@wazo/sdk 0.37.4 → 0.37.8

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

@@ -5788,6 +5788,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5830,6 +5832,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/esm/checker/Checker.js

@@ -6371,6 +6371,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6413,6 +6415,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/esm/checker/checks/index.js

@@ -6371,6 +6371,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6413,6 +6415,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

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

@@ -5788,6 +5788,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5830,6 +5832,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/esm/checker/checks/webrtc.js

@@ -5788,6 +5788,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5830,6 +5832,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/esm/domain/Phone/WebRTCPhone.js

@@ -1041,7 +1041,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -1068,7 +1068,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -1087,9 +1089,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -1119,6 +1121,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -1281,6 +1288,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {

package/esm/index.js

@@ -6597,6 +6597,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6639,6 +6641,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8858,7 +8862,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8885,7 +8889,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8904,9 +8910,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8936,6 +8942,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -9098,6 +9109,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -12781,7 +12798,8 @@ class Room extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/esm/simple/Auth.js

@@ -6771,6 +6771,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6813,6 +6815,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8142,7 +8146,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8169,7 +8173,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8188,9 +8194,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8220,6 +8226,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -8382,6 +8393,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -11544,7 +11561,8 @@ class Room$1 extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/esm/simple/Configuration.js

@@ -6771,6 +6771,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6813,6 +6815,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8142,7 +8146,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8169,7 +8173,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8188,9 +8194,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8220,6 +8226,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -8382,6 +8393,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -11544,7 +11561,8 @@ class Room$1 extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/esm/simple/Phone.js

@@ -6771,6 +6771,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6813,6 +6815,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8142,7 +8146,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8169,7 +8173,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8188,9 +8194,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8220,6 +8226,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -8382,6 +8393,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -11417,7 +11434,8 @@ class Room$1 extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/esm/simple/index.js

@@ -7033,6 +7033,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7075,6 +7077,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8404,7 +8408,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8431,7 +8435,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8450,9 +8456,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8482,6 +8488,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -8644,6 +8655,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -11806,7 +11823,8 @@ class Room$1 extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/esm/simple/room/LocalParticipant.js

@@ -7033,6 +7033,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7075,6 +7077,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8404,7 +8408,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8431,7 +8435,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8450,9 +8456,9 @@ class WebRTCPhone extends Emitter                  {
 
     // Check if a video sender already exists
     let videoSender;
+    const videoTransceiver = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
     if (isConference) {
-      const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans ? trans.sender : null;
+      videoSender = videoTransceiver ? videoTransceiver.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -8482,6 +8488,11 @@ class WebRTCPhone extends Emitter                  {
       videoSender.replaceTrack(videoTrack);
     }
 
+    // Put back last direction (was set to `recvonly` when downgrading)
+    if (videoTransceiver && videoTransceiver.currentDirection !== videoTransceiver.direction) {
+      videoTransceiver.direction = videoTransceiver.currentDirection;
+    }
+
     this.client.setLocalMediaStream(this.getSipSessionId(sipSession), newStream);
     this._sendReinviteMessage(callSession, true);
 
@@ -8644,6 +8655,12 @@ class WebRTCPhone extends Emitter                  {
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
+        if (!conference) {
+          // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+          this._downgradeToAudio(targetCallSession, false);
+          // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+          await new Promise(resolve => setTimeout(resolve, 500));
+        }
         reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
@@ -12672,7 +12689,8 @@ class Room extends Emitter {
     }
 
     // Find in all the streams by streamId (used on FF where we can't map by trackId)
-    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }