@wazo/sdk 0.38.17 → 0.38.19

package/lib/simple/room/RemoteParticipant.js

@@ -7030,6 +7030,7 @@ 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';
@@ -7113,6 +7114,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7181,6 +7183,7 @@ 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             ) {
@@ -7468,6 +7471,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8860,6 +8868,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8911,15 +8930,24 @@ 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$8.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;
@@ -9251,6 +9279,7 @@ 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';
@@ -9292,6 +9321,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10721,6 +10751,10 @@ 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', {
@@ -11004,6 +11038,7 @@ 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/lib/simple/room/Room.js

@@ -7032,6 +7032,7 @@ 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';
@@ -7115,6 +7116,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7183,6 +7185,7 @@ 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             ) {
@@ -7470,6 +7473,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8862,6 +8870,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8913,15 +8932,24 @@ 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$8.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;
@@ -9253,6 +9281,7 @@ 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';
@@ -9294,6 +9323,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10723,6 +10753,10 @@ 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', {
@@ -11006,6 +11040,7 @@ 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/lib/simple/room/SipRoom.js

@@ -7030,6 +7030,7 @@ 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';
@@ -7113,6 +7114,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7181,6 +7183,7 @@ 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             ) {
@@ -7468,6 +7471,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8860,6 +8868,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8911,15 +8930,24 @@ 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$8.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;
@@ -9251,6 +9279,7 @@ 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';
@@ -9292,6 +9321,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10721,6 +10751,10 @@ 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', {
@@ -11004,6 +11038,7 @@ 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/lib/simple/room/Stream.js

@@ -7030,6 +7030,7 @@ 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';
@@ -7113,6 +7114,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7181,6 +7183,7 @@ 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             ) {
@@ -7468,6 +7471,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8860,6 +8868,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8911,15 +8930,24 @@ 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$8.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;
@@ -9251,6 +9279,7 @@ 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';
@@ -9292,6 +9321,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10721,6 +10751,10 @@ 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', {
@@ -11004,6 +11038,7 @@ 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/lib/simple/room/__tests__/Participant.test.js

@@ -7030,6 +7030,7 @@ 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';
@@ -7113,6 +7114,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7181,6 +7183,7 @@ 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             ) {
@@ -7468,6 +7471,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8860,6 +8868,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8911,15 +8930,24 @@ 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$8.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;
@@ -9251,6 +9279,7 @@ 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';
@@ -9292,6 +9321,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10721,6 +10751,10 @@ 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', {
@@ -11004,6 +11038,7 @@ 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/lib/simple/room/__tests__/RemoteParticipant.test.js

@@ -7030,6 +7030,7 @@ 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';
@@ -7113,6 +7114,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                         
                           
 
   static isAPrivateIp(ip        )          {
@@ -7181,6 +7183,7 @@ 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             ) {
@@ -7468,6 +7471,11 @@ class WebRTCClient extends Emitter {
           }
           this._onAccepted(session, response.session, true);
         },
+        onProgress: (payload) => {
+          if (payload.message.statusCode === 183) {
+            this._onEarlyProgress(payload.session);
+          }
+        },
         onReject: (response                  ) => {
           logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
@@ -8860,6 +8868,17 @@ class WebRTCClient extends Emitter {
     };
   }
 
+  _onEarlyProgress(session               ) {
+    this._setupMedias(session);
+
+    const sessionId = this.getSipSessionId(session).substr(0, 20);
+    this.updateRemoteStream(sessionId);
+
+    logger$8.info('Early media progress progress received', { sessionId });
+
+    this.eventEmitter.emit(ON_EARLY_MEDIA$1, session);
+  }
+
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
     logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
@@ -8911,15 +8930,24 @@ 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$8.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;
@@ -9251,6 +9279,7 @@ 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';
@@ -9292,6 +9321,7 @@ const events = [
   ON_SIGNAL,
   ON_NETWORK_STATS,
   ON_DISCONNECTED,
+  ON_EARLY_MEDIA,
 ];
 
 const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
@@ -10721,6 +10751,10 @@ 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', {
@@ -11004,6 +11038,7 @@ 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,