diff --git a/plugins/webrtc/package-lock.json b/plugins/webrtc/package-lock.json
index 12726f82d..c63571620 100644
--- a/plugins/webrtc/package-lock.json
+++ b/plugins/webrtc/package-lock.json
@@ -1,12 +1,12 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.3",
+   "version": "0.1.18",
    "lockfileVersion": 2,
    "requires": true,
    "packages": {
       "": {
          "name": "@scrypted/webrtc",
-         "version": "0.1.3",
+         "version": "0.1.18",
          "dependencies": {
             "@scrypted/common": "file:../../common",
             "@scrypted/sdk": "file:../../sdk",
diff --git a/plugins/webrtc/package.json b/plugins/webrtc/package.json
index 7e5230454..d95aa3839 100644
--- a/plugins/webrtc/package.json
+++ b/plugins/webrtc/package.json
@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.3",
+   "version": "0.1.18",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",
diff --git a/plugins/webrtc/src/main.ts b/plugins/webrtc/src/main.ts
index 3d0f6261b..3d059b162 100644
--- a/plugins/webrtc/src/main.ts
+++ b/plugins/webrtc/src/main.ts
@@ -72,14 +72,14 @@ class WebRTCMixin extends SettingsMixinDeviceBase<RTCSignalingClient & VideoCame
         if ((this.type === ScryptedDeviceType.Speaker || this.type === ScryptedDeviceType.SmartSpeaker)
             && this.mixinDeviceInterfaces.includes(ScryptedInterface.RTCSignalingChannel)) {
 
-                this.console.log('starting webrtc speaker intercom');
+            this.console.log('starting webrtc speaker intercom');
 
             const pc = new RTCPeerConnection();
             const atrack = new MediaStreamTrack({ kind: 'audio' });
             const audioTransceiver = pc.addTransceiver(atrack);
             const weriftSignalingSession = new WeriftSignalingSession(this.console, pc);
             const control = await this.mixinDevice.startRTCSignalingSession(weriftSignalingSession);
-            
+
             const forwarder = await createTrackForwarder({
                 timeStart: Date.now(),
                 videoTransceiver: undefined,
@@ -509,7 +509,7 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
 
 export async function fork() {
     return {
-        async createConnection(message: any, port: number, clientSession: RTCSignalingSession, maximumCompatibilityMode: boolean, transcodeWidth: number, sessionSupportsH264High: boolean, options: { disableIntercom?: boolean; configuration: RTCConfiguration, weriftConfiguration: PeerConfig; }) {
+        async createConnection(message: any, portOrDummy: number | boolean, clientSession: RTCSignalingSession, maximumCompatibilityMode: boolean, transcodeWidth: number, sessionSupportsH264High: boolean, options: { disableIntercom?: boolean; configuration: RTCConfiguration, weriftConfiguration: PeerConfig; }) {
             const cleanup = new Deferred<string>();
             cleanup.promise.catch(e => this.console.log('cleaning up rtc connection:', e.message));
             cleanup.promise.finally(() => setTimeout(() => process.exit(), 10000));
@@ -529,7 +529,8 @@ export async function fork() {
                 }
             }
 
-            if (port) {
+            if (typeof portOrDummy === 'number') {
+                const port = portOrDummy;
                 const socket = net.connect(port, '127.0.0.1');
                 cleanup.promise.finally(() => socket.destroy());
 
@@ -548,8 +549,10 @@ export async function fork() {
             }
             else {
                 pc.createDataChannel('dummy');
-                const offer = await pc.createOffer();
-                pc.setLocalDescription(offer);
+                if (portOrDummy) {
+                    const offer = await pc.createOffer();
+                    pc.setLocalDescription(offer);
+                }
             }
 
             return connection;