@wazo/sdk 1.0.0-alpha.5 → 1.0.0-alpha.7

package/lib/web-rtc-client.js

@@ -85,7 +85,9 @@ class WebRTCClient extends Emitter_1.default {
         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;
@@ -218,7 +220,7 @@ class WebRTCClient extends Emitter_1.default {
                     // @ts-ignore: private
                     this.registerer.waitingToggle(false);
                 }
-                this.eventEmitter.emit(UNREGISTERED);
+                this.eventEmitter.emit(UNREGISTERED, error);
             }
         });
     }
@@ -425,7 +427,7 @@ class WebRTCClient extends Emitter_1.default {
             this._setupSession(session);
         }
         if (conference) {
-            this.conferences[(0, sdp_1.getSipSessionId)(session)] = true;
+            this.conferences[(0, sdp_1.getSipCallId)(session)] = true;
         }
         const inviteOptions = {
             requestDelegate: {
@@ -506,7 +508,7 @@ class WebRTCClient extends Emitter_1.default {
             try {
                 this._stopSendingStats(session);
                 this._cleanupMedia(session);
-                delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
+                delete this.sipSessions[(0, sdp_1.getSipCallId)(session)];
                 // Check if Invitation or Inviter (Invitation = incoming call)
                 const isInviter = session instanceof api_1.Inviter;
                 // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
@@ -529,7 +531,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     // Fetch and emit an event at `interval` with session network stats
     startNetworkMonitoring(session, interval = 1000) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         logger.info('starting network inspection', {
             id: sessionId,
             clientId: this.clientId,
@@ -538,7 +540,7 @@ class WebRTCClient extends Emitter_1.default {
         this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
     }
     stopNetworkMonitoring(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const exists = (sessionId in this.networkMonitoringInterval);
         logger.info('stopping network inspection', {
             clientId: this.clientId,
@@ -666,7 +668,7 @@ class WebRTCClient extends Emitter_1.default {
         (0, webrtc_1.toggleVideo)(session, true);
     }
     hold(session, isConference = false, hadVideo = false) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
         logger.info('sdk webrtc hold', {
             sessionId,
@@ -714,7 +716,7 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     unhold(session, isConference = false) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
         logger.info('sdk webrtc unhold', {
             sessionId,
@@ -729,7 +731,7 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.resolve();
         }
         this.unmute(session);
-        delete this.heldSessions[(0, sdp_1.getSipSessionId)(session)];
+        delete this.heldSessions[(0, sdp_1.getSipCallId)(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             hold: false,
             conference: isConference,
@@ -754,7 +756,7 @@ class WebRTCClient extends Emitter_1.default {
         return this.heldSessions[sessionId];
     }
     isCallHeld(session) {
-        return (0, sdp_1.getSipSessionId)(session) in this.heldSessions;
+        return (0, sdp_1.getSipCallId)(session) in this.heldSessions;
     }
     isVideoRemotelyHeld(sessionId) {
         return (0, webrtc_1.isVideoRemotelyHeld)(this.heldSessions[sessionId]);
@@ -764,7 +766,7 @@ class WebRTCClient extends Emitter_1.default {
             return false;
         }
         logger.info('Sending DTMF', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
             tone,
         });
         return session.sessionDescriptionHandler.sendDtmf(tone);
@@ -781,7 +783,7 @@ class WebRTCClient extends Emitter_1.default {
     transfer(session, target) {
         this.hold(session);
         logger.info('Transfering a session', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
             target,
         });
         const options = {
@@ -804,21 +806,21 @@ class WebRTCClient extends Emitter_1.default {
     atxfer(session) {
         this.hold(session);
         logger.info('webrtc transfer started', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
         });
         const result = {
             newSession: null,
             init: (target) => __awaiter(this, void 0, void 0, function* () {
                 logger.info('webrtc transfer initialized', {
-                    id: (0, sdp_1.getSipSessionId)(session),
+                    id: (0, sdp_1.getSipCallId)(session),
                     target,
                 });
                 result.newSession = yield this.call(target);
             }),
             complete: () => {
                 logger.info('webrtc transfer completed', {
-                    id: (0, sdp_1.getSipSessionId)(session),
-                    referId: (0, sdp_1.getSipSessionId)(result.newSession),
+                    id: (0, sdp_1.getSipCallId)(session),
+                    referId: (0, sdp_1.getSipCallId)(result.newSession),
                 });
                 session.refer(result.newSession);
             },
@@ -990,7 +992,7 @@ class WebRTCClient extends Emitter_1.default {
         }
         const sdh = session.sessionDescriptionHandler;
         const pc = sdh.peerConnection;
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const localStream = this.getLocalStream(sessionId);
         logger.info('changing video input device', {
             id,
@@ -1053,7 +1055,7 @@ class WebRTCClient extends Emitter_1.default {
         const desktop = newConstraints && newConstraints.desktop;
         logger.info('Sending reinvite', {
             clientId: this.clientId,
-            id: (0, sdp_1.getSipSessionId)(sipSession),
+            id: (0, sdp_1.getSipCallId)(sipSession),
             newConstraints,
             conference,
             audioOnly,
@@ -1085,7 +1087,7 @@ class WebRTCClient extends Emitter_1.default {
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
-                        id: (0, sdp_1.getSipSessionId)(sipSession),
+                        id: (0, sdp_1.getSipCallId)(sipSession),
                         wasMuted,
                         shouldDoScreenSharing,
                     });
@@ -1200,7 +1202,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     onCallEnded(session) {
         this._cleanupMedia(session);
-        delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
+        delete this.sipSessions[(0, sdp_1.getSipCallId)(session)];
         this._stopSendingStats(session);
         this.stopNetworkMonitoring(session);
     }
@@ -1213,14 +1215,14 @@ class WebRTCClient extends Emitter_1.default {
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
-        const id = (0, sdp_1.getSipSessionId)(session);
+        const id = (0, sdp_1.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, sdp_1.SIP_ID_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id.substring(0, sdp_1.SIP_ID_MIN_LENGTH)] : null;
     }
-    getSipSessionIds() {
+    getSipCallIds() {
         return Object.keys(this.sipSessions);
     }
     updateLocalStream(sipSessionId, newStream) {
@@ -1377,7 +1379,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     // Invitation and Inviter extends Session
     _setupSession(session) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(session);
+        const sipSessionId = (0, sdp_1.getSipCallId)(session);
         // When receiving an Invitation, the delegate is not defined.
         if (!session.delegate) {
             session.delegate = {};
@@ -1441,7 +1443,7 @@ class WebRTCClient extends Emitter_1.default {
     _onProgress(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, early = false) {
             yield this._setupMedias(session);
-            const sessionId = (0, sdp_1.getSipSessionId)(session).substr(0, 20);
+            const sessionId = (0, sdp_1.getSipCallId)(session).substr(0, 20);
             (0, webrtc_1.updateRemoteStream)(session);
             logger.info('progress received', {
                 sessionId,
@@ -1499,7 +1501,7 @@ class WebRTCClient extends Emitter_1.default {
                 return;
             }
             // Safari hack, because you cannot call .play() from a non user action
-            const sessionId = (0, sdp_1.getSipSessionId)(session);
+            const sessionId = (0, sdp_1.getSipCallId)(session);
             const isConference = this.isConference(sessionId);
             if (sessionId in this.audioElements) {
                 logger.info('html element already exists for session', {
@@ -1545,7 +1547,7 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     _cleanupMedia(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const localStream = this.getLocalStream(sessionId);
         if (localStream) {
             (0, webrtc_1.cleanupStream)(localStream);
@@ -1564,7 +1566,7 @@ class WebRTCClient extends Emitter_1.default {
         };
         if (this._hasAudio() && (0, webrtc_1.isWeb)()) {
             if (session) {
-                cleanLocalElement((0, sdp_1.getSipSessionId)(session));
+                cleanLocalElement((0, sdp_1.getSipCallId)(session));
             }
             else {
                 Object.keys(this.audioElements).forEach(id => cleanLocalElement(id));
@@ -1584,7 +1586,7 @@ class WebRTCClient extends Emitter_1.default {
         if (!pc) {
             return;
         }
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         (0, getstats_1.default)(pc, (result) => {
             const { results, internal, nomore } = result, stats = __rest(result, ["results", "internal", "nomore"]);
             this.statsIntervals[sessionId] = nomore;
@@ -1592,7 +1594,7 @@ class WebRTCClient extends Emitter_1.default {
         }, SEND_STATS_DELAY);
     }
     _stopSendingStats(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         logger.trace('Check for stopping stats', {
             sessionId,
             ids: Object.keys(this.statsIntervals),

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "1.0.0-alpha.5",
+  "version": "1.0.0-alpha.7",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "index.js",
   "types": "lib/types/index.d.ts",

package/src/__tests__/__voice__/call.test.ts

@@ -133,4 +133,19 @@ describe('Call', () => {
       expect(softphone.client.hangup).toHaveBeenCalled();
     });
   });
+
+  describe('state propagation', () => {
+    it('Should update the call state in events', (done) => {
+      const call = new Call(sipCall, softphone);
+      // @ts-ignore: private method
+      call._sendAction(Actions.MAKE_CALL);
+
+      call.on('remotelyAccepted', () => {
+        expect(call.state).toStrictEqual({ [States.ESTABLISHED]: EstablishedStates.ONGOING });
+        done();
+      });
+
+      call.onAccepted();
+    });
+  });
 });

package/src/checker/checks/webrtc.ts

@@ -1,5 +1,5 @@
 /* global MediaStream */
-import { getSipSessionId } from '../../utils/sdp';
+import { getSipCallId } from '../../utils/sdp';
 import WebRTCClient from '../../web-rtc-client';
 
 export default {
@@ -36,7 +36,7 @@ export default {
     client.on(client.REGISTERED, () => {
       const sipSession = client.call('*10');
 
-      if (!sipSession || !getSipSessionId(sipSession)) {
+      if (!sipSession || !getSipCallId(sipSession)) {
         return handleError('Unable to make call through WebRTC');
       }
 

package/src/domain/Phone/WebRTCPhone.ts

@@ -12,7 +12,7 @@ import WazoWebRTCClient from '../../web-rtc-client';
 import Emitter from '../../utils/Emitter';
 import IssueReporter from '../../service/IssueReporter';
 import { PeerConnection, WazoSession } from '../types';
-import { getCallDisplayName, getCallNumber, getSipSessionId } from '../../utils/sdp';
+import { getCallDisplayName, getCallNumber, getSipCallId } from '../../utils/sdp';
 import { downgradeToAudio, getStreamFromConstraints, getWebRtcStats, setLocalMediaStream, upgradeToVideo } from '../../utils/webrtc';
 
 export const ON_USER_AGENT = 'onUserAgent';
@@ -360,7 +360,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         content: {
           type: ON_MESSAGE_TRACK_UPDATED,
           update: isUpgrade ? 'upgrade' : 'downgrade',
-          sipCallId: sipSession && getSipSessionId(sipSession),
+          sipCallId: sipSession && getSipCallId(sipSession),
           callId: callSession ? callSession.callId : null,
           number: callSession ? callSession.number : null,
         },
@@ -369,7 +369,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   }
 
   _bindEvents(sipSession: WazoSession) {
-    const sipSessionId = getSipSessionId(sipSession);
+    const sipSessionId = getSipCallId(sipSession);
 
     if (sipSession instanceof Invitation) {
       // Monkey patch to know when canceled with the CANCEL message
@@ -437,7 +437,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       const [stream] = event.streams;
       const kind = event && event.track && event.track.kind;
       logger.info('on track stream called on the peer connection', {
-        callId: getSipSessionId(sipSession),
+        callId: getSipCallId(sipSession),
         streamId: stream ? stream.id : null,
         tracks: stream ? stream.getTracks() : null,
         kind,
@@ -472,9 +472,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     const screenTrack = screenShareStream.getVideoTracks()[0];
     const sipSession = this.currentSipSession;
-    const pc = sipSession && this.client.getPeerConnection(getSipSessionId(sipSession));
+    const pc = sipSession && this.client.getPeerConnection(getSipCallId(sipSession));
     const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-    const localStream = sipSession && this.client.getLocalStream(getSipSessionId(sipSession));
+    const localStream = sipSession && this.client.getLocalStream(getSipCallId(sipSession));
     const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
     const hadVideo = !!videoTrack;
 
@@ -549,12 +549,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
   _onScreenSharing(screenStream: MediaStream, sipSession: WazoSession, callSession: CallSession | null | undefined, hadVideo: boolean, desktop?: boolean | null | undefined) {
     const screenTrack = screenStream.getVideoTracks()[0];
-    const sipSessionId = getSipSessionId(sipSession);
+    const sipSessionId = 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: getSipSessionId(sipSession),
+      id: getSipCallId(sipSession),
       screenTrack,
     });
 
@@ -562,7 +562,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       screenTrack.onended = () => {
         logger.info('WebRTC phone - on screenshare ended', {
           hadVideo,
-          id: getSipSessionId(sipSession),
+          id: getSipCallId(sipSession),
           screenTrack,
         });
         this.eventEmitter.emit(ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
@@ -602,7 +602,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     this.currentSipSession = sipSession;
     this.currentCallSession = callSession;
     this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call accepted');
-    const sipSessionId = getSipSessionId(sipSession);
+    const sipSessionId = getSipCallId(sipSession);
 
     if (sipSessionId) {
       this.removeIncomingSessions(sipSessionId);
@@ -674,7 +674,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   }
 
   _onCallTerminated(sipSession: WazoSession): boolean {
-    const sipSessionId = getSipSessionId(sipSession);
+    const sipSessionId = getSipCallId(sipSession);
 
     logger.info('WebRTC phone - on call terminated', {
       sipId: sipSession.id,
@@ -788,7 +788,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return false;
     }
 
-    return this.currentSipSession && getSipSessionId(this.currentSipSession) === callSession.getId();
+    return this.currentSipSession && getSipCallId(this.currentSipSession) === callSession.getId();
   }
 
   hasVideo(callSession: CallSession): boolean {
@@ -831,7 +831,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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.', {
@@ -872,7 +872,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     logger.warn(error, {
       id: callSession ? callSession.getId() : 'n/a',
-      ids: this.client.getSipSessionIds(),
+      ids: this.client.getSipCallIds(),
     });
 
     if (callSession) {
@@ -980,7 +980,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return new Promise((resolve, reject) => reject(new Error('No session to hold')));
     }
 
-    const sessionId = getSipSessionId(sipSession);
+    const sessionId = getSipCallId(sipSession);
     const hasVideo = this.client.hasLocalVideo(sessionId);
     logger.info('WebRTC hold sip session', {
       sipId: sipSession.id,
@@ -1014,7 +1014,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     const isConference = !!callSession && callSession.isConference();
-    const sessionId = getSipSessionId(sipSession);
+    const sessionId = getSipCallId(sipSession);
     logger.info('WebRTC unhold sip session', {
       sessionId,
       isConference,
@@ -1296,7 +1296,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
       return false;
     }
-    const sipSessionId = getSipSessionId(sipSession);
+    const sipSessionId = getSipCallId(sipSession);
 
     logger.info('WebRTC hangup', {
       sipId: sipSession.id,
@@ -1527,13 +1527,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
       // 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[getSipSessionId(sipSession)]);
+        this._createCallSession(sipSession, this.callSessions[getSipCallId(sipSession)]);
 
         this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
       }, 2000);
     });
     this.client.on(this.client.ACCEPTED, async (sipSession: WazoSession) => {
-      const sessionId = getSipSessionId(sipSession);
+      const sessionId = getSipCallId(sipSession);
       const hasSession = (sessionId in this.callSessions);
       logger.info('WebRTC call accepted', {
         sipId: sessionId,
@@ -1565,7 +1565,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
       // `_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[getSipSessionId(session)];
+      delete this.callSessions[getSipCallId(session)];
 
       callSession.endTime = new Date();
 
@@ -1585,7 +1585,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       const callSession = this._createCallSession(session, null, {
         incoming: false,
         ringing: true,
-        callId: getSipSessionId(session),
+        callId: getSipCallId(session),
       });
 
       this.eventEmitter.emit(ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
@@ -1596,7 +1596,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       const callSession = this._createCallSession(session, null, {
         incoming: false,
         ringing: true,
-        callId: getSipSessionId(session),
+        callId: getSipCallId(session),
       });
 
       this.eventEmitter.emit(ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
@@ -1657,12 +1657,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.eventEmitter.emit(ON_MESSAGE, message);
     });
     this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-      const callSession = this.getCallSession(getSipSessionId(session));
+      const callSession = this.getCallSession(getSipCallId(session));
       this.eventEmitter.emit(ON_NETWORK_STATS, callSession, stats, previousStats);
     });
     // Used when upgrading directly in screenshare mode
     this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession: WazoSession, response: any, desktop: boolean) => {
-      const sipSessionId = getSipSessionId(sipSession);
+      const sipSessionId = getSipCallId(sipSession);
       const localStream = this.client.getLocalStream(sipSessionId);
       const callSession = this.callSessions[sipSessionId];
       logger.info('Updrading directly in screensharing mode', {
@@ -1676,11 +1676,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
   // Find a corresponding sipSession from a CallSession
   findSipSession(callSession?: CallSession): WazoSession | null | undefined {
-    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 ? getSipSessionId(this.currentSipSession) : this.client.getSipSessionIds()[0];
+      const currentSipSessionId = this.currentSipSession ? getSipCallId(this.currentSipSession) : this.client.getSipCallIds()[0];
       return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
     }
 
@@ -1691,13 +1691,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.callSessions[sipSessionId];
   }
 
-  getSipSessionIdFromCallSession(callSession: CallSession): string | null {
+  getSipCallIdFromCallSession(callSession: CallSession): string | null {
     const sipSession = this.findSipSession(callSession);
-    return sipSession ? getSipSessionId(sipSession) : null;
+    return sipSession ? getSipCallId(sipSession) : null;
   }
 
   _updateCallSession(callSession: CallSession): void {
-    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+    const sipSessionId = this.getSipCallIdFromCallSession(callSession);
 
     if (sipSessionId) {
       this.callSessions[sipSessionId] = callSession;
@@ -1788,7 +1788,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const number = getCallNumber(sipSession);
     const displayName = getCallDisplayName(sipSession);
     const { state } = sipSession || {};
-    const sessionId = getSipSessionId(sipSession);
+    const sessionId = getSipCallId(sipSession);
 
     fromSession = fromSession || this.callSessions[sessionId];
     const callSession = new CallSession({

package/src/simple/Phone.ts

@@ -12,7 +12,7 @@ import Emitter from '../utils/Emitter';
 import Wazo from './index';
 import SFUNotAvailableError from '../domain/SFUNotAvailableError';
 import { WazoSession, WebRtcConfig } from '../domain/types';
-import { getSipSessionId } from '../utils/sdp';
+import { getSipCallId } from '../utils/sdp';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -330,11 +330,11 @@ export class Phone extends Emitter {
     return this.phone ? this.phone.stopNetworkMonitoring(callSession) : null;
   }
 
-  getSipSessionId(sipSession: WazoSession): string | null | undefined {
+  getSipCallId(sipSession: WazoSession): string | null | undefined {
     if (!sipSession || !this.phone) {
       return null;
     }
-    return getSipSessionId(sipSession);
+    return getSipCallId(sipSession);
   }
 
   sendMessage(body: string, sipSession?: WazoSession, contentType = 'text/plain'): void {

package/src/simple/room/Room.ts

@@ -10,7 +10,7 @@ import RemoteParticipant from './RemoteParticipant';
 import IssueReporter from '../../service/IssueReporter';
 import LocalParticipant from './LocalParticipant';
 import { PeerConnection, WazoSession } from '../../domain/types';
-import { getSipSessionId } from '../../utils/sdp';
+import { getSipCallId } from '../../utils/sdp';
 
 export const SIGNAL_TYPE_PARTICIPANT_UPDATE = 'signal/PARTICIPANT_UPDATE';
 export const SIGNAL_TYPE_PARTICIPANT_REQUEST = 'signal/PARTICIPANT_REQUEST';
@@ -498,7 +498,7 @@ class Room extends Emitter {
       this.audioStream = stream;
 
       if (!this.roomAudioElement) {
-        const sessionId = getSipSessionId(Wazo.Phone.phone?.currentSipSession as WazoSession);
+        const sessionId = getSipCallId(Wazo.Phone.phone?.currentSipSession as WazoSession);
         this.roomAudioElement = await Wazo.Phone.phone?.createAudioElementFor(sessionId as string) as HTMLAudioElement;
         this.roomAudioElement.srcObject = stream;
       } else {

package/src/simple/room/SipRoom.ts

@@ -4,7 +4,7 @@ import IssueReporter from '../../service/IssueReporter';
 import Wazo from '../index';
 import Room from './Room';
 import { WazoSession } from '../../domain/types';
-import { getSipSessionId } from '../../utils/sdp';
+import { getSipCallId } from '../../utils/sdp';
 
 const logger = IssueReporter.loggerFor('sdk-sip-room');
 
@@ -157,7 +157,7 @@ class SipRoom extends Room {
   }
 
   _getCurrentSipCallIs() {
-    return getSipSessionId(Wazo.Phone.phone?.currentSipSession as WazoSession);
+    return getSipCallId(Wazo.Phone.phone?.currentSipSession as WazoSession);
   }
 
 }

package/src/state-machine/utils.ts

@@ -16,8 +16,15 @@ export const can = (actor: SoftphoneActorRef | CallActorRef, action: SoftphoneAc
   return actor.getSnapshot().can({ type: action });
 };
 
-export const hasState = (actor: SoftphoneActorRef | CallActorRef, state: SoftphoneStates | CallStates): boolean => {
-  return getState(actor) === state;
+export const hasState = (actor: SoftphoneActorRef | CallActorRef, state: SoftphoneStates | CallStates, innerState?: EstablishedStateTypes): boolean => {
+  const currentState = getState(actor);
+
+  // When using nested states, we'll have an object here
+  if (typeof currentState === 'object') {
+    return innerState ? state in currentState && currentState[state as CallStates] === innerState : state in currentState;
+  }
+
+  return currentState === state;
 };
 
 export const waitUntilState = (actor: SoftphoneActorRef | CallActorRef, state: SoftphoneStates | CallStates, timeout = 5000): Promise<unknown> => {
@@ -45,7 +52,7 @@ export const assertCan = (actor: SoftphoneActorRef | CallActorRef, action: Softp
 
 export const assertState = (actor: SoftphoneActorRef | CallActorRef, state: SoftphoneStates | CallStates): void => {
   if (!hasState(actor, state)) {
-    const message = `Invalid state ${state}`;
+    const message = `Invalid state ${state}, has state ${getState(actor)}`;
     logger.warn(message);
 
     throw new InvalidState(message, state);

package/src/utils/sdp.ts

@@ -6,7 +6,7 @@ import { URI } from 'sip.js/lib/grammar/uri';
 
 import type { SipCall } from '../domain/types';
 
-export const SIP_ID_LENGTH = 36;
+export const SIP_ID_MIN_LENGTH = 20;
 
 export type Candidate = {
   foundation: string;
@@ -147,25 +147,25 @@ export const addIcesInAllBundles = (sdp: string) => {
   return sdpParser.write(parsedSdp);
 };
 
-export const getSipSessionId = (sipSession: SipCall | null | undefined): string => {
+export const getSipCallId = (sipSession: SipCall | null | undefined): string => {
   if (!sipSession) {
     return '';
   }
 
   const message = sipSession.message as Partial<{ callId?: string }>;
   if (message?.callId) {
-    return message.callId.substring(0, SIP_ID_LENGTH);
+    return message.callId.substring(0, SIP_ID_MIN_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.substring(0, SIP_ID_MIN_LENGTH);
   }
 
   // For Invitation
-  return (sipSession.id || '').substring(0, SIP_ID_LENGTH);
+  return (sipSession.id || '').substring(0, SIP_ID_MIN_LENGTH);
 };
 
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.