@wazo/sdk 0.43.8 → 0.43.9

package/src/domain/Phone/WebRTCPhone.ts

@@ -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/web-rtc-client.ts

@@ -42,7 +42,6 @@ export const replaceLocalIpModifier = (description: Record<string, any>) => Prom
   ...JSON.parse(JSON.stringify(description)),
   sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1'),
 });
-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'];
@@ -268,7 +267,7 @@ export default class WebRTCClient extends Emitter {
         logger.info('sdk webrtc on invite', {
           method: 'delegate.onInvite',
           clientId: this.clientId,
-          id: invitation.id,
+          id: this.getSipSessionId(invitation),
           remoteURI: (invitation as any).remoteURI,
         });
 
@@ -604,7 +603,7 @@ export default class WebRTCClient extends Emitter {
         },
         onReject: (response: IncomingResponse) => {
           logger.info('on call rejected', {
-            id: session?.id,
+            id: this.getSipSessionId(session),
             // @ts-ignore: fromTag does not exist
             fromTag: session?.fromTag,
           });
@@ -641,7 +640,7 @@ 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: this.getSipSessionId(session),
       enableVideo,
     });
 
@@ -659,7 +658,7 @@ 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: this.getSipSessionId(session),
         });
         this.onCallEnded(session);
         throw new CanceledCallError(message);
@@ -669,7 +668,7 @@ 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 ${this.getSipSessionId(session)}`, e);
       throw e;
     });
   }
@@ -747,7 +746,7 @@ 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: this.getSipSessionId(session),
     });
 
     try {
@@ -819,7 +818,7 @@ export default class WebRTCClient extends Emitter {
 
   mute(session: WazoSession): void {
     logger.info('sdk webrtc mute', {
-      id: session.id,
+      id: this.getSipSessionId(session),
     });
 
     this._toggleAudio(session, true);
@@ -827,7 +826,7 @@ export default class WebRTCClient extends Emitter {
 
   unmute(session: WazoSession): void {
     logger.info('sdk webrtc unmute', {
-      id: session.id,
+      id: this.getSipSessionId(session),
     });
 
     this._toggleAudio(session, false);
@@ -872,7 +871,7 @@ export default class WebRTCClient extends Emitter {
 
   toggleCameraOn(session: WazoSession): void {
     logger.info('sdk webrtc toggle camera on', {
-      id: session.id,
+      id: this.getSipSessionId(session),
     });
 
     this._toggleVideo(session, false);
@@ -880,7 +879,7 @@ export default class WebRTCClient extends Emitter {
 
   toggleCameraOff(session: WazoSession): void {
     logger.info('sdk webrtc toggle camera off', {
-      id: session.id,
+      id: this.getSipSessionId(session),
     });
 
     this._toggleVideo(session, true);
@@ -1178,7 +1177,7 @@ export default class WebRTCClient extends Emitter {
     }
 
     logger.info('send WebRTC message', {
-      sipId: sipSession.id,
+      sipId: this.getSipSessionId(sipSession),
       contentType,
     });
 
@@ -1611,20 +1610,21 @@ export default class WebRTCClient extends Emitter {
       return '';
     }
 
-    const message = sipSession.message as Partial<{ callId?: string }>;
-    if (message?.callId) {
-      return message.callId.substring(0, SIP_ID_LENGTH);
-    }
-
     // For Inviter
     // @ts-ignore: private
     if (sipSession instanceof 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 Invitation && sipSession.incomingInviteRequest.message) {
+      // @ts-ignore: private
+      return sipSession.incomingInviteRequest.message.callId;
+    }
+
+    return '';
   }
 
   async waitForRegister(): Promise<void> {
@@ -1704,7 +1704,7 @@ export default class WebRTCClient extends Emitter {
   }
 
   getSipSession(id: string): WazoSession | null | undefined {
-    return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
+    return id in this.sipSessions ? this.sipSessions[id] : null;
   }
 
   getSipSessionIds(): string[] {
@@ -2168,7 +2168,7 @@ export default class WebRTCClient extends Emitter {
   async _onProgress(session: WazoSession, early = false): Promise<void> {
     await this._setupMedias(session);
 
-    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    const sessionId = this.getSipSessionId(session);
     this.updateRemoteStream(sessionId);
     logger.info('progress received', {
       sessionId,
@@ -2179,7 +2179,7 @@ export default class WebRTCClient extends Emitter {
 
   async _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent = true, initAllTracks = true): Promise<void> {
     logger.info('on call accepted', {
-      id: session.id,
+      id: this.getSipSessionId(session),
       clientId: this.clientId,
       remoteTag: session.remoteTag,
     });
@@ -2610,20 +2610,20 @@ 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: this.getSipSessionId(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: 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);
     }
 
@@ -2636,13 +2636,13 @@ 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: 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;
     }
 
@@ -2655,7 +2655,7 @@ 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: this.getSipSessionId(session) });
       return;
     }
 
@@ -2668,7 +2668,7 @@ 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: this.getSipSessionId(session) });
       return null;
     }