@wazo/sdk 0.38.16 → 0.38.18

package/lib/web-rtc-client.js

@@ -4792,10 +4792,13 @@ var chatdMethods = (client              , baseUrl        ) => ({
   createRoom: async (name        , users                 )                    =>
     client.post(`${baseUrl}/users/me/rooms`, { name, users }).then(ChatRoom.parse),
 
-  getRoomMessages: async (roomUuid        )                              =>
-    client
-      .get(`${baseUrl}/users/me/rooms/${roomUuid}/messages`)
-      .then((response                         ) => ChatMessage.parseMany(response)),
+  getRoomMessages: async (roomUuid        , params                     = {})                              => {
+    const qs = ApiRequester.getQueryString(params);
+
+    return client
+      .get(`${baseUrl}/users/me/rooms/${roomUuid}/messages${qs.length ? `?${qs}` : ''}`)
+      .then((response                         ) => ChatMessage.parseMany(response));
+  },
 
   sendRoomMessage: async (roomUuid        , message             )                       =>
     client.post(`${baseUrl}/users/me/rooms/${roomUuid}/messages`, message).then(ChatMessage.parse),
@@ -5616,6 +5619,7 @@ const MESSAGE = 'message';
 const ACCEPTED = 'accepted';
 const REJECTED = 'rejected';
 const ON_TRACK = 'onTrack';
+const ON_EARLY_MEDIA = 'onEarlyMedia';
 const ON_REINVITE = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
@@ -5699,6 +5703,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -5767,6 +5772,7 @@ class WebRTCClient extends Emitter {
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS;
     this.ON_DISCONNECTED = ON_DISCONNECTED;
+    this.ON_EARLY_MEDIA = ON_EARLY_MEDIA;
   }
 
   configureMedia(media             ) {
@@ -6054,6 +6060,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -7446,6 +7457,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -7497,15 +7519,22 @@ class WebRTCClient extends Emitter {
 
   _setupMedias(session         , newStream              ) {
     // Safari hack, because you cannot call .play() from a non user action
-    const sessionId = this.getSipSessionId(session);
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
     const isConference = this.isConference(sessionId);
 
+    if (sessionId in this.audioElements) {
+      logger.info('html element already exists for session', { sessionId });
+      return;
+    }
+
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
       this.createAudioElementFor(sessionId);
     }
 
     const audioElement = this.audioElements[sessionId];
-    const stream = newStream || this.getRemoteStream(sessionId);
+    const sipSession = this.sipSessions[session.callId];
+    const removeStream = this.getRemoteStream(sessionId);
+    const stream = newStream || removeStream || sipSession.sessionDescriptionHandler.remoteMediaStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.38.16",
+  "version": "0.38.18",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/api/chatd.js

@@ -64,10 +64,13 @@ export default (client: ApiRequester, baseUrl: string) => ({
   createRoom: async (name: string, users: Array<ChatUser>): Promise<ChatRoom> =>
     client.post(`${baseUrl}/users/me/rooms`, { name, users }).then(ChatRoom.parse),
 
-  getRoomMessages: async (roomUuid: string): Promise<Array<ChatMessage>> =>
-    client
-      .get(`${baseUrl}/users/me/rooms/${roomUuid}/messages`)
-      .then((response: ChatMessageListResponse) => ChatMessage.parseMany(response)),
+  getRoomMessages: async (roomUuid: string, params: GetMessagesOptions = {}): Promise<Array<ChatMessage>> => {
+    const qs = ApiRequester.getQueryString(params);
+
+    return client
+      .get(`${baseUrl}/users/me/rooms/${roomUuid}/messages${qs.length ? `?${qs}` : ''}`)
+      .then((response: ChatMessageListResponse) => ChatMessage.parseMany(response));
+  },
 
   sendRoomMessage: async (roomUuid: string, message: ChatMessage): Promise<ChatMessage> =>
     client.post(`${baseUrl}/users/me/rooms/${roomUuid}/messages`, message).then(ChatMessage.parse),

package/src/domain/Phone/WebRTCPhone.js

@@ -54,6 +54,7 @@ export const ON_NETWORK_STATS = 'onNetworkStats';
 export const ON_CHAT = 'phone/ON_CHAT';
 export const ON_SIGNAL = 'phone/ON_SIGNAL';
 export const ON_DISCONNECTED = 'onDisconnected';
+export const ON_EARLY_MEDIA = 'onEarlyMedia';
 
 export const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
 export const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
@@ -95,6 +96,7 @@ export const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger = IssueReporter.loggerFor('webrtc-phone');
@@ -1526,6 +1528,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
       logger.info('WebRTC disconnected');
       this.eventEmitter.emit(ON_DISCONNECTED);
     });
+    this.client.on(this.client.ON_EARLY_MEDIA, () => {
+      logger.info('WebRTC early media');
+      this.eventEmitter.emit(ON_EARLY_MEDIA);
+    });
 
     this.client.on(this.client.REGISTERED, () => {
       logger.info('WebRTC registered', {

package/src/web-rtc-client.js

@@ -65,6 +65,7 @@ const MESSAGE = 'message';
 const ACCEPTED = 'accepted';
 const REJECTED = 'rejected';
 const ON_TRACK = 'onTrack';
+const ON_EARLY_MEDIA = 'onEarlyMedia';
 const ON_REINVITE = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
@@ -148,6 +149,7 @@ export default class WebRTCClient extends Emitter {
   ON_ERROR: string;
   ON_SCREEN_SHARING_REINVITE: string;
   ON_NETWORK_STATS: string;
+  ON_EARLY_MEDIA: string;
   ON_DISCONNECTED: string;
 
   static isAPrivateIp(ip: string): boolean {
@@ -216,6 +218,7 @@ export default class WebRTCClient extends Emitter {
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS;
     this.ON_DISCONNECTED = ON_DISCONNECTED;
+    this.ON_EARLY_MEDIA = ON_EARLY_MEDIA;
   }
 
   configureMedia(media: MediaConfig) {
@@ -503,6 +506,11 @@ export default class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response: IncomingResponse) => {
           logger.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -1895,6 +1903,17 @@ export default class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session: SessionDialog) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA, session);
+  }
+
   _onAccepted(session: Session, sessionDialog?: SessionDialog, withEvent: boolean = true) {
     logger.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -1946,15 +1965,22 @@ export default class WebRTCClient extends Emitter {
 
   _setupMedias(session: Session, newStream: ?MediaStream) {
     // Safari hack, because you cannot call .play() from a non user action
-    const sessionId = this.getSipSessionId(session);
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
     const isConference = this.isConference(sessionId);
 
+    if (sessionId in this.audioElements) {
+      logger.info('html element already exists for session', { sessionId });
+      return;
+    }
+
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
       this.createAudioElementFor(sessionId);
     }
 
     const audioElement = this.audioElements[sessionId];
-    const stream = newStream || this.getRemoteStream(sessionId);
+    const sipSession = this.sipSessions[session.callId];
+    const removeStream = this.getRemoteStream(sessionId);
+    const stream = newStream || removeStream || sipSession.sessionDescriptionHandler.remoteMediaStream;
 
     if (!this._isWeb() || !stream) {
       return;