@wazo/sdk 0.38.17 → 0.38.19

package/esm/__tests__/web-rtc-client.test.js

@@ -5608,6 +5608,7 @@ 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';
@@ -5688,6 +5689,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -5756,6 +5758,7 @@ 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             ) {
@@ -6043,6 +6046,11 @@ 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);
@@ -7435,6 +7443,17 @@ 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 });
 
@@ -7486,15 +7505,24 @@ 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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/esm/checker/Checker.js

@@ -6195,6 +6195,7 @@ 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';
@@ -6275,6 +6276,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -6343,6 +6345,7 @@ 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             ) {
@@ -6630,6 +6633,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$1.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8022,6 +8030,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$1.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$1.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8073,15 +8092,24 @@ 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$1.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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/esm/checker/checks/index.js

@@ -6195,6 +6195,7 @@ 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';
@@ -6275,6 +6276,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -6343,6 +6345,7 @@ 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             ) {
@@ -6630,6 +6633,11 @@ 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);
@@ -8022,6 +8030,17 @@ 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 });
 
@@ -8073,15 +8092,24 @@ 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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/esm/checker/checks/webrtc-transport.js

@@ -5608,6 +5608,7 @@ 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';
@@ -5688,6 +5689,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -5756,6 +5758,7 @@ 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             ) {
@@ -6043,6 +6046,11 @@ 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);
@@ -7435,6 +7443,17 @@ 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 });
 
@@ -7486,15 +7505,24 @@ 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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/esm/checker/checks/webrtc.js

@@ -5608,6 +5608,7 @@ 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';
@@ -5688,6 +5689,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -5756,6 +5758,7 @@ 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             ) {
@@ -6043,6 +6046,11 @@ 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);
@@ -7435,6 +7443,17 @@ 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 });
 
@@ -7486,15 +7505,24 @@ 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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;

package/esm/domain/Phone/WebRTCPhone.js

@@ -959,6 +959,7 @@ const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_CHAT = 'phone/ON_CHAT';
 const ON_SIGNAL = 'phone/ON_SIGNAL';
 const ON_DISCONNECTED = 'onDisconnected';
+const ON_EARLY_MEDIA = 'onEarlyMedia';
 
 const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
 const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
@@ -1000,6 +1001,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger = IssueReporter$1.loggerFor('webrtc-phone');
@@ -2429,6 +2431,10 @@ class WebRTCPhone extends Emitter                  {
       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', {
@@ -2670,4 +2676,4 @@ class WebRTCPhone extends Emitter                  {
 }
 
 export default WebRTCPhone;
-export { MESSAGE_TYPE_CHAT, MESSAGE_TYPE_SIGNAL, ON_AUDIO_STREAM, ON_CALL_ACCEPTED, ON_CALL_ANSWERED, ON_CALL_CANCELED, ON_CALL_ENDED, ON_CALL_ENDING, ON_CALL_ERROR, ON_CALL_FAILED, ON_CALL_HELD, ON_CALL_INCOMING, ON_CALL_MUTED, ON_CALL_OUTGOING, ON_CALL_REJECTED, ON_CALL_RESUMED, ON_CALL_UNHELD, ON_CALL_UNMUTED, ON_CAMERA_DISABLED, ON_CAMERA_RESUMED, ON_CHAT, ON_DISCONNECTED, ON_MESSAGE, ON_MESSAGE_TRACK_UPDATED, ON_NETWORK_STATS, ON_PLAY_HANGUP_SOUND, ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, ON_PLAY_PROGRESS_SOUND, ON_PLAY_RING_SOUND, ON_PROGRESS, ON_REGISTERED, ON_REINVITE, ON_REMOVE_STREAM, ON_SHARE_SCREEN_ENDED, ON_SHARE_SCREEN_ENDING, ON_SHARE_SCREEN_STARTED, ON_SIGNAL, ON_TERMINATE_SOUND, ON_TRACK, ON_UNREGISTERED, ON_USER_AGENT, ON_VIDEO_INPUT_CHANGE, ON_VIDEO_STREAM, events };
+export { MESSAGE_TYPE_CHAT, MESSAGE_TYPE_SIGNAL, ON_AUDIO_STREAM, ON_CALL_ACCEPTED, ON_CALL_ANSWERED, ON_CALL_CANCELED, ON_CALL_ENDED, ON_CALL_ENDING, ON_CALL_ERROR, ON_CALL_FAILED, ON_CALL_HELD, ON_CALL_INCOMING, ON_CALL_MUTED, ON_CALL_OUTGOING, ON_CALL_REJECTED, ON_CALL_RESUMED, ON_CALL_UNHELD, ON_CALL_UNMUTED, ON_CAMERA_DISABLED, ON_CAMERA_RESUMED, ON_CHAT, ON_DISCONNECTED, ON_EARLY_MEDIA, ON_MESSAGE, ON_MESSAGE_TRACK_UPDATED, ON_NETWORK_STATS, ON_PLAY_HANGUP_SOUND, ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, ON_PLAY_PROGRESS_SOUND, ON_PLAY_RING_SOUND, ON_PROGRESS, ON_REGISTERED, ON_REINVITE, ON_REMOVE_STREAM, ON_SHARE_SCREEN_ENDED, ON_SHARE_SCREEN_ENDING, ON_SHARE_SCREEN_STARTED, ON_SIGNAL, ON_TERMINATE_SOUND, ON_TRACK, ON_UNREGISTERED, ON_USER_AGENT, ON_VIDEO_INPUT_CHANGE, ON_VIDEO_STREAM, events };

package/esm/index.js

@@ -6511,6 +6511,7 @@ const MESSAGE = 'message';
 const ACCEPTED = 'accepted';
 const REJECTED = 'rejected';
 const ON_TRACK$1 = 'onTrack';
+const ON_EARLY_MEDIA$1 = 'onEarlyMedia';
 const ON_REINVITE$1 = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
@@ -6594,6 +6595,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -6662,6 +6664,7 @@ class WebRTCClient extends Emitter {
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS$1;
     this.ON_DISCONNECTED = ON_DISCONNECTED$1;
+    this.ON_EARLY_MEDIA = ON_EARLY_MEDIA$1;
   }
 
   configureMedia(media             ) {
@@ -6949,6 +6952,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$9.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8341,6 +8349,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$9.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$9.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8392,15 +8411,24 @@ 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$9.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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;
@@ -9640,6 +9668,7 @@ const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_CHAT = 'phone/ON_CHAT';
 const ON_SIGNAL = 'phone/ON_SIGNAL';
 const ON_DISCONNECTED = 'onDisconnected';
+const ON_EARLY_MEDIA = 'onEarlyMedia';
 
 const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
 const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
@@ -9681,6 +9710,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -11110,6 +11140,10 @@ class WebRTCPhone extends Emitter                  {
       logger$7.info('WebRTC disconnected');
       this.eventEmitter.emit(ON_DISCONNECTED);
     });
+    this.client.on(this.client.ON_EARLY_MEDIA, () => {
+      logger$7.info('WebRTC early media');
+      this.eventEmitter.emit(ON_EARLY_MEDIA);
+    });
 
     this.client.on(this.client.REGISTERED, () => {
       logger$7.info('WebRTC registered', {
@@ -11393,6 +11427,7 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   ON_CHAT: ON_CHAT,
   ON_SIGNAL: ON_SIGNAL,
   ON_DISCONNECTED: ON_DISCONNECTED,
+  ON_EARLY_MEDIA: ON_EARLY_MEDIA,
   MESSAGE_TYPE_CHAT: MESSAGE_TYPE_CHAT,
   MESSAGE_TYPE_SIGNAL: MESSAGE_TYPE_SIGNAL,
   events: events,

package/esm/simple/Auth.js

@@ -6689,6 +6689,7 @@ const MESSAGE = 'message';
 const ACCEPTED = 'accepted';
 const REJECTED = 'rejected';
 const ON_TRACK$1 = 'onTrack';
+const ON_EARLY_MEDIA$1 = 'onEarlyMedia';
 const ON_REINVITE$1 = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
@@ -6772,6 +6773,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -6840,6 +6842,7 @@ class WebRTCClient extends Emitter {
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS$1;
     this.ON_DISCONNECTED = ON_DISCONNECTED$1;
+    this.ON_EARLY_MEDIA = ON_EARLY_MEDIA$1;
   }
 
   configureMedia(media             ) {
@@ -7127,6 +7130,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$9.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8519,6 +8527,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$9.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$9.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8570,15 +8589,24 @@ 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$9.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 earlyStream = sipSession && sipSession.sessionDescriptionHandler
+      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
       return;
@@ -8910,6 +8938,7 @@ const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_CHAT = 'phone/ON_CHAT';
 const ON_SIGNAL = 'phone/ON_SIGNAL';
 const ON_DISCONNECTED = 'onDisconnected';
+const ON_EARLY_MEDIA = 'onEarlyMedia';
 
 const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
 const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
@@ -8951,6 +8980,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$8 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10380,6 +10410,10 @@ class WebRTCPhone extends Emitter                  {
       logger$8.info('WebRTC disconnected');
       this.eventEmitter.emit(ON_DISCONNECTED);
     });
+    this.client.on(this.client.ON_EARLY_MEDIA, () => {
+      logger$8.info('WebRTC early media');
+      this.eventEmitter.emit(ON_EARLY_MEDIA);
+    });
 
     this.client.on(this.client.REGISTERED, () => {
       logger$8.info('WebRTC registered', {
@@ -10663,6 +10697,7 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   ON_CHAT: ON_CHAT,
   ON_SIGNAL: ON_SIGNAL,
   ON_DISCONNECTED: ON_DISCONNECTED,
+  ON_EARLY_MEDIA: ON_EARLY_MEDIA,
   MESSAGE_TYPE_CHAT: MESSAGE_TYPE_CHAT,
   MESSAGE_TYPE_SIGNAL: MESSAGE_TYPE_SIGNAL,
   events: events,