;
     }
     getSipSession(id) {
-        return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id] : null;
     }
     getSipSessionIds() {
         return Object.keys(this.sipSessions);
@@ -1791,7 +1791,7 @@ class WebRTCClient extends Emitter_1.default {
     _onProgress(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, early = false) {
             yield this._setupMedias(session);
-            const sessionId = this.getSipSessionId(session).substr(0, 20);
+            const sessionId = this.getSipSessionId(session);
             this.updateRemoteStream(sessionId);
             logger.info('progress received', {
                 sessionId,
@@ -1804,7 +1804,7 @@ class WebRTCClient extends Emitter_1.default {
         return __awaiter(this, arguments, void 0, function* (session, sessionDialog, withEvent = true, initAllTracks = true) {
             var _a;
             logger.info('on call accepted', {
-                id: session.id,
+                id: this.getSipSessionId(session),
                 clientId: this.clientId,
                 remoteTag: session.remoteTag,
             });
@@ -2161,18 +2161,18 @@ class WebRTCClient extends Emitter_1.default {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state === session_state_1.SessionState.Terminated || session.state === session_state_1.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: this.getSipSessionId(session) });
                 throw new Error(error);
             }
             if (session.state !== session_state_1.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: this.getSipSessionId(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: this.getSipSessionId(session) });
                 throw new Error(error);
             }
             try {
@@ -2186,12 +2186,12 @@ class WebRTCClient extends Emitter_1.default {
     _reject(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.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: this.getSipSessionId(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: this.getSipSessionId(session) });
                 return;
             }
             try {
@@ -2205,7 +2205,7 @@ class WebRTCClient extends Emitter_1.default {
     _cancel(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.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: this.getSipSessionId(session) });
                 return;
             }
             try {
@@ -2219,7 +2219,7 @@ class WebRTCClient extends Emitter_1.default {
     _bye(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.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: this.getSipSessionId(session) });
                 return null;
             }
             try {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.43.7",
+  "version": "0.43.9",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "index.js",
   "types": "lib/types/index.d.ts",

package/src/domain/CallLog.ts

@@ -118,8 +118,7 @@ export default class CallLog {
 
   static parseNew(plain: CallLogResponse, session: Session): CallLog {
     const callLog: CallLog = CallLog.parse(plain);
-    // @TODO: FIXME add verification declined vs missed call
-    callLog.newMissedCall = session && session.hasExtension((plain.requested_extension || plain.destination_extension) as string) && !plain.answered;
+    callLog.newMissedCall = session && plain.destination_user_uuid === session.uuid && !plain.answered;
     return callLog;
   }
 
@@ -203,7 +202,7 @@ export default class CallLog {
   }
 
   isAnOutgoingCall(session: Session): boolean {
-    console.warn(`@wazo/sdk 
+    console.warn(`@wazo/sdk
       CallLog.isAnOutgoingcall(session) method is obsolete.
       Please use CallLog.isOutgoing(session).
     `);

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';
@@ -60,7 +60,7 @@ export const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
 export const events = [ON_USER_AGENT, ON_REGISTERED, ON_UNREGISTERED, ON_PROGRESS, ON_CALL_ACCEPTED, ON_CALL_ANSWERED, ON_CALL_INCOMING, ON_CALL_OUTGOING, ON_CALL_MUTED, ON_CALL_UNMUTED, ON_CALL_RESUMED, ON_CALL_HELD, ON_CALL_UNHELD, ON_CAMERA_DISABLED, ON_CALL_FAILED, ON_CALL_ENDED, ON_CALL_REJECTED, ON_MESSAGE, ON_REINVITE, ON_TRACK, ON_AUDIO_STREAM, ON_VIDEO_STREAM, ON_REMOVE_STREAM, ON_SHARE_SCREEN_ENDED, ON_TERMINATE_SOUND, ON_PLAY_RING_SOUND, ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, ON_PLAY_HANGUP_SOUND, ON_PLAY_PROGRESS_SOUND, ON_VIDEO_INPUT_CHANGE, ON_SHARE_SCREEN_STARTED, ON_CALL_ERROR, ON_CHAT, ON_SIGNAL, ON_NETWORK_STATS, ON_DISCONNECTED, ON_EARLY_MEDIA];
 const logger = IssueReporter.loggerFor('webrtc-phone');
 export default class WebRTCPhone extends Emitter implements Phone {
-  client: WazoWebRTCClient;
+  client: WebRTCClient;
 
   allowVideo: boolean;
 
@@ -94,7 +94,7 @@ 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;
@@ -159,7 +159,7 @@ 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: this.getSipSessionId(sipSession),
       constraints,
       audioOnly,
       conference,
@@ -380,7 +380,9 @@ 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);
       };
     }
@@ -389,7 +391,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       switch (newState) {
         case SessionState.Terminating:
           logger.info('WebRTC phone - call terminating', {
-            sipId: sipSession.id,
+            sipId: this.getSipSessionId(sipSession),
           });
           this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
           break;
@@ -397,7 +399,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         case SessionState.Terminated:
         {
           logger.info('WebRTC phone - call terminated', {
-            sipId: sipSession.id,
+            sipId: this.getSipSessionId(sipSession),
           });
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
@@ -580,7 +582,7 @@ 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: this.getSipSessionId(sipSession),
       cameraEnabled,
     });
 
@@ -588,7 +590,7 @@ 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: this.getSipSessionId(sipSession),
         cameraEnabled,
       });
 
@@ -673,7 +675,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSessionId = this.getSipSessionId(sipSession);
 
     logger.info('WebRTC phone - on call terminated', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       sipSessionId,
       ids: Object.keys(this.callSessions),
     });
@@ -697,9 +699,9 @@ 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;
@@ -709,7 +711,7 @@ 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: this.getSipSessionId(sipSession),
       shouldRetrigger,
       hasIncomingCallSession,
     });
@@ -853,7 +855,7 @@ 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: this.getSipSessionId(sipSession) });
 
       return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
         return callSession.sipCallId;
@@ -898,7 +900,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (sipSession) {
       logger.info('WebRTC rejecting', {
-        sipId: sipSession.id,
+        sipId: this.getSipSessionId(sipSession),
       });
       this.client.hangup(sipSession);
     }
@@ -919,7 +921,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     if (sipSession) {
       logger.info('WebRTC atxfer', {
-        sipId: sipSession.id,
+        sipId: this.getSipSessionId(sipSession),
       });
       return this.client.atxfer(sipSession);
     }
@@ -950,13 +952,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC resuming', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(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 && this.getSipSessionId(this.currentSipSession) !== this.getSipSessionId(sipSession)) {
       logger.info('WebRTC hold call after resume', {
-        id: this.currentSipSession.id,
+        id: this.getSipSessionId(this.currentSipSession),
       });
       this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
     }
@@ -979,7 +981,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sessionId = this.getSipSessionId(sipSession);
     const hasVideo = this.client.hasLocalVideo(sessionId);
     logger.info('WebRTC hold sip session', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       hasVideo,
     });
 
@@ -1076,7 +1078,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC muting', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
     });
     this.client.mute(sipSession);
 
@@ -1104,7 +1106,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC unmuting', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
     });
     this.client.unmute(sipSession);
 
@@ -1139,7 +1141,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC turn camera on', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
     });
     this.client.toggleCameraOn(sipSession);
     this.eventEmitter.emit(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
@@ -1153,7 +1155,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC turn camera off', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
     });
     this.client.toggleCameraOff(sipSession);
     this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -1167,7 +1169,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC send key', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       tone,
     });
     this.client.sendDTMF(sipSession, tone);
@@ -1245,7 +1247,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC transfer', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       target,
     });
     this.client.transfer(sipSession, target);
@@ -1255,8 +1257,8 @@ 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 ? this.getSipSessionId(sipSession) : null,
+      target: sipSessionTarget ? this.getSipSessionId(sipSessionTarget) : null,
     });
 
     if (!sipSessionTarget) {
@@ -1295,7 +1297,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const sipSessionId = this.getSipSessionId(sipSession);
 
     logger.info('WebRTC hangup', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       sipSessionId,
     });
     await this.client.hangup(sipSession);
@@ -1484,7 +1486,7 @@ 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: this.getSipSessionId(sipSession),
         withVideo,
         autoAnswer,
       });
@@ -1555,7 +1557,7 @@ 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: this.getSipSessionId(session) });
 
       const callSession = this._createCallSession(session);
 
@@ -1687,7 +1689,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.callSessions[sipSessionId];
   }
 
-  getSipSessionId(sipSession: WazoSession): string {
+  getSipSessionId(sipSession: WazoSession | null | undefined): string {
+    if (!sipSession) {
+      return '';
+    }
     return this.client.getSipSessionId(sipSession);
   }
 

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';
@@ -19,7 +19,7 @@ const protocolLogger = IssueReporter.loggerFor('sip');
 const protocolDebugMessages = ['Received WebSocket text message:', 'Sending WebSocket message:'];
 
 export class Phone extends Emitter {
-  client: WazoWebRTCClient;
+  client: WebRTCClient;
 
   phone: WebRTCPhone | null | undefined;
 
@@ -186,7 +186,7 @@ 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,