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

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

@@ -45,7 +45,7 @@ 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.

package/src/utils/webrtc.ts

@@ -7,7 +7,7 @@ import { stripVideo } from 'sip.js/lib/platform/web';
 
 import WazoSessionDescriptionHandler, { wazoMediaStreamFactory } from '../lib/WazoSessionDescriptionHandler';
 import { SipCall, PeerConnection, UserAgentOptions, WebRtcConfig, UserAgentConfigOverrides } from '../domain/types';
-import { getVideoDirection, hasAnActiveVideo, makeURI, replaceLocalIpModifier } from './sdp';
+import { getSipCallId, getVideoDirection, hasAnActiveVideo, makeURI, replaceLocalIpModifier } from './sdp';
 import IssueReporter from '../service/IssueReporter';
 import { lastIndexOf } from './array';
 
@@ -274,7 +274,7 @@ export const cleanupStream = (stream: MediaStream): void => {
 
 export const setLocalMediaStream = (sipCall: SipCall, newStream: MediaStream): void => {
   logger.info('setting local media stream', {
-    id: sipCall.id,
+    id: getSipCallId(sipCall),
     tracks: newStream ? newStream.getTracks() : null,
   });
 
@@ -290,7 +290,7 @@ export const updateRemoteStream = (sipCall: SipCall, allTracks = true): void =>
   const pc = getPeerConnection(sipCall);
 
   logger.info('Updating remote stream', {
-    id: sipCall.id,
+    id: getSipCallId(sipCall),
     tracks: remoteStream ? remoteStream.getTracks() : null,
     receiverTracks: pc && pc.getReceivers ? pc.getReceivers().map((receiver) => receiver.track) : null,
   });

package/src/voice/call.ts

@@ -15,7 +15,7 @@ import newFrom from '../utils/new-from';
 import updateFrom from '../utils/update-from';
 import IssueReporter from '../service/IssueReporter';
 import softphone, { EVENT_TERMINATE_SOUND, Softphone } 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';
 
@@ -356,7 +356,7 @@ class Call extends EventEmitter {
   }
 
   isEstablished() {
-    return States.ESTABLISHED in (this.state as Record<StateTypes, EstablishedStateTypes>);
+    return typeof this.state === 'object' && States.ESTABLISHED in (this.state as Record<StateTypes, EstablishedStateTypes>);
   }
 
   isRinging() {
@@ -632,11 +632,11 @@ class Call extends EventEmitter {
   }
 
   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(): StateTypes | Record<StateTypes, EstablishedStateTypes> {

package/src/voice/softphone.ts

@@ -16,6 +16,7 @@ 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';
@@ -194,7 +195,7 @@ export class Softphone extends EventEmitter {
       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);
     }
 
@@ -207,6 +208,7 @@ export class Softphone extends EventEmitter {
       sipCall = this.client.call(options.params.To, options.withCamera, options.audioOnly, options.conference, options) as SipCall;
       const call = new Call(sipCall, this);
 
+      this.calls.push(call);
       this.emit(EVENT_OUTGOING, call);
 
       return call;
@@ -401,7 +403,7 @@ export class Softphone extends EventEmitter {
 
     this.client.on(this.client.ON_REINVITE, (sipCall: SipCall, request: IncomingRequestMessage, updatedCalleeName: string, updatedNumber: string, hadRemoteVideo: boolean) => {
       logger.info('softphone - on reinvite', {
-        callId: sipCall.id,
+        callId: getSipCallId(sipCall),
         inviteId: request.callId,
         updatedCalleeName,
         updatedNumber,
@@ -530,15 +532,19 @@ export class Softphone extends EventEmitter {
   }
 
   private _getCall(sipCall: SipCall): Call | undefined {
-    return this.calls.find(call => call.id === sipCall.id);
+    return this.calls.find(call => call.id === getSipCallId(sipCall));
+  }
+
+  private _getCallIds(): string[] {
+    return this.calls.map(call => call.id);
   }
 
   private _onCallEvent(sipCall: SipCall, eventName: string, ...args: any[]) {
-    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;
     }