@wazo/sdk 1.0.0-alpha.1 → 1.0.0-alpha.10

package/src/web-rtc-client.ts

@@ -29,9 +29,9 @@ import Emitter from './utils/Emitter';
 import ApiClient from './api-client';
 import IssueReporter from './service/IssueReporter';
 import Heartbeat from './utils/Heartbeat';
-import { getSipSessionId, makeURI, replaceLocalIpModifier, SIP_ID_LENGTH } from './utils/sdp';
-import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, IncomingResponse, PeerConnection, WazoSession, WazoTransport } from './domain/types';
-import { cleanupStream, createUaOptions, fetchNetworkStats, getAudioConstraints, getLocalStream, getLocalTracks, getLocalVideoStream, getMediaConfiguration, getPeerConnection, getRemoteStream, getRemoteTracks, getRemoteVideoStream, getRemoteVideoStreamFromPc, getStreamFromConstraints, getVideoConstraints, hasALocalVideoTrack, hasARemoteVideoTrack, hasAVideoTrack, hasLocalVideo, hasRemoteVideo, hasVideoTrack, isAudioOnly, isVideoRemotelyHeld, isWeb, sessionWantsToDoVideo, setLocalMediaStream, toggleAudio, toggleVideo, updateRemoteStream } from './utils/webrtc';
+import { getSipCallId, makeURI, replaceLocalIpModifier } from './utils/sdp';
+import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, IncomingResponse, PeerConnection, SipCall, WazoTransport } from './domain/types';
+import { cleanupStream, createUaOptions, fetchNetworkStats, getAudioConstraints, getLocalStream, getLocalTracks, getLocalVideoStream, getMediaConfiguration, getPeerConnection, getRemoteStream, getRemoteTracks, getRemoteVideoStream, getRemoteVideoStreamFromPc, getVideoConstraints, hasALocalVideoTrack, hasARemoteVideoTrack, hasAVideoTrack, hasLocalVideo, hasRemoteVideo, hasVideoTrack, isAudioOnly, isVideoRemotelyHeld, isWeb, sessionWantsToDoVideo, setLocalMediaStream, toggleAudio, toggleVideo, updateRemoteStream } from './utils/webrtc';
 
 const SEND_STATS_DELAY = 5000;
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
@@ -39,6 +39,7 @@ const logger = IssueReporter ? IssueReporter.loggerFor('webrtc-client') : consol
 const statsLogger = IssueReporter ? IssueReporter.loggerFor('webrtc-stats') : console;
 // events
 const REGISTERED = 'registered';
+const REGISTERING = 'registering';
 const UNREGISTERED = 'unregistered';
 const REGISTRATION_FAILED = 'registrationFailed';
 const INVITE = 'invite';
@@ -57,7 +58,7 @@ const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_DISCONNECTED = 'onDisconnected';
 
-export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
+export const events = [REGISTERED, REGISTERING, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {}
 
@@ -102,7 +103,7 @@ export default class WebRTCClient extends Emitter {
 
   statsIntervals: Record<string, any>;
 
-  sipSessions: Record<string, WazoSession>;
+  sipSessions: Record<string, SipCall>;
 
   conferences: Record<string, boolean>;
 
@@ -119,6 +120,8 @@ export default class WebRTCClient extends Emitter {
 
   REGISTERED: string;
 
+  REGISTERING: string;
+
   UNREGISTERED: string;
 
   REGISTRATION_FAILED: string;
@@ -153,7 +156,7 @@ export default class WebRTCClient extends Emitter {
 
   ON_DISCONNECTED: string;
 
-  constructor(config: WebRtcConfig, session?: WazoSession, uaConfigOverrides?: UserAgentConfigOverrides) {
+  constructor(config: WebRtcConfig, session?: SipCall, uaConfigOverrides?: UserAgentConfigOverrides) {
     super();
 
     // For debug purpose
@@ -166,7 +169,9 @@ export default class WebRTCClient extends Emitter {
 
     this._buildConfig(config, session).then((newConfig: WebRtcConfig) => {
       this.config = newConfig;
-      this.userAgent = this.createUserAgent(uaConfigOverrides);
+      if (!this.userAgent) {
+        this.userAgent = this.createUserAgent(uaConfigOverrides);
+      }
     });
 
     this.audioOutputDeviceId = config.audioDeviceOutput;
@@ -194,6 +199,7 @@ export default class WebRTCClient extends Emitter {
     this.heartbeat.setOnHeartbeatTimeout(this._onHeartbeatTimeout.bind(this));
     // sugar
     this.REGISTERED = REGISTERED;
+    this.REGISTERING = REGISTERING;
     this.UNREGISTERED = UNREGISTERED;
     this.REGISTRATION_FAILED = REGISTRATION_FAILED;
     this.INVITE = INVITE;
@@ -239,7 +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,
         });
 
@@ -316,7 +322,7 @@ export default class WebRTCClient extends Emitter {
         this.registerer.waitingToggle(false);
       }
 
-      this.eventEmitter.emit(UNREGISTERED);
+      this.eventEmitter.emit(UNREGISTERED, error);
     }
   }
 
@@ -341,6 +347,8 @@ export default class WebRTCClient extends Emitter {
 
     logger.info('sdk webrtc registering...', logInfo);
 
+    this.eventEmitter.emit(REGISTERING);
+
     if (!this.userAgent) {
       logger.info('sdk webrtc recreating User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
@@ -521,7 +529,7 @@ export default class WebRTCClient extends Emitter {
     });
   }
 
-  call(number: string, enableVideo?: boolean, audioOnly = false, conference = false, options = {}): WazoSession | null {
+  call(number: string, enableVideo?: boolean, audioOnly = false, conference = false, options = {}): SipCall | null {
     logger.info('sdk webrtc creating call', {
       clientId: this.clientId,
       number,
@@ -544,7 +552,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     const uri = makeURI(number, this.config.host);
-    let session: WazoSession | null = null;
+    let session: SipCall | null = null;
 
     if (uri) {
       session = this.userAgent ? new Inviter(this.userAgent, uri, inviterOptions) : null;
@@ -559,7 +567,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     if (conference) {
-      this.conferences[getSipSessionId(session)] = true;
+      this.conferences[getSipCallId(session)] = true;
     }
 
     const inviteOptions: InviterInviteOptions = {
@@ -568,21 +576,21 @@ export default class WebRTCClient extends Emitter {
           if ((session?.sessionDescriptionHandler as any)?.peerConnection) {
             (session?.sessionDescriptionHandler as any).peerConnection.sfu = conference;
           }
-          this._onAccepted(session as WazoSession, response.session, true);
+          this._onAccepted(session as SipCall, response.session, true);
         },
         onProgress: (payload: IncomingResponse) => {
           this._onProgress(payload.session, payload.message.statusCode === 183);
         },
         onReject: (response: IncomingResponse) => {
           logger.info('on call rejected', {
-            id: session?.id,
+            id: getSipCallId(session),
             // @ts-ignore: fromTag does not exist
             fromTag: session?.fromTag,
           });
 
-          this._stopSendingStats(session as WazoSession);
+          this._stopSendingStats(session as SipCall);
 
-          this.stopNetworkMonitoring(session as WazoSession);
+          this.stopNetworkMonitoring(session as SipCall);
           this.eventEmitter.emit(REJECTED, session, response);
         },
       },
@@ -612,7 +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,
     });
 
@@ -630,7 +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);
@@ -640,12 +648,12 @@ 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;
     });
   }
 
-  async hangup(session: WazoSession): Promise<OutgoingByeRequest | null> {
+  async hangup(session: SipCall): Promise<OutgoingByeRequest | null> {
     const { state, id } = session;
     logger.info('sdk webrtc hangup call', { clientId: this.clientId, id, state });
 
@@ -654,7 +662,7 @@ export default class WebRTCClient extends Emitter {
 
       this._cleanupMedia(session);
 
-      delete this.sipSessions[getSipSessionId(session)];
+      delete this.sipSessions[getSipCallId(session)];
       // Check if Invitation or Inviter (Invitation = incoming call)
       const isInviter = session instanceof Inviter;
 
@@ -679,8 +687,8 @@ export default class WebRTCClient extends Emitter {
   }
 
   // Fetch and emit an event at `interval` with session network stats
-  startNetworkMonitoring(session: WazoSession, interval = 1000): void {
-    const sessionId = getSipSessionId(session);
+  startNetworkMonitoring(session: SipCall, interval = 1000): void {
+    const sessionId = getSipCallId(session);
     logger.info('starting network inspection', {
       id: sessionId,
       clientId: this.clientId,
@@ -689,8 +697,8 @@ export default class WebRTCClient extends Emitter {
     this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
   }
 
-  stopNetworkMonitoring(session: WazoSession): void {
-    const sessionId = getSipSessionId(session);
+  stopNetworkMonitoring(session: SipCall): void {
+    const sessionId = getSipCallId(session);
     const exists = (sessionId in this.networkMonitoringInterval);
     logger.info('stopping network inspection', {
       clientId: this.clientId,
@@ -705,10 +713,10 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  async reject(session: WazoSession): Promise<void> {
+  async reject(session: SipCall): Promise<void> {
     logger.info('sdk webrtc reject call', {
       clientId: this.clientId,
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     try {
@@ -769,23 +777,23 @@ export default class WebRTCClient extends Emitter {
     logger.info('sdk webrtc client closed', { clientId: this.clientId });
   }
 
-  mute(session: WazoSession): void {
+  mute(session: SipCall): void {
     logger.info('sdk webrtc mute', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleAudio(session, true);
   }
 
-  unmute(session: WazoSession): void {
+  unmute(session: SipCall): void {
     logger.info('sdk webrtc unmute', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleAudio(session, false);
   }
 
-  isAudioMuted(session: WazoSession): boolean {
+  isAudioMuted(session: SipCall): boolean {
     if (!session || !session.sessionDescriptionHandler) {
       return false;
     }
@@ -822,24 +830,24 @@ export default class WebRTCClient extends Emitter {
     return muted;
   }
 
-  toggleCameraOn(session: WazoSession): void {
+  toggleCameraOn(session: SipCall): void {
     logger.info('sdk webrtc toggle camera on', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleVideo(session, false);
   }
 
-  toggleCameraOff(session: WazoSession): void {
+  toggleCameraOff(session: SipCall): void {
     logger.info('sdk webrtc toggle camera off', {
-      id: session.id,
+      id: getSipCallId(session),
     });
 
     toggleVideo(session, true);
   }
 
-  hold(session: WazoSession, isConference = false, hadVideo = false): Promise<OutgoingInviteRequest | void> {
-    const sessionId = getSipSessionId(session);
+  hold(session: SipCall, isConference = false, hadVideo = false): Promise<OutgoingInviteRequest | void> {
+    const sessionId = getSipCallId(session);
     const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
     logger.info('sdk webrtc hold', {
       sessionId,
@@ -896,8 +904,8 @@ export default class WebRTCClient extends Emitter {
     });
   }
 
-  unhold(session: WazoSession, isConference = false): Promise<OutgoingInviteRequest | void> {
-    const sessionId = getSipSessionId(session);
+  unhold(session: SipCall, isConference = false): Promise<OutgoingInviteRequest | void> {
+    const sessionId = getSipCallId(session);
     const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
     logger.info('sdk webrtc unhold', {
       sessionId,
@@ -915,7 +923,7 @@ export default class WebRTCClient extends Emitter {
 
     this.unmute(session);
 
-    delete this.heldSessions[getSipSessionId(session)];
+    delete this.heldSessions[getSipCallId(session)];
     session.sessionDescriptionHandlerOptionsReInvite = {
       hold: false,
       conference: isConference,
@@ -943,21 +951,21 @@ export default class WebRTCClient extends Emitter {
     return this.heldSessions[sessionId];
   }
 
-  isCallHeld(session: WazoSession): boolean {
-    return getSipSessionId(session) in this.heldSessions;
+  isCallHeld(session: SipCall): boolean {
+    return getSipCallId(session) in this.heldSessions;
   }
 
   isVideoRemotelyHeld(sessionId: string): boolean {
     return isVideoRemotelyHeld(this.heldSessions[sessionId]);
   }
 
-  sendDTMF(session: WazoSession, tone: string): boolean {
+  sendDTMF(session: SipCall, tone: string): boolean {
     if (!session.sessionDescriptionHandler) {
       return false;
     }
 
     logger.info('Sending DTMF', {
-      id: getSipSessionId(session),
+      id: getSipCallId(session),
       tone,
     });
     return session.sessionDescriptionHandler.sendDtmf(tone);
@@ -973,10 +981,10 @@ export default class WebRTCClient extends Emitter {
     messager.message();
   }
 
-  transfer(session: WazoSession, target: string): void {
+  transfer(session: SipCall, target: string): void {
     this.hold(session);
     logger.info('Transfering a session', {
-      id: getSipSessionId(session),
+      id: getSipCallId(session),
       target,
     });
     const options = {
@@ -997,24 +1005,24 @@ export default class WebRTCClient extends Emitter {
   }
 
   // check https://sipjs.com/api/0.12.0/refer/referClientContext/
-  atxfer(session: WazoSession): Record<string, any> {
+  atxfer(session: SipCall): Record<string, any> {
     this.hold(session);
     logger.info('webrtc transfer started', {
-      id: getSipSessionId(session),
+      id: getSipCallId(session),
     });
     const result: Record<string, any> = {
       newSession: null,
       init: async (target: string) => {
         logger.info('webrtc transfer initialized', {
-          id: getSipSessionId(session),
+          id: getSipCallId(session),
           target,
         });
         result.newSession = await this.call(target);
       },
       complete: () => {
         logger.info('webrtc transfer completed', {
-          id: getSipSessionId(session),
-          referId: getSipSessionId(result.newSession),
+          id: getSipCallId(session),
+          referId: getSipCallId(result.newSession),
         });
         session.refer(result.newSession);
       },
@@ -1027,13 +1035,13 @@ export default class WebRTCClient extends Emitter {
   }
 
   // eslint-disable-next-line @typescript-eslint/default-param-last
-  sendMessage(sipSession: WazoSession | null = null, body: string, contentType = 'text/plain'): void {
+  sendMessage(sipSession: SipCall | null = null, body: string, contentType = 'text/plain'): void {
     if (!sipSession) {
       return;
     }
 
     logger.info('send WebRTC message', {
-      sipId: sipSession.id,
+      sipId: getSipCallId(sipSession),
       contentType,
     });
 
@@ -1107,7 +1115,7 @@ export default class WebRTCClient extends Emitter {
     await Promise.allSettled(promises);
   }
 
-  async changeAudioInputDevice(id: string, session: WazoSession | null | undefined, force?: boolean): Promise<MediaStream | null> {
+  async changeAudioInputDevice(id: string, session: SipCall | null | undefined, force?: boolean): Promise<MediaStream | null> {
     const currentId = this.getAudioDeviceId();
     logger.info('setting audio input device', {
       id,
@@ -1178,7 +1186,7 @@ export default class WebRTCClient extends Emitter {
     return null;
   }
 
-  async changeVideoInputDevice(id: string, session?: WazoSession): Promise<MediaStream | void> {
+  async changeVideoInputDevice(id: string, session?: SipCall): Promise<MediaStream | void> {
     this.setVideoInputDevice(id);
 
     if (session) {
@@ -1206,14 +1214,14 @@ export default class WebRTCClient extends Emitter {
     };
   }
 
-  changeSessionVideoInputDevice(id: string | null | undefined, session: WazoSession): Promise<MediaStream | void> {
+  changeSessionVideoInputDevice(id: string | null | undefined, session: SipCall): Promise<MediaStream | void> {
     if (!sessionWantsToDoVideo(session)) {
       return Promise.resolve();
     }
 
     const sdh = session.sessionDescriptionHandler as SessionDescriptionHandler;
     const pc = sdh.peerConnection;
-    const sessionId = getSipSessionId(session);
+    const sessionId = getSipCallId(session);
     const localStream = this.getLocalStream(sessionId);
     logger.info('changing video input device', {
       id,
@@ -1270,7 +1278,7 @@ export default class WebRTCClient extends Emitter {
     return this.video && typeof this.video === 'object' && 'deviceId' in this.video ? (this.video.deviceId as any)?.exact : undefined;
   }
 
-  reinvite(sipSession: WazoSession, newConstraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
+  reinvite(sipSession: SipCall, newConstraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
     if (!sipSession) {
       return Promise.resolve();
     }
@@ -1286,7 +1294,7 @@ export default class WebRTCClient extends Emitter {
     const desktop = newConstraints && newConstraints.desktop;
     logger.info('Sending reinvite', {
       clientId: this.clientId,
-      id: getSipSessionId(sipSession),
+      id: getSipCallId(sipSession),
       newConstraints,
       conference,
       audioOnly,
@@ -1324,7 +1332,7 @@ export default class WebRTCClient extends Emitter {
           }
 
           logger.info('on re-INVITE accepted', {
-            id: getSipSessionId(sipSession),
+            id: getSipCallId(sipSession),
             wasMuted,
             shouldDoScreenSharing,
           });
@@ -1461,10 +1469,10 @@ export default class WebRTCClient extends Emitter {
     this.heartbeatCb = cb;
   }
 
-  onCallEnded(session: WazoSession): void {
+  onCallEnded(session: SipCall): void {
     this._cleanupMedia(session);
 
-    delete this.sipSessions[getSipSessionId(session)];
+    delete this.sipSessions[getSipCallId(session)];
 
     this._stopSendingStats(session);
 
@@ -1482,18 +1490,18 @@ export default class WebRTCClient extends Emitter {
     this.userAgent.attemptReconnection();
   }
 
-  storeSipSession(session: WazoSession): void {
-    const id = getSipSessionId(session);
+  storeSipSession(session: SipCall): void {
+    const id = getSipCallId(session);
     logger.info('storing sip session', { id, clientId: this.clientId });
 
     this.sipSessions[id] = session;
   }
 
-  getSipSession(id: string): WazoSession | null | undefined {
-    return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
+  getSipSession(id: string): SipCall | null | undefined {
+    return id in this.sipSessions ? this.sipSessions[id] : null;
   }
 
-  getSipSessionIds(): string[] {
+  getSipCallIds(): string[] {
     return Object.keys(this.sipSessions);
   }
 
@@ -1545,8 +1553,8 @@ export default class WebRTCClient extends Emitter {
     return audio;
   }
 
-  _onTransportError(): void {
-    logger.error('on transport error');
+  _onTransportError(reason?: string): void {
+    logger.warn('on transport error', { reason });
     this.eventEmitter.emit(TRANSPORT_ERROR);
     this.attemptReconnection();
   }
@@ -1608,7 +1616,7 @@ export default class WebRTCClient extends Emitter {
       }
 
       // We can invoke disconnect() with an error that can be catcher by `onDisconnect`, so we have to trigger it here.
-      this._onTransportError();
+      this._onTransportError('heartbeat timeout');
     }
   }
 
@@ -1664,7 +1672,7 @@ export default class WebRTCClient extends Emitter {
     return this.connectionPromise;
   }
 
-  _buildConfig(config: WebRtcConfig, session: WazoSession | null | undefined): Promise<WebRtcConfig> {
+  _buildConfig(config: WebRtcConfig, session: SipCall | null | undefined): Promise<WebRtcConfig> {
     // If no session provided, return the configuration directly
     if (!session) {
       return new Promise(resolve => resolve(config));
@@ -1685,8 +1693,8 @@ export default class WebRTCClient extends Emitter {
   }
 
   // Invitation and Inviter extends Session
-  _setupSession(session: WazoSession): void {
-    const sipSessionId = getSipSessionId(session);
+  _setupSession(session: SipCall): void {
+    const sipSessionId = getSipCallId(session);
 
     // When receiving an Invitation, the delegate is not defined.
     if (!session.delegate) {
@@ -1758,21 +1766,20 @@ export default class WebRTCClient extends Emitter {
     };
   }
 
-  async _onProgress(session: WazoSession, early = false): Promise<void> {
+  async _onProgress(session: SipCall, early = false): Promise<void> {
     await this._setupMedias(session);
 
-    const sessionId = getSipSessionId(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> {
+  async _onAccepted(session: SipCall, 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,
     });
@@ -1822,14 +1829,14 @@ export default class WebRTCClient extends Emitter {
     this._startSendingStats(session);
   }
 
-  async _setupMedias(session: WazoSession, newStream: MediaStream | null | undefined = null): Promise<void> {
+  async _setupMedias(session: SipCall, newStream: MediaStream | null | undefined = null): Promise<void> {
     if (!isWeb()) {
       logger.info('Setup media on mobile, no need to setup html element, bailing');
       return;
     }
 
     // Safari hack, because you cannot call .play() from a non user action
-    const sessionId = getSipSessionId(session);
+    const sessionId = getSipCallId(session);
     const isConference = this.isConference(sessionId);
 
     if (sessionId in this.audioElements) {
@@ -1883,8 +1890,8 @@ export default class WebRTCClient extends Emitter {
     audioElement.play().catch(() => {});
   }
 
-  _cleanupMedia(session?: WazoSession): void {
-    const sessionId = getSipSessionId(session);
+  _cleanupMedia(session?: SipCall): void {
+    const sessionId = getSipCallId(session);
     const localStream = this.getLocalStream(sessionId);
 
     if (localStream) {
@@ -1910,7 +1917,7 @@ export default class WebRTCClient extends Emitter {
 
     if (this._hasAudio() && isWeb()) {
       if (session) {
-        cleanLocalElement(getSipSessionId(session));
+        cleanLocalElement(getSipCallId(session));
       } else {
         Object.keys(this.audioElements).forEach(id => cleanLocalElement(id));
       }
@@ -1925,7 +1932,7 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  _startSendingStats(session: WazoSession): void {
+  _startSendingStats(session: SipCall): void {
     const sdh = session.sessionDescriptionHandler as SessionDescriptionHandler;
     const pc = sdh.peerConnection;
 
@@ -1933,7 +1940,7 @@ export default class WebRTCClient extends Emitter {
       return;
     }
 
-    const sessionId = getSipSessionId(session);
+    const sessionId = getSipCallId(session);
     getStats(pc, (result: Record<string, any>) => {
       const {
         results,
@@ -1949,8 +1956,8 @@ export default class WebRTCClient extends Emitter {
     }, SEND_STATS_DELAY);
   }
 
-  _stopSendingStats(session: WazoSession): void {
-    const sessionId = getSipSessionId(session);
+  _stopSendingStats(session: SipCall): void {
+    const sessionId = getSipCallId(session);
     logger.trace('Check for stopping stats', {
       sessionId,
       ids: Object.keys(this.statsIntervals),
@@ -2010,20 +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);
     }
 
@@ -2036,13 +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;
     }
 
@@ -2055,7 +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;
     }
 
@@ -2066,9 +2073,9 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  async _bye(session: WazoSession, options: SessionByeOptions = {}) {
+  async _bye(session: SipCall, 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;
     }