@wazo/sdk 0.37.17 → 0.37.21

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

@@ -69,6 +69,7 @@ const ON_REINVITE = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
+const ON_DISCONNECTED = 'onDisconnected';
 
 export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
@@ -145,6 +146,7 @@ export default class WebRTCClient extends Emitter {
   ON_ERROR: string;
   ON_SCREEN_SHARING_REINVITE: string;
   ON_NETWORK_STATS: string;
+  ON_DISCONNECTED: string;
 
   static isAPrivateIp(ip: string): boolean {
     const regex = /^(?:10|127|172\.(?:1[6-9]|2[0-9]|3[01])|192\.168)\..*/;
@@ -210,6 +212,7 @@ export default class WebRTCClient extends Emitter {
     this.ON_ERROR = ON_ERROR;
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS;
+    this.ON_DISCONNECTED = ON_DISCONNECTED;
   }
 
   configureMedia(media: MediaConfig) {
@@ -475,7 +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) => {
@@ -811,7 +816,7 @@ export default class WebRTCClient extends Emitter {
       // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
       const transceivers = pc.getTransceivers();
       const transceiverIdx = lastIndexOf(transceivers, transceiver =>
-        transceiver.sender.track === null && transceiver.mid.indexOf('video') === -1);
+        transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
 
       videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
     } else {
@@ -1347,6 +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);
   }
@@ -1836,6 +1851,7 @@ export default class WebRTCClient extends Emitter {
 
     this.updateRemoteStream(this.getSipSessionId(session));
 
+    const pc = session.sessionDescriptionHandler.peerConnection;
     const onTrack = (event: any) => {
       const isAudioOnly = this._isAudioOnly(session);
       const { kind, label, readyState, id, muted } = event.track;
@@ -1855,6 +1871,12 @@ export default class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
+    pc.oniceconnectionstatechange = () => {
+      if (pc.iceConnectionState === 'disconnected') {
+        this.eventEmitter.emit(ON_DISCONNECTED);
+      }
+    };
+
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
     }
@@ -2074,6 +2096,7 @@ export default class WebRTCClient extends Emitter {
     let videoBytesReceived = 0;
     let transportSent = 0;
     let transportReceived = 0;
+    let packetsLost = 0;
 
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
@@ -2086,7 +2109,7 @@ export default class WebRTCClient extends Emitter {
         videoBytesSent += report.bytesSent + report.headerBytesSent;
       }
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-        networkStats.packetsLost = report.packetsLost;
+        packetsLost += report.packetsLost;
         networkStats.packetsReceived = report.packetsReceived;
         audioBytesReceived += report.bytesReceived + report.headerBytesReceived;
       }
@@ -2103,27 +2126,36 @@ export default class WebRTCClient extends Emitter {
         }
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
+        packetsLost += report.packetsLost;
         networkStats.roundTripTime = report.roundTripTime;
         networkStats.jitter = report.jitter;
       }
+      if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
+        packetsLost += report.packetsLost;
+      }
       if (report.type === 'transport') {
         transportSent += report.bytesSent;
         transportReceived += report.bytesReceived;
       }
     });
 
-    networkStats.audioBytesSent = audioBytesSent;
+    networkStats.packetsLost = packetsLost;
+    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;