@wazo/sdk 0.37.4 → 0.37.8

package/lib/checker/checks/webrtc.js

@@ -5797,6 +5797,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5839,6 +5841,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/lib/domain/Phone/WebRTCPhone.js

@@ -1050,7 +1050,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -1077,7 +1077,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -1096,9 +1098,9 @@ 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);
     }
@@ -1128,6 +1130,11 @@ 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);
 
@@ -1290,6 +1297,12 @@ 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/lib/index.js

@@ -6608,6 +6608,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6650,6 +6652,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8869,7 +8873,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8896,7 +8900,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8915,9 +8921,9 @@ 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);
     }
@@ -8947,6 +8953,11 @@ 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);
 
@@ -9109,6 +9120,12 @@ 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) {
@@ -12792,7 +12809,8 @@ 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/lib/simple/Auth.js

@@ -6783,6 +6783,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6825,6 +6827,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8154,7 +8158,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8181,7 +8185,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8200,9 +8206,9 @@ 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);
     }
@@ -8232,6 +8238,11 @@ 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);
 
@@ -8394,6 +8405,12 @@ 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) {
@@ -11556,7 +11573,8 @@ 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/lib/simple/Configuration.js

@@ -6781,6 +6781,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6823,6 +6825,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8152,7 +8156,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8179,7 +8183,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8198,9 +8204,9 @@ 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);
     }
@@ -8230,6 +8236,11 @@ 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);
 
@@ -8392,6 +8403,12 @@ 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) {
@@ -11554,7 +11571,8 @@ 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/lib/simple/Phone.js

@@ -6781,6 +6781,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6823,6 +6825,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8152,7 +8156,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8179,7 +8183,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8198,9 +8204,9 @@ 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);
     }
@@ -8230,6 +8236,11 @@ 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);
 
@@ -8392,6 +8403,12 @@ 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) {
@@ -11427,7 +11444,8 @@ 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/lib/simple/index.js

@@ -7043,6 +7043,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7085,6 +7087,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8414,7 +8418,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8441,7 +8445,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8460,9 +8466,9 @@ 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);
     }
@@ -8492,6 +8498,11 @@ 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);
 
@@ -8654,6 +8665,12 @@ 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) {
@@ -11816,7 +11833,8 @@ 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/lib/simple/room/LocalParticipant.js

@@ -7043,6 +7043,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7085,6 +7087,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8414,7 +8418,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8441,7 +8445,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8460,9 +8466,9 @@ 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);
     }
@@ -8492,6 +8498,11 @@ 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);
 
@@ -8654,6 +8665,12 @@ 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) {
@@ -12682,7 +12699,8 @@ 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/lib/simple/room/Participant.js

@@ -7043,6 +7043,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7085,6 +7087,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8414,7 +8418,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8441,7 +8445,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8460,9 +8466,9 @@ 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);
     }
@@ -8492,6 +8498,11 @@ 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);
 
@@ -8654,6 +8665,12 @@ 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) {
@@ -12687,7 +12704,8 @@ 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];
   }