diff --git a/package.json b/package.json
index c641ca61..024d6b73 100644
--- a/package.json
+++ b/package.json
@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.42.0",
+  "version": "0.43.0-rc.0",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "index.js",
   "types": "lib/types/index.d.ts",
diff --git a/src/__tests__/web-rtc-client.test.ts b/src/__tests__/web-rtc-client.test.ts
index 7f5cef49..9e89de52 100644
--- a/src/__tests__/web-rtc-client.test.ts
+++ b/src/__tests__/web-rtc-client.test.ts
@@ -12,7 +12,7 @@ jest.mock('sip.js/lib/api/user-agent', () => ({
   },
 }));
 
-const client = new WebRTCClient({} as any, null, undefined);
+const client = new WebRTCClient({} as any, undefined, undefined);
 
 describe('WebRTC client', () => {
   it('should compute muted/unmuted state', async () => {
diff --git a/src/domain/Phone/CTIPhone.ts b/src/domain/Phone/CTIPhone.ts
index e7beb581..44c35f8d 100644
--- a/src/domain/Phone/CTIPhone.ts
+++ b/src/domain/Phone/CTIPhone.ts
@@ -300,7 +300,7 @@ export default class CTIPhone extends Emitter implements Phone {
 
   changeVideoInputDevice() { return Promise.resolve(null); }
 
-  changeAudioDevice() {}
+  changeAudioDevice() { return Promise.resolve(); }
 
   changeRingDevice() {}
 
diff --git a/src/domain/Phone/Phone.ts b/src/domain/Phone/Phone.ts
index d2c69cca..81a48521 100644
--- a/src/domain/Phone/Phone.ts
+++ b/src/domain/Phone/Phone.ts
@@ -31,7 +31,7 @@ export type AvailablePhoneOptions = {
 };
 export interface Phone extends IEmitter {
   accept(callSession: CallSession, enableVideo?: boolean): Promise<string | null>;
-  changeAudioDevice(id: string): PhoneVoid;
+  changeAudioDevice(id: string): Promise<void>;
   changeRingDevice(id: string): PhoneVoid;
   changeAudioVolume(volume: number): PhoneVoid;
   changeRingVolume(volume: number): PhoneVoid;
diff --git a/src/domain/Phone/WebRTCPhone.ts b/src/domain/Phone/WebRTCPhone.ts
index 51a1c390..34e4b753 100644
--- a/src/domain/Phone/WebRTCPhone.ts
+++ b/src/domain/Phone/WebRTCPhone.ts
@@ -612,12 +612,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return callSession;
   }
 
-  changeAudioDevice(id: string) {
+  async changeAudioDevice(id: string) {
     logger.info('WebRTC phone - change audio device', {
       deviceId: id,
     });
     this.audioOutputDeviceId = id;
-    this.client.changeAudioOutputDevice(id);
+    return this.client.changeAudioOutputDevice(id);
   }
 
   createAudioElementFor(sessionId: string) {
@@ -1528,7 +1528,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
       }, 2000);
     });
-    this.client.on(this.client.ACCEPTED, (sipSession: WazoSession) => {
+    this.client.on(this.client.ACCEPTED, async (sipSession: WazoSession) => {
       const sessionId = this.getSipSessionId(sipSession);
       const hasSession = (sessionId in this.callSessions);
       logger.info('WebRTC call accepted', {
@@ -1546,7 +1546,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
 
       if (this.audioOutputDeviceId) {
-        this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
+        await this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
       }
     });
     this.client.on('ended', () => {});
diff --git a/src/domain/types.ts b/src/domain/types.ts
index 98fe7118..0e77584f 100644
--- a/src/domain/types.ts
+++ b/src/domain/types.ts
@@ -243,21 +243,18 @@ export type WebRtcConfig = {
   media?: MediaConfig;
   iceCheckingTimeout?: number;
   log?: Record<string, any>;
-  audioOutputDeviceId?: string;
   audioOutputVolume?: number;
   userAgentString?: string;
   heartbeatDelay?: number;
   heartbeatTimeout?: number;
   maxHeartbeats?: number;
   skipRegister?: boolean;
-  userUuid?: string;
+  userUuid?: string,
+  uaConfigOverrides?: UserAgentConfigOverrides,
+  audioDeviceOutput?: string,
+  audioDeviceRing?: string
 }; // @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
 
-export type ConnectionOptions = WebRtcConfig & {
-  uaConfigOverrides: UserAgentConfigOverrides;
-  audioDeviceOutput: string;
-  audioDeviceRing: string;
-};
 export type IncomingResponse = SipIncomingResponse & { session: any };
 
 export type PeerConnection = RTCPeerConnection & {
diff --git a/src/simple/Phone.ts b/src/simple/Phone.ts
index 7634f5b4..c41b2f92 100644
--- a/src/simple/Phone.ts
+++ b/src/simple/Phone.ts
@@ -11,7 +11,7 @@ import IssueReporter from '../service/IssueReporter';
 import Emitter from '../utils/Emitter';
 import Wazo from './index';
 import SFUNotAvailableError from '../domain/SFUNotAvailableError';
-import { ConnectionOptions, WazoSession } from '../domain/types';
+import { WazoSession, WebRtcConfig } from '../domain/types';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -127,7 +127,7 @@ export class Phone extends Emitter {
     this.SessionState = SessionState;
   }
 
-  async connect(rawOptions: Partial<ConnectionOptions> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
+  async connect(rawOptions: Partial<WebRtcConfig> = {}, sipLine: SipLine | null | undefined = null): Promise<void> {
     const options = rawOptions;
     if (this.phone) {
       // Already connected
@@ -162,7 +162,7 @@ export class Phone extends Emitter {
     server: string,
     sipLine: SipLine,
     displayName: string,
-    rawOptions: Partial<ConnectionOptions> = {},
+    rawOptions: Partial<WebRtcConfig> = {},
   ): void {
     if (this.phone) {
       // Already connected
@@ -194,7 +194,7 @@ export class Phone extends Emitter {
       password: sipLine.secret,
       uri: `${sipLine.username}@${server}`,
       ...options,
-    }, null, options.uaConfigOverrides);
+    }, undefined, options.uaConfigOverrides);
     this.phone = new WebRTCPhone(this.client, options.audioDeviceOutput, true, options.audioDeviceRing);
 
     this._transferEvents();
diff --git a/src/simple/room/Room.ts b/src/simple/room/Room.ts
index 522f4da3..b52708a7 100644
--- a/src/simple/room/Room.ts
+++ b/src/simple/room/Room.ts
@@ -492,13 +492,13 @@ class Room extends Emitter {
     // Listen to REINVITE to ba able to map msid after upgrading to video in a  audio only conference
     // This allow to map msid with the non parsed (eg without the `stripVideo` modifier) SDP
     Wazo.Phone.phone.on(Wazo.Phone.phone.client.ON_REINVITE, this._boundOnReinvite);
-    this.on(this.ON_AUDIO_STREAM, stream => {
+    this.on(this.ON_AUDIO_STREAM, async stream => {
       logger.info('on room audio stream');
       this.audioStream = stream;
 
       if (!this.roomAudioElement) {
         const sessionId = Wazo.Phone.phone?.getSipSessionId(Wazo.Phone.phone?.currentSipSession as WazoSession);
-        this.roomAudioElement = Wazo.Phone.phone?.createAudioElementFor(sessionId as string) as HTMLAudioElement;
+        this.roomAudioElement = await Wazo.Phone.phone?.createAudioElementFor(sessionId as string) as HTMLAudioElement;
         this.roomAudioElement.srcObject = stream;
       } else {
         this.roomAudioElement.srcObject = stream;
diff --git a/src/web-rtc-client.ts b/src/web-rtc-client.ts
index 283ba0b4..9be0cfd8 100644
--- a/src/web-rtc-client.ts
+++ b/src/web-rtc-client.ts
@@ -90,7 +90,7 @@ export default class WebRTCClient extends Emitter {
 
   audio: MediaTrackConstraintSet | boolean | undefined;
 
-  audioElements: Record<string, HTMLAudioElement & { setSinkId?: (id: string) => void }>;
+  audioElements: Record<string, HTMLAudioElement>;
 
   video: MediaTrackConstraintSet | boolean | undefined;
 
@@ -182,7 +182,7 @@ export default class WebRTCClient extends Emitter {
     return [];
   }
 
-  constructor(config: WebRtcConfig, session: WazoSession | null | undefined, uaConfigOverrides?: UserAgentConfigOverrides) {
+  constructor(config: WebRtcConfig, session?: WazoSession, uaConfigOverrides?: UserAgentConfigOverrides) {
     super();
 
     // For debug purpose
@@ -198,7 +198,7 @@ export default class WebRTCClient extends Emitter {
       this.userAgent = this.createUserAgent(uaConfigOverrides);
     });
 
-    this.audioOutputDeviceId = config.audioOutputDeviceId;
+    this.audioOutputDeviceId = config.audioDeviceOutput;
     this.audioOutputVolume = config.audioOutputVolume || 1;
 
     if (config.media) {
@@ -1246,20 +1246,14 @@ export default class WebRTCClient extends Emitter {
     this.audioOutputVolume = volume;
   }
 
-  changeAudioOutputDevice(id: string) {
+  async changeAudioOutputDevice(id: string) {
     logger.info('Changing audio output device', {
       id,
     });
     this.audioOutputDeviceId = id;
-    Object.values(this.audioElements).forEach(audioElement => {
-      // `setSinkId` method is not included in any flow type definitions for HTMLAudioElements but is a valid method
-      // audioElement is an array of HTMLAudioElements, and HTMLAudioElement inherits the method from HTMLMediaElement
-      // https://developer.mozilla.org/en-US/docs/Web/API/HTMLAudioElement
-      // https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/setSinkId
-      if (audioElement.setSinkId) {
-        audioElement.setSinkId(id);
-      }
-    });
+    const promises: Promise<void>[] = [];
+    Object.values(this.audioElements).forEach(audioElement => promises.push(audioElement.setSinkId(id)));
+    await Promise.allSettled(promises);
   }
 
   async changeAudioInputDevice(id: string, session: WazoSession | null | undefined, force?: boolean): Promise<MediaStream | null> {
@@ -1735,7 +1729,7 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  updateLocalStream(sipSessionId: string, newStream: MediaStream): void {
+  async updateLocalStream(sipSessionId: string, newStream: MediaStream): Promise<void> {
     const sipSession = this.sipSessions[sipSessionId];
 
     if (!sipSession) {
@@ -1756,7 +1750,7 @@ export default class WebRTCClient extends Emitter {
     this.setLocalMediaStream(sipSessionId, newStream);
 
     // Update the local video element
-    this._setupMedias(sipSession, newStream);
+    await this._setupMedias(sipSession, newStream);
   }
 
   updateRemoteStream(sessionId: string, allTracks = true): void {
@@ -1832,14 +1826,14 @@ export default class WebRTCClient extends Emitter {
     return sessionId in this.conferences;
   }
 
-  createAudioElementFor(sessionId: string): HTMLAudioElement {
+  async createAudioElementFor(sessionId: string): Promise<HTMLAudioElement> {
     const audio: HTMLAudioElement = document.createElement('audio');
     audio.setAttribute('id', `audio-${sessionId}`);
 
     logger.info('creating audio element', { sessionId, audioOutputDeviceId: this.audioOutputDeviceId });
 
-    if ((audio as any).setSinkId && this.audioOutputDeviceId) {
-      (audio as any).setSinkId(this.audioOutputDeviceId);
+    if (this.audioOutputDeviceId) {
+      await audio.setSinkId(this.audioOutputDeviceId);
     }
 
     if (document.body) {
@@ -2135,7 +2129,7 @@ export default class WebRTCClient extends Emitter {
       oldInviteRequest(request);
     };
 
-    session.delegate.onInvite = (request: IncomingRequestMessage) => {
+    session.delegate.onInvite = async (request: IncomingRequestMessage) => {
       let updatedCalleeName: string | null = null;
       let updatedNumber = null;
 
@@ -2165,14 +2159,14 @@ export default class WebRTCClient extends Emitter {
 
       this.updateRemoteStream(sipSessionId, false);
 
-      this._setupMedias(session);
+      await this._setupMedias(session);
 
       return this.eventEmitter.emit(ON_REINVITE, session, request, updatedCalleeName, updatedNumber, hadRemoteVideo);
     };
   }
 
-  _onProgress(session: WazoSession, early = false): void {
-    this._setupMedias(session);
+  async _onProgress(session: WazoSession, early = false): Promise<void> {
+    await this._setupMedias(session);
 
     const sessionId = this.getSipSessionId(session).substr(0, 20);
     this.updateRemoteStream(sessionId);
@@ -2183,7 +2177,7 @@ export default class WebRTCClient extends Emitter {
     this.eventEmitter.emit(early ? ON_EARLY_MEDIA : ON_PROGRESS, session);
   }
 
-  _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): void {
+  async _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): Promise<void> {
     logger.info('on call accepted', {
       id: session.id,
       clientId: this.clientId,
@@ -2191,7 +2185,7 @@ export default class WebRTCClient extends Emitter {
     });
     this.storeSipSession(session);
 
-    this._setupMedias(session);
+    await this._setupMedias(session);
 
     this.updateRemoteStream(this.getSipSessionId(session), initAllTracks);
     const pc = (session.sessionDescriptionHandler as SessionDescriptionHandler)?.peerConnection;
@@ -2253,7 +2247,7 @@ export default class WebRTCClient extends Emitter {
     return Boolean(session.sessionDescriptionHandlerModifiersReInvite.find(modifier => modifier === stripVideo));
   }
 
-  _setupMedias(session: WazoSession, newStream: MediaStream | null | undefined = null): void {
+  async _setupMedias(session: WazoSession, newStream: MediaStream | null | undefined = null): Promise<void> {
     if (!this._isWeb()) {
       logger.info('Setup media on mobile, no need to setup html element, bailing');
       return;
@@ -2271,7 +2265,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
-      this.createAudioElementFor(sessionId);
+      await this.createAudioElementFor(sessionId);
     }
 
     const audioElement = this.audioElements[sessionId];