@wazo/sdk 1.0.0-alpha.4 → 1.0.0-alpha.6

package/esm/voice/call.js

@@ -8,7 +8,7 @@ import newFrom from '../utils/new-from';
 import updateFrom from '../utils/update-from';
 import IssueReporter from '../service/IssueReporter';
 import softphone, { EVENT_TERMINATE_SOUND } from './softphone';
-import { getCallDisplayName, getCallNumber, getSipSessionId } from '../utils/sdp';
+import { getCallDisplayName, getCallNumber, getSipCallId } from '../utils/sdp';
 import { assertCan, assertState, getState } from '../state-machine/utils';
 import { downgradeToAudio, getLocalStream, getLocalVideoStream, getPeerConnection, getRemoteStream, getRemoteVideoStream, getStreamFromConstraints, getWebRtcStats, hasALocalVideoTrack, hasLocalVideo, hasRemoteVideo, isVideoMuted, isVideoRemotelyHeld, setLocalMediaStream, toggleVideo, upgradeToVideo } from '../utils/webrtc';
 const logger = IssueReporter.loggerFor('call');
@@ -254,7 +254,7 @@ class Call extends EventEmitter {
         return this.sipCall.state === SessionState.Establishing;
     }
     isEstablished() {
-        return States.ESTABLISHED in this.state;
+        return typeof this.state === 'object' && States.ESTABLISHED in this.state;
     }
     isRinging() {
         return this.state === States.RINGING;
@@ -461,10 +461,10 @@ class Call extends EventEmitter {
         return otherCall && this.id && this.id === otherCall.id;
     }
     shouldAutoAnswer() {
-        return !!this.sipCall.request.getHeader('alert-info');
+        return !!this.sipCall.request?.getHeader('alert-info');
     }
     get id() {
-        return getSipSessionId(this.sipCall);
+        return getSipCallId(this.sipCall);
     }
     get state() {
         return getState(this.callActor);

package/esm/voice/softphone.js

@@ -9,6 +9,7 @@ import IssueReporter from '../service/IssueReporter';
 import configureLogger from '../utils/sip-logger';
 import Call from './call';
 import { assertCan, can, getState, waitUntilState } from '../state-machine/utils';
+import { getSipCallId } from '../utils/sdp';
 export const EVENT_INCOMING = 'incoming';
 export const EVENT_OUTGOING = 'outgoing';
 export const EVENT_HEARTBEAT = 'heartbeat';
@@ -125,7 +126,7 @@ export class Softphone extends EventEmitter {
             logger.info('softphone - calling when registering, waiting for Softphone to be registered ...');
             await waitUntilState(this.softphoneActor, States.REGISTERED);
         }
-        if (this.state !== States.UNREGISTERED) {
+        if (this.state !== States.REGISTERED) {
             throw new InvalidState(`Invalid state ${States.UNREGISTERED}`, States.UNREGISTERED);
         }
         // Hold other calls
@@ -134,6 +135,7 @@ export class Softphone extends EventEmitter {
         try {
             sipCall = this.client.call(options.params.To, options.withCamera, options.audioOnly, options.conference, options);
             const call = new Call(sipCall, this);
+            this.calls.push(call);
             this.emit(EVENT_OUTGOING, call);
             return call;
         }
@@ -289,7 +291,7 @@ export class Softphone extends EventEmitter {
         });
         this.client.on(this.client.ON_REINVITE, (sipCall, request, updatedCalleeName, updatedNumber, hadRemoteVideo) => {
             logger.info('softphone - on reinvite', {
-                callId: sipCall.id,
+                callId: getSipCallId(sipCall),
                 inviteId: request.callId,
                 updatedCalleeName,
                 updatedNumber,
@@ -387,13 +389,16 @@ export class Softphone extends EventEmitter {
         this.softphoneActor.send({ type: action });
     }
     _getCall(sipCall) {
-        return this.calls.find(call => call.id === sipCall.id);
+        return this.calls.find(call => call.id === getSipCallId(sipCall));
+    }
+    _getCallIds() {
+        return this.calls.map(call => call.id);
     }
     _onCallEvent(sipCall, eventName, ...args) {
-        logger.warn('softphone - received call event', { eventName, sipId: sipCall.id });
+        logger.info('softphone - received call event', { eventName, sipId: getSipCallId(sipCall) });
         const call = this._getCall(sipCall);
         if (!call) {
-            logger.warn('softphone - no call found to send the event', { eventName, sipId: sipCall.id });
+            logger.warn('softphone - no call found to send the event', { eventName, sipId: getSipCallId(sipCall), ids: this._getCallIds() });
             return;
         }
         // @ts-ignore

package/esm/web-rtc-client.js

@@ -17,7 +17,7 @@ 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 { getSipCallId, makeURI, replaceLocalIpModifier, SIP_ID_MIN_LENGTH } from './utils/sdp';
 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'];
@@ -103,7 +103,9 @@ export default class WebRTCClient extends Emitter {
         this.skipRegister = config.skipRegister;
         this._buildConfig(config, session).then((newConfig) => {
             this.config = newConfig;
-            this.userAgent = this.createUserAgent(uaConfigOverrides);
+            if (!this.userAgent) {
+                this.userAgent = this.createUserAgent(uaConfigOverrides);
+            }
         });
         this.audioOutputDeviceId = config.audioDeviceOutput;
         this.audioOutputVolume = config.audioOutputVolume || 1;
@@ -230,7 +232,7 @@ export default class WebRTCClient extends Emitter {
                 // @ts-ignore: private
                 this.registerer.waitingToggle(false);
             }
-            this.eventEmitter.emit(UNREGISTERED);
+            this.eventEmitter.emit(UNREGISTERED, error);
         }
     }
     async register(tries = 0) {
@@ -435,7 +437,7 @@ export default class WebRTCClient extends Emitter {
             this._setupSession(session);
         }
         if (conference) {
-            this.conferences[getSipSessionId(session)] = true;
+            this.conferences[getSipCallId(session)] = true;
         }
         const inviteOptions = {
             requestDelegate: {
@@ -514,7 +516,7 @@ export default class WebRTCClient extends Emitter {
         try {
             this._stopSendingStats(session);
             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;
             // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
@@ -536,7 +538,7 @@ export default class WebRTCClient extends Emitter {
     }
     // Fetch and emit an event at `interval` with session network stats
     startNetworkMonitoring(session, interval = 1000) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         logger.info('starting network inspection', {
             id: sessionId,
             clientId: this.clientId,
@@ -545,7 +547,7 @@ export default class WebRTCClient extends Emitter {
         this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
     }
     stopNetworkMonitoring(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const exists = (sessionId in this.networkMonitoringInterval);
         logger.info('stopping network inspection', {
             clientId: this.clientId,
@@ -669,7 +671,7 @@ export default class WebRTCClient extends Emitter {
         toggleVideo(session, true);
     }
     hold(session, isConference = false, hadVideo = false) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
         logger.info('sdk webrtc hold', {
             sessionId,
@@ -717,7 +719,7 @@ export default class WebRTCClient extends Emitter {
         });
     }
     unhold(session, isConference = false) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
         logger.info('sdk webrtc unhold', {
             sessionId,
@@ -732,7 +734,7 @@ export default class WebRTCClient extends Emitter {
             return Promise.resolve();
         }
         this.unmute(session);
-        delete this.heldSessions[getSipSessionId(session)];
+        delete this.heldSessions[getSipCallId(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             hold: false,
             conference: isConference,
@@ -757,7 +759,7 @@ export default class WebRTCClient extends Emitter {
         return this.heldSessions[sessionId];
     }
     isCallHeld(session) {
-        return getSipSessionId(session) in this.heldSessions;
+        return getSipCallId(session) in this.heldSessions;
     }
     isVideoRemotelyHeld(sessionId) {
         return isVideoRemotelyHeld(this.heldSessions[sessionId]);
@@ -767,7 +769,7 @@ export default class WebRTCClient extends Emitter {
             return false;
         }
         logger.info('Sending DTMF', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
             tone,
         });
         return session.sessionDescriptionHandler.sendDtmf(tone);
@@ -784,7 +786,7 @@ export default class WebRTCClient extends Emitter {
     transfer(session, target) {
         this.hold(session);
         logger.info('Transfering a session', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
             target,
         });
         const options = {
@@ -807,21 +809,21 @@ export default class WebRTCClient extends Emitter {
     atxfer(session) {
         this.hold(session);
         logger.info('webrtc transfer started', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
         });
         const result = {
             newSession: null,
             init: async (target) => {
                 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);
             },
@@ -988,7 +990,7 @@ export default class WebRTCClient extends Emitter {
         }
         const sdh = session.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,
@@ -1049,7 +1051,7 @@ 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,
@@ -1083,7 +1085,7 @@ export default class WebRTCClient extends Emitter {
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
-                        id: getSipSessionId(sipSession),
+                        id: getSipCallId(sipSession),
                         wasMuted,
                         shouldDoScreenSharing,
                     });
@@ -1194,7 +1196,7 @@ export default class WebRTCClient extends Emitter {
     }
     onCallEnded(session) {
         this._cleanupMedia(session);
-        delete this.sipSessions[getSipSessionId(session)];
+        delete this.sipSessions[getSipCallId(session)];
         this._stopSendingStats(session);
         this.stopNetworkMonitoring(session);
     }
@@ -1207,14 +1209,14 @@ export default class WebRTCClient extends Emitter {
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
-        const id = getSipSessionId(session);
+        const id = getSipCallId(session);
         logger.info('storing sip session', { id, clientId: this.clientId });
         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.substring(0, SIP_ID_MIN_LENGTH)] : null;
     }
-    getSipSessionIds() {
+    getSipCallIds() {
         return Object.keys(this.sipSessions);
     }
     async updateLocalStream(sipSessionId, newStream) {
@@ -1369,7 +1371,7 @@ export default class WebRTCClient extends Emitter {
     }
     // Invitation and Inviter extends Session
     _setupSession(session) {
-        const sipSessionId = getSipSessionId(session);
+        const sipSessionId = getSipCallId(session);
         // When receiving an Invitation, the delegate is not defined.
         if (!session.delegate) {
             session.delegate = {};
@@ -1432,7 +1434,7 @@ export default class WebRTCClient extends Emitter {
     }
     async _onProgress(session, early = false) {
         await this._setupMedias(session);
-        const sessionId = getSipSessionId(session).substr(0, 20);
+        const sessionId = getSipCallId(session).substr(0, 20);
         updateRemoteStream(session);
         logger.info('progress received', {
             sessionId,
@@ -1486,7 +1488,7 @@ export default class WebRTCClient extends Emitter {
             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) {
             logger.info('html element already exists for session', {
@@ -1531,7 +1533,7 @@ export default class WebRTCClient extends Emitter {
         audioElement.play().catch(() => { });
     }
     _cleanupMedia(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const localStream = this.getLocalStream(sessionId);
         if (localStream) {
             cleanupStream(localStream);
@@ -1550,7 +1552,7 @@ 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));
@@ -1570,7 +1572,7 @@ export default class WebRTCClient extends Emitter {
         if (!pc) {
             return;
         }
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         getStats(pc, (result) => {
             const { results, internal, nomore, ...stats } = result;
             this.statsIntervals[sessionId] = nomore;
@@ -1581,7 +1583,7 @@ export default class WebRTCClient extends Emitter {
         }, SEND_STATS_DELAY);
     }
     _stopSendingStats(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         logger.trace('Check for stopping stats', {
             sessionId,
             ids: Object.keys(this.statsIntervals),

package/lib/__tests__/__voice__/call.test.js

@@ -126,4 +126,16 @@ describe('Call', () => {
             expect(softphone_1.default.client.hangup).toHaveBeenCalled();
         }));
     });
+    describe('state propagation', () => {
+        it('Should update the call state in events', (done) => {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            // @ts-ignore: private method
+            call._sendAction(call_state_machine_1.Actions.MAKE_CALL);
+            call.on('remotelyAccepted', () => {
+                expect(call.state).toStrictEqual({ [call_state_machine_1.States.ESTABLISHED]: call_state_machine_1.EstablishedStates.ONGOING });
+                done();
+            });
+            call.onAccepted();
+        });
+    });
 });

package/lib/checker/checks/webrtc.js

@@ -43,7 +43,7 @@ exports.default = {
         });
         client.on(client.REGISTERED, () => {
             const sipSession = client.call('*10');
-            if (!sipSession || !(0, sdp_1.getSipSessionId)(sipSession)) {
+            if (!sipSession || !(0, sdp_1.getSipCallId)(sipSession)) {
                 return handleError('Unable to make call through WebRTC');
             }
             setTimeout(() => {

package/lib/domain/Phone/WebRTCPhone.js

@@ -283,7 +283,7 @@ class WebRTCPhone extends Emitter_1.default {
                 content: {
                     type: exports.ON_MESSAGE_TRACK_UPDATED,
                     update: isUpgrade ? 'upgrade' : 'downgrade',
-                    sipCallId: sipSession && (0, sdp_1.getSipSessionId)(sipSession),
+                    sipCallId: sipSession && (0, sdp_1.getSipCallId)(sipSession),
                     callId: callSession ? callSession.callId : null,
                     number: callSession ? callSession.number : null,
                 },
@@ -291,7 +291,7 @@ class WebRTCPhone extends Emitter_1.default {
         }, 2500);
     }
     _bindEvents(sipSession) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         if (sipSession instanceof api_1.Invitation) {
             // Monkey patch to know when canceled with the CANCEL message
             const onCancel = sipSession._onCancel.bind(sipSession);
@@ -345,7 +345,7 @@ class WebRTCPhone extends Emitter_1.default {
             const [stream] = event.streams;
             const kind = event && event.track && event.track.kind;
             logger.info('on track stream called on the peer connection', {
-                callId: (0, sdp_1.getSipSessionId)(sipSession),
+                callId: (0, sdp_1.getSipCallId)(sipSession),
                 streamId: stream ? stream.id : null,
                 tracks: stream ? stream.getTracks() : null,
                 kind,
@@ -375,9 +375,9 @@ class WebRTCPhone extends Emitter_1.default {
             }
             const screenTrack = screenShareStream.getVideoTracks()[0];
             const sipSession = this.currentSipSession;
-            const pc = sipSession && this.client.getPeerConnection((0, sdp_1.getSipSessionId)(sipSession));
+            const pc = sipSession && this.client.getPeerConnection((0, sdp_1.getSipCallId)(sipSession));
             const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-            const localStream = sipSession && this.client.getLocalStream((0, sdp_1.getSipSessionId)(sipSession));
+            const localStream = sipSession && this.client.getLocalStream((0, sdp_1.getSipCallId)(sipSession));
             const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
             const hadVideo = !!videoTrack;
             // Stop local video tracks
@@ -442,19 +442,19 @@ class WebRTCPhone extends Emitter_1.default {
     }
     _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
         const screenTrack = screenStream.getVideoTracks()[0];
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         const pc = this.client.getPeerConnection(sipSessionId);
         const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
         logger.info('WebRTC phone - on screensharing', {
             hadVideo,
-            id: (0, sdp_1.getSipSessionId)(sipSession),
+            id: (0, sdp_1.getSipCallId)(sipSession),
             screenTrack,
         });
         if (screenTrack) {
             screenTrack.onended = () => {
                 logger.info('WebRTC phone - on screenshare ended', {
                     hadVideo,
-                    id: (0, sdp_1.getSipSessionId)(sipSession),
+                    id: (0, sdp_1.getSipCallId)(sipSession),
                     screenTrack,
                 });
                 this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
@@ -488,7 +488,7 @@ class WebRTCPhone extends Emitter_1.default {
         this.currentSipSession = sipSession;
         this.currentCallSession = callSession;
         this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call accepted');
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
         }
@@ -551,7 +551,7 @@ class WebRTCPhone extends Emitter_1.default {
         return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
     }
     _onCallTerminated(sipSession) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         logger.info('WebRTC phone - on call terminated', {
             sipId: sipSession.id,
             sipSessionId,
@@ -643,7 +643,7 @@ class WebRTCPhone extends Emitter_1.default {
         if (!this.currentSipSession) {
             return false;
         }
-        return this.currentSipSession && (0, sdp_1.getSipSessionId)(this.currentSipSession) === callSession.getId();
+        return this.currentSipSession && (0, sdp_1.getSipCallId)(this.currentSipSession) === callSession.getId();
     }
     hasVideo(callSession) {
         return this.client.hasVideo(callSession.getId());
@@ -677,7 +677,7 @@ class WebRTCPhone extends Emitter_1.default {
             logger.warn('no CallSession to accept.');
             return Promise.resolve(null);
         }
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId && !(sipSessionId in this.callSessions)) {
             logger.warn('Call session already ended or not found, can\'t accept it.', {
                 sipSessionId,
@@ -709,7 +709,7 @@ class WebRTCPhone extends Emitter_1.default {
         const error = 'No CallSession found to accept.';
         logger.warn(error, {
             id: callSession ? callSession.getId() : 'n/a',
-            ids: this.client.getSipSessionIds(),
+            ids: this.client.getSipCallIds(),
         });
         if (callSession) {
             this.eventEmitter.emit(exports.ON_CALL_ERROR, new Error(error), callSession);
@@ -801,7 +801,7 @@ class WebRTCPhone extends Emitter_1.default {
             if (!sipSession) {
                 return new Promise((resolve, reject) => reject(new Error('No session to hold')));
             }
-            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             const hasVideo = this.client.hasLocalVideo(sessionId);
             logger.info('WebRTC hold sip session', {
                 sipId: sipSession.id,
@@ -830,7 +830,7 @@ class WebRTCPhone extends Emitter_1.default {
                 return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
             }
             const isConference = !!callSession && callSession.isConference();
-            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             logger.info('WebRTC unhold sip session', {
                 sessionId,
                 isConference,
@@ -1059,7 +1059,7 @@ class WebRTCPhone extends Emitter_1.default {
                 console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
                 return false;
             }
-            const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
             logger.info('WebRTC hangup', {
                 sipId: sipSession.id,
                 sipSessionId,
@@ -1251,12 +1251,12 @@ class WebRTCPhone extends Emitter_1.default {
             // if `_createCallSession` is called too soon `hasVideo` will return false because the stream doesn't exists yet
             setTimeout(() => {
                 // Update callSession
-                this._createCallSession(sipSession, this.callSessions[(0, sdp_1.getSipSessionId)(sipSession)]);
+                this._createCallSession(sipSession, this.callSessions[(0, sdp_1.getSipCallId)(sipSession)]);
                 this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
             }, 2000);
         });
         this.client.on(this.client.ACCEPTED, (sipSession) => __awaiter(this, void 0, void 0, function* () {
-            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             const hasSession = (sessionId in this.callSessions);
             logger.info('WebRTC call accepted', {
                 sipId: sessionId,
@@ -1283,7 +1283,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session);
             // `_createCallSession` add the CallSession in `this.callSessions`.
             // so we have to remove it or `callCount` will always return 1 even if the call was rejected
-            delete this.callSessions[(0, sdp_1.getSipSessionId)(session)];
+            delete this.callSessions[(0, sdp_1.getSipCallId)(session)];
             callSession.endTime = new Date();
             this.eventEmitter.emit(exports.ON_CALL_REJECTED, callSession);
         });
@@ -1300,7 +1300,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: (0, sdp_1.getSipSessionId)(session),
+                callId: (0, sdp_1.getSipCallId)(session),
             });
             this.eventEmitter.emit(exports.ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1309,7 +1309,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: (0, sdp_1.getSipSessionId)(session),
+                callId: (0, sdp_1.getSipCallId)(session),
             });
             this.eventEmitter.emit(exports.ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1366,12 +1366,12 @@ class WebRTCPhone extends Emitter_1.default {
             this.eventEmitter.emit(exports.ON_MESSAGE, message);
         });
         this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-            const callSession = this.getCallSession((0, sdp_1.getSipSessionId)(session));
+            const callSession = this.getCallSession((0, sdp_1.getSipCallId)(session));
             this.eventEmitter.emit(exports.ON_NETWORK_STATS, callSession, stats, previousStats);
         });
         // Used when upgrading directly in screenshare mode
         this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession, response, desktop) => {
-            const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
             const localStream = this.client.getLocalStream(sipSessionId);
             const callSession = this.callSessions[sipSessionId];
             logger.info('Updrading directly in screensharing mode', {
@@ -1383,10 +1383,10 @@ class WebRTCPhone extends Emitter_1.default {
     }
     // Find a corresponding sipSession from a CallSession
     findSipSession(callSession) {
-        const keys = this.client.getSipSessionIds();
+        const keys = this.client.getSipCallIds();
         const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
         if (keyIndex === -1) {
-            const currentSipSessionId = this.currentSipSession ? (0, sdp_1.getSipSessionId)(this.currentSipSession) : this.client.getSipSessionIds()[0];
+            const currentSipSessionId = this.currentSipSession ? (0, sdp_1.getSipCallId)(this.currentSipSession) : this.client.getSipCallIds()[0];
             return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
         }
         return this.client.getSipSession(keys[keyIndex]);
@@ -1394,12 +1394,12 @@ class WebRTCPhone extends Emitter_1.default {
     getCallSession(sipSessionId) {
         return this.callSessions[sipSessionId];
     }
-    getSipSessionIdFromCallSession(callSession) {
+    getSipCallIdFromCallSession(callSession) {
         const sipSession = this.findSipSession(callSession);
-        return sipSession ? (0, sdp_1.getSipSessionId)(sipSession) : null;
+        return sipSession ? (0, sdp_1.getSipCallId)(sipSession) : null;
     }
     _updateCallSession(callSession) {
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId) {
             this.callSessions[sipSessionId] = callSession;
         }
@@ -1472,7 +1472,7 @@ class WebRTCPhone extends Emitter_1.default {
         const number = (0, sdp_1.getCallNumber)(sipSession);
         const displayName = (0, sdp_1.getCallDisplayName)(sipSession);
         const { state } = sipSession || {};
-        const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sessionId = (0, sdp_1.getSipCallId)(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
         const callSession = new CallSession_1.default(Object.assign({ callId: fromSession && fromSession.callId, sipCallId: sessionId, sipStatus: state, displayName, 
             // @ts-ignore: date / number

package/lib/simple/Phone.js

@@ -237,11 +237,11 @@ class Phone extends Emitter_1.default {
     stopNetworkMonitoring(callSession) {
         return this.phone ? this.phone.stopNetworkMonitoring(callSession) : null;
     }
-    getSipSessionId(sipSession) {
+    getSipCallId(sipSession) {
         if (!sipSession || !this.phone) {
             return null;
         }
-        return (0, sdp_1.getSipSessionId)(sipSession);
+        return (0, sdp_1.getSipCallId)(sipSession);
     }
     sendMessage(body, sipSession, contentType = 'text/plain') {
         const toSipSession = sipSession || this.getCurrentSipSession();

package/lib/simple/room/Room.js

@@ -372,7 +372,7 @@ class Room extends Emitter_1.default {
             logger.info('on room audio stream');
             this.audioStream = stream;
             if (!this.roomAudioElement) {
-                const sessionId = (0, sdp_1.getSipSessionId)((_b = index_1.default.Phone.phone) === null || _b === void 0 ? void 0 : _b.currentSipSession);
+                const sessionId = (0, sdp_1.getSipCallId)((_b = index_1.default.Phone.phone) === null || _b === void 0 ? void 0 : _b.currentSipSession);
                 this.roomAudioElement = (yield ((_c = index_1.default.Phone.phone) === null || _c === void 0 ? void 0 : _c.createAudioElementFor(sessionId)));
                 this.roomAudioElement.srcObject = stream;
             }

package/lib/simple/room/SipRoom.js

@@ -136,7 +136,7 @@ class SipRoom extends Room_1.default {
     }
     _getCurrentSipCallIs() {
         var _a;
-        return (0, sdp_1.getSipSessionId)((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
+        return (0, sdp_1.getSipCallId)((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
     }
 }
 exports.default = SipRoom;

package/lib/state-machine/utils.js

@@ -44,7 +44,7 @@ const assertCan = (actor, action) => {
 exports.assertCan = assertCan;
 const assertState = (actor, state) => {
     if (!(0, exports.hasState)(actor, state)) {
-        const message = `Invalid state ${state}`;
+        const message = `Invalid state ${state}, has state ${(0, exports.getState)(actor)}`;
         logger.warn(message);
         throw new InvalidState_1.default(message, state);
     }

package/lib/types/domain/Phone/WebRTCPhone.d.ts

@@ -164,7 +164,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     bindClientEvents(): void;
     findSipSession(callSession?: CallSession): WazoSession | null | undefined;
     getCallSession(sipSessionId: string): CallSession | null | undefined;
-    getSipSessionIdFromCallSession(callSession: CallSession): string | null;
+    getSipCallIdFromCallSession(callSession: CallSession): string | null;
     _updateCallSession(callSession: CallSession): void;
     _onMessage(message: Message & {
         method?: string;