@wazo/sdk 0.37.20 → 0.37.21

package/src/api/confd.js

@@ -69,8 +69,11 @@ export default (client: ApiRequester, baseUrl: string) => ({
     }
   },
 
-  createMyMeeting: (name: string): Promise<Meeting> =>
-    client.post(`${baseUrl}/users/me/meetings`, { name }).then(Meeting.parse),
+  getMyMeetings: (): Promise<Meeting> =>
+    client.get(`${baseUrl}/users/me/meetings`).then(response => Meeting.parseMany(response.items)),
+
+  createMyMeeting: (name: string, persistent: boolean): Promise<Meeting> =>
+    client.post(`${baseUrl}/users/me/meetings`, { name, persistent }).then(Meeting.parse),
 
   deleteMyMeeting: (meetingUuid: string): Promise<Meeting> =>
     client.delete(`${baseUrl}/users/me/meetings/${meetingUuid}`, null),

package/src/domain/Meeting.js

@@ -37,11 +37,24 @@ export default class Meeting {
     });
   }
 
+  static parseMany(items: MeetingCreationResponse[]): Array<Meeting> {
+    return items.map(plain => Meeting.parse(plain));
+  }
+
   static newFrom(meeting: Meeting) {
     return newFrom(meeting, Meeting);
   }
 
-  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, uri, number, persistent }: Object = {}) {
+  constructor({
+    uuid,
+    name,
+    guestSipAuthorization,
+    ownerUuids,
+    port,
+    uri,
+    number,
+    persistent,
+  }: Object = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
     this.name = name;

package/src/domain/Phone/WebRTCPhone.js

@@ -414,9 +414,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
         case SessionState.Terminated: {
           logger.info('WebRTC phone - call terminated', { sipId: sipSession.id });
 
+          // Should be called before `_onCallTerminated` or the callCount will not decrement...
+          const callSession = this._createCallSession(sipSession);
           const wasCurrentSession = this._onCallTerminated(sipSession);
 
-          return this.eventEmitter.emit(ON_CALL_ENDED, this._createCallSession(sipSession), wasCurrentSession);
+          return this.eventEmitter.emit(ON_CALL_ENDED, callSession, wasCurrentSession);
         }
         default:
           break;
@@ -1366,6 +1368,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return callSession ? this.client.getRemoteVideoStream(callSession.sipCallId) : null;
   }
 
+  getRemoteVideoStreamFromPc(callSession: CallSession): ?MediaStream {
+    return callSession ? this.client.getRemoteVideoStreamFromPc(callSession.sipCallId) : null;
+  }
+
   hasRemoteVideo(callSession: CallSession): boolean {
     return callSession ? this.client.hasRemoteVideo(callSession.sipCallId) : false;
   }

package/src/service/IssueReporter.js

@@ -64,6 +64,7 @@ class IssueReporter {
   bufferTimeout: ?TimeoutID;
   _boundProcessBuffer: Function;
   _boundParseLoggerBody: Function;
+  _callback: ?Function;
 
   constructor() {
     addLevelsTo(this);
@@ -75,12 +76,17 @@ class IssueReporter {
     this.bufferTimeout = null;
     this._boundProcessBuffer = this._processBuffer.bind(this);
     this._boundParseLoggerBody = this._parseLoggerBody.bind(this);
+    this._callback = null;
   }
 
   init() {
     this._catchConsole();
   }
 
+  setCallback(cb: Function) {
+    this._callback = cb;
+  }
+
   configureRemoteClient(configuration: Object = { tag: 'wazo-sdk', host: null, port: null, level: null, extra: {} }) {
     this.remoteClientConfiguration = configuration;
   }
@@ -116,6 +122,7 @@ class IssueReporter {
 
     // Handle category label
     let category = null;
+    let skipSendToRemote = false;
     let extra = {};
     if (args[0].indexOf(CATEGORY_PREFIX) === 0) {
       category = args[0].split('=')[1];
@@ -131,6 +138,7 @@ class IssueReporter {
           errorMessage: lastArg.message,
           errorStack: lastArg.stack,
           errorType: lastArg.constructor.name,
+          skipSendToRemote: lastArg.skipSendToRemote,
         };
       } else {
         extra = lastArg;
@@ -139,6 +147,11 @@ class IssueReporter {
       args.splice(1, 1);
     }
 
+    if (extra.skipSendToRemote) {
+      skipSendToRemote = true;
+      delete extra.skipSendToRemote;
+    }
+
     const date = moment().format('YYYY-MM-DD HH:mm:ss.SSS');
     const message = args.join(', ');
     let consoleMessage = message;
@@ -159,7 +172,13 @@ class IssueReporter {
     const oldMethod = this.oldConsoleMethods[consoleLevel] || this.oldConsoleMethods.log;
     oldMethod.apply(oldMethod, [date, consoleMessage]);
 
-    this._sendToRemoteLogger(level, { date, message, category, ...extra });
+    if (this._callback) {
+      this._callback(level, consoleMessage);
+    }
+
+    if (!skipSendToRemote) {
+      this._sendToRemoteLogger(level, { date, message, category, ...extra });
+    }
   }
 
   logRequest(url: string, options: Object, response: Object, start: Date) {
@@ -304,7 +323,11 @@ class IssueReporter {
       },
       // $FlowFixMe
       body,
-    }).catch(() => {
+    }).catch((e: Error) => {
+      // $FlowFixMe
+      e.skipSendToRemote = true;
+      this.log('error', this._makeCategory('grafana'), 'Sending log to grafana, error', e);
+
       setTimeout(() => {
         if (isArray) {
           payload = payload.map(message => {

package/src/simple/Phone.js

@@ -347,6 +347,11 @@ class Phone extends Emitter {
     return this.getRemoteVideoStream(callSession);
   }
 
+  //  Useful in a react-native environment when remoteMediaStream is not updated
+  getRemoteVideoStreamFromPc(callSession: CallSession) {
+    return this.phone ? this.phone.getRemoteVideoStreamFromPc(callSession) : false;
+  }
+
   hasVideo(callSession: CallSession): boolean {
     return this.phone ? this.phone.hasVideo(callSession) : false;
   }

package/src/types/WebSocketMessage.js

@@ -7,6 +7,7 @@ import type {
 } from '../domain/SwitchboardCall';
 import type { ChatMessageResponse } from '../domain/ChatMessage';
 import ChatRoom from '../domain/ChatRoom';
+import Meeting from '../domain/Meeting';
 
 type WebSocketBaseMessage = {
   op: string,
@@ -187,6 +188,8 @@ type AgentStatusUpdate = WebSocketBaseMessage & { data: AgentStatusUpdateEvent,
 type AgentPaused = WebSocketBaseMessage & { data: any, name: 'agent_paused' };
 type AgentUnpaused = WebSocketBaseMessage & { data: any, name: 'agent_unpaused' };
 
+type MeetingUserProgress = WebSocketBaseMessage & { data: Meeting, name: 'meeting_user_progress' };
+
 export type WebSocketMessage =
   | CallCreated
   | CallUpdated
@@ -219,4 +222,5 @@ export type WebSocketMessage =
   | FaxOutboundSucceeded
   | AgentStatusUpdate
   | AgentPaused
-  | AgentUnpaused;
+  | AgentUnpaused
+  | MeetingUserProgress;

package/src/web-rtc-client.js

@@ -478,7 +478,9 @@ export default class WebRTCClient extends Emitter {
     const inviteOptions: InviterInviteOptions = {
       requestDelegate: {
         onAccept: (response: IncomingResponse) => {
-          session.sessionDescriptionHandler.peerConnection.sfu = conference;
+          if (session.sessionDescriptionHandler.peerConnection) {
+            session.sessionDescriptionHandler.peerConnection.sfu = conference;
+          }
           this._onAccepted(session, response.session, true);
         },
         onReject: (response: IncomingResponse) => {
@@ -1350,6 +1352,16 @@ export default class WebRTCClient extends Emitter {
     return this.hasRemoteVideo(sessionId) ? this.getRemoteStream(sessionId) : null;
   }
 
+  //  Useful in a react-native environment when remoteMediaStream is not updated
+  getRemoteVideoStreamFromPc(sessionId: string): ?MediaStream {
+    const pc = this.getPeerConnection(sessionId);
+    if (!pc) {
+      return null;
+    }
+
+    return pc.getRemoteStreams().find(stream => !!stream.getVideoTracks().length);
+  }
+
   hasVideo(sessionId: string): boolean {
     return this.hasLocalVideo(sessionId) || this.hasRemoteVideo(sessionId);
   }
@@ -2128,18 +2140,22 @@ export default class WebRTCClient extends Emitter {
     });
 
     networkStats.packetsLost = packetsLost;
-    networkStats.audioBytesSent = audioBytesSent;
+    networkStats.totalAudioBytesSent = audioBytesSent;
+    networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
+    networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
-    networkStats.videoBytesSent = videoBytesSent;
-    networkStats.videoBytesReceived = videoBytesReceived;
+    networkStats.totalVideoBytesSent = videoBytesSent;
+    networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
+    networkStats.totalVideoBytesReceived = videoBytesReceived;
+    networkStats.videoBytesReceived = videoBytesReceived - lastVideoReceived;
+    networkStats.totalTransportSent = transportSent - lastTransportSent;
     networkStats.transportSent = transportSent;
-    networkStats.transportReceived = transportReceived;
-
-    const bandwidth = (networkStats.audioBytesSent - lastAudioSent) + (networkStats.videoBytesSent - lastVideoSent)
-      + (networkStats.audioBytesReceived - lastAudioReceived) + (networkStats.videoBytesReceived - lastVideoReceived)
-      + (networkStats.transportReceived - lastTransportReceived) + (networkStats.transportSent - lastTransportSent);
+    networkStats.totalTransportReceived = transportReceived;
+    networkStats.transportReceived = transportReceived - lastTransportReceived;
 
-    networkStats.bandwidth = bandwidth;
+    networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived
+      + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived
+      + networkStats.transportSent;
 
     this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
     this.sessionNetworkStats[sessionId].push(networkStats);

package/src/websocket-client.js

@@ -93,6 +93,7 @@ export const AGENT_PAUSED = 'agent_paused';
 export const AGENT_UNPAUSED = 'agent_unpaused';
 export const CONFERENCE_ADHOC_PARTICIPANT_LEFT = 'conference_adhoc_participant_left';
 export const CONFERENCE_ADHOC_DELETED = 'conference_adhoc_deleted';
+export const MEETING_USER_PROGRESS = 'meeting_user_progress';
 
 const BLACKLIST_EVENTS = [
   CHAT_MESSAGE_SENT,
@@ -446,6 +447,7 @@ WebSocketClient.eventLists = [
   AGENT_UNPAUSED,
   CONFERENCE_ADHOC_PARTICIPANT_LEFT,
   CONFERENCE_ADHOC_DELETED,
+  MEETING_USER_PROGRESS,
 ];
 
 export default WebSocketClient;