@wazo/sdk 0.38.12 → 0.38.15

package/lib/simple/utils.js

@@ -4618,6 +4618,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -7208,6 +7210,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7429,6 +7437,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12304,10 +12313,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12950,7 +12964,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13052,7 +13066,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {
@@ -13720,7 +13734,7 @@ SwitchboardCall.STATE = {
 
 //      
 
-const FEATURES = ['chat', 'video', 'call_recording', 'fax', 'mobile_double_call', 'mobile_gsm'];
+const FEATURES = ['chat', 'video', 'call_recording', 'fax', 'mobile_double_call', 'mobile_gsm', 'meeting'];
 
 const getScopeName = (featureName        ) => `enterprise.app.${featureName}`;
 
@@ -13734,6 +13748,7 @@ class Features {
                    
                                 
                          
+                       
 
   constructor() {
     this._hasChat = true;
@@ -13742,6 +13757,7 @@ class Features {
     this._hasFax = true;
     this._hasMobileDoubleCall = true;
     this._hasMobileGsm = true;
+    this._hasMeeting = true;
   }
 
   async fetchAccess() {
@@ -13762,6 +13778,7 @@ class Features {
     this._hasFax = this._hasFeatures(scopes, 'fax');
     this._hasMobileDoubleCall = this._hasFeatures(scopes, 'mobile_double_call');
     this._hasMobileGsm = this._hasFeatures(scopes, 'mobile_gsm');
+    this._hasMeeting = this._hasFeatures(scopes, 'meeting');
   }
 
   hasChat() { return this._hasChat; }
@@ -13770,6 +13787,7 @@ class Features {
   hasFax() { return this._hasFax; }
   hasMobileDoubleCall() { return this._hasMobileDoubleCall; }
   hasMobileGsm() { return this._hasMobileGsm; }
+  hasMeeting() { return this._hasMeeting; }
 
   _hasFeatures(scopes        , featureName        ) {
     const scopeName = getScopeName(featureName);

package/lib/web-rtc-client.js

@@ -5797,6 +5797,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -6018,6 +6024,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.38.12",
+  "version": "0.38.15",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/domain/Features.js

@@ -2,7 +2,7 @@
 
 import getApiClient from '../service/getApiClient';
 
-export const FEATURES = ['chat', 'video', 'call_recording', 'fax', 'mobile_double_call', 'mobile_gsm'];
+export const FEATURES = ['chat', 'video', 'call_recording', 'fax', 'mobile_double_call', 'mobile_gsm', 'meeting'];
 
 export const getScopeName = (featureName: string) => `enterprise.app.${featureName}`;
 
@@ -16,6 +16,7 @@ class Features {
   _hasFax: boolean;
   _hasMobileDoubleCall: boolean;
   _hasMobileGsm: boolean;
+  _hasMeeting: boolean;
 
   constructor() {
     this._hasChat = true;
@@ -24,6 +25,7 @@ class Features {
     this._hasFax = true;
     this._hasMobileDoubleCall = true;
     this._hasMobileGsm = true;
+    this._hasMeeting = true;
   }
 
   async fetchAccess() {
@@ -44,6 +46,7 @@ class Features {
     this._hasFax = this._hasFeatures(scopes, 'fax');
     this._hasMobileDoubleCall = this._hasFeatures(scopes, 'mobile_double_call');
     this._hasMobileGsm = this._hasFeatures(scopes, 'mobile_gsm');
+    this._hasMeeting = this._hasFeatures(scopes, 'meeting');
   }
 
   hasChat() { return this._hasChat; }
@@ -52,6 +55,7 @@ class Features {
   hasFax() { return this._hasFax; }
   hasMobileDoubleCall() { return this._hasMobileDoubleCall; }
   hasMobileGsm() { return this._hasMobileGsm; }
+  hasMeeting() { return this._hasMeeting; }
 
   _hasFeatures(scopes: Object, featureName: string) {
     const scopeName = getScopeName(featureName);

package/src/domain/Phone/CTIPhone.js

@@ -341,4 +341,6 @@ export default class CTIPhone extends Emitter implements Phone {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }

package/src/domain/Phone/Phone.js

@@ -110,4 +110,6 @@ export interface Phone {
   isRegistered(): boolean;
 
   hasIncomingCallSession(): boolean;
+
+  setMediaConstraints(media: MediaStreamConstraints): PhoneVoid;
 }

package/src/simple/room/Participant.js

@@ -2,6 +2,7 @@
 import Emitter from '../../utils/Emitter';
 import IssueReporter from '../../service/IssueReporter';
 
+import Phone from '../Phone';
 import Room, { SIGNAL_TYPE_PARTICIPANT_UPDATE } from './Room';
 import Contact from '../../domain/Contact';
 import getApiClient from '../../service/getApiClient';
@@ -296,10 +297,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus: Object = null) {
+  broadcastStatus(inboundStatus: Object = null, sendReinvite: ?boolean) {
     const status = inboundStatus || this.getStatus();
     logger.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,

package/src/simple/room/Room.js

@@ -602,7 +602,7 @@ class Room extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -704,7 +704,7 @@ class Room extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/src/web-rtc-client.js

@@ -251,6 +251,12 @@ export default class WebRTCClient extends Emitter {
     };
 
     const ua = new UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage: string) => {
       const message = Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -472,6 +478,7 @@ export default class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {