diff --git a/src/domain/Phone/WebRTCPhone.ts b/src/domain/Phone/WebRTCPhone.ts
index 34e4b753..90cd3fdf 100644
--- a/src/domain/Phone/WebRTCPhone.ts
+++ b/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;
diff --git a/src/simple/Phone.ts b/src/simple/Phone.ts
index c41b2f92..05f7bcd2 100644
--- a/src/simple/Phone.ts
+++ b/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,