@wazo/sdk 0.37.4 → 0.37.8

package/lib/simple/room/RemoteParticipant.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) {
@@ -12961,7 +12978,8 @@ 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/Room.js

@@ -7045,6 +7045,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7087,6 +7089,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8416,7 +8420,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8443,7 +8447,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8462,9 +8468,9 @@ 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);
     }
@@ -8494,6 +8500,11 @@ 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);
 
@@ -8656,6 +8667,12 @@ 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) {
@@ -12974,7 +12991,8 @@ 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/SipRoom.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/Stream.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/__tests__/RemoteParticipant.test.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/utils.js

@@ -7045,6 +7045,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -7087,6 +7089,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -8416,7 +8420,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession              ) {
+  _downgradeToAudio(callSession              , withMessage          = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -8443,7 +8447,9 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -8462,9 +8468,9 @@ 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);
     }
@@ -8494,6 +8500,11 @@ 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);
 
@@ -8656,6 +8667,12 @@ 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) {
@@ -11818,7 +11835,8 @@ 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/web-rtc-client.js

@@ -5802,6 +5802,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5844,6 +5846,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.37.4",
+  "version": "0.37.8",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/domain/Phone/WebRTCPhone.js

@@ -290,7 +290,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.client.onDisconnect();
   }
 
-  _downgradeToAudio(callSession: ?CallSession) {
+  _downgradeToAudio(callSession: ?CallSession, withMessage: boolean = true) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return false;
@@ -317,7 +317,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
       localStream.removeTrack(videoTrack);
     });
 
-    this._sendReinviteMessage(callSession, false);
+    if (withMessage) {
+      this._sendReinviteMessage(callSession, false);
+    }
 
     return true;
   }
@@ -336,9 +338,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     // 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);
     }
@@ -368,6 +370,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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);
 
@@ -532,6 +539,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
         // 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/src/simple/room/Room.js

@@ -813,7 +813,8 @@ 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: Object) => stream.id === streamId);
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream: Object) =>
+      stream && stream.id === streamId);
     return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
   }
 

package/src/web-rtc-client.js

@@ -706,6 +706,8 @@ export default class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -748,6 +750,8 @@ export default class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE