diff --git a/app.js b/app.js
index 8c11bdc..62f0844 100644
--- a/app.js
+++ b/app.js
@@ -366,79 +366,22 @@ peers.on('connection', async socket => {
 
       console.log('🟠 videoCalls[callId].producerAudio', videoCalls[callId].producerAudio);
 
-      // Check if the router can consume the specified producer
-      if (
-        videoCalls[callId].router.canConsume({
-          producerId: videoCalls[callId].producerVideo.id,
-          rtpCapabilities
-        }) &&
-        videoCalls[callId].router.canConsume({
-          producerId: videoCalls[callId].producerAudio.id,
-          rtpCapabilities
-        })
-      ) {
-        console.log('[consume] Can consume', callId);
-        // Transport can now consume and return a consumer
-        videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
-          producerId: videoCalls[callId].producerVideo.id,
-          rtpCapabilities,
-          paused: true,
-        });
+      const canConsumeVideo = videoCalls[callId].producerVideo && videoCalls[callId].router.canConsume({
+        producerId: videoCalls[callId].producerVideo.id,
+        rtpCapabilities
+      })
 
-        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-        videoCalls[callId].consumerVideo.on('transportclose', () => {
-          const callId = socketDetails[socket.id];
-          console.log('transport close from consumer', callId);
-          closeCall(callId);
-        });
+      const canConsumeAudio = videoCalls[callId].producerAudio && videoCalls[callId].router.canConsume({
+        producerId: videoCalls[callId].producerAudio.id,
+        rtpCapabilities
+      })
 
-        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
-        videoCalls[callId].consumerVideo.on('producerclose', () => {
-          const callId = socketDetails[socket.id];
-          console.log('producer of consumer closed', callId);
-          closeCall(callId);
-        });
-
-
-        // videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
-        //   producerId: videoCalls[callId].producerAudio.id,
-        //   rtpCapabilities,
-        //   paused: true,
-        // });
-
-        // // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-        // videoCalls[callId].consumerAudio.on('transportclose', () => {
-        //   const callId = socketDetails[socket.id];
-        //   console.log('transport close from consumer', callId);
-        //   closeCall(callId);
-        // });
-
-        // // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
-        // videoCalls[callId].consumerAudio.on('producerclose', () => {
-        //   const callId = socketDetails[socket.id];
-        //   console.log('producer of consumer closed', callId);
-        //   closeCall(callId);
-        // });
-        console.log('🟠 videoCalls[callId].consumerAudio', videoCalls[callId].consumerAudio);
-
-        // From the consumer extract the following params to send back to the Client
-        const videoParams = {
-          id: videoCalls[callId].consumerVideo.id,
-          producerId: videoCalls[callId].producerVideo.id,
-          kind: 'video',
-          rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
-        };
-
-        const audioParams = {
-          id: videoCalls[callId].consumerAudio.id,
-          producerId: videoCalls[callId].producerAudio.id,
-          kind: 'audio',
-          rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
-        };
-
-        console.log('[consume] videoParams', videoParams);
-        console.log('[consume] audioParams', audioParams);
-        // Send the parameters to the client
+      if (canConsumeVideo && !canConsumeAudio) {
+        const videoParams = consumeVideo()
+        callback({ videoParams, audioParams: null });
+      } if (canConsumeVideo && canConsumeAudio) {
+        const videoParams = consumeVideo()
+        const audioParams = consumeAudio()
         callback({ videoParams, audioParams });
       } else {
         console.log(`[canConsume] Can't consume | callId ${callId}`);
@@ -466,6 +409,63 @@ peers.on('connection', async socket => {
   });
 });
 
+const consumeVideo = async () => {
+  videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
+    producerId: videoCalls[callId].producerVideo.id,
+    rtpCapabilities,
+    paused: true,
+  });
+
+  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+  videoCalls[callId].consumerVideo.on('transportclose', () => {
+    const callId = socketDetails[socket.id];
+    console.log('transport close from consumer', callId);
+    closeCall(callId);
+  });
+
+  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
+  videoCalls[callId].consumerVideo.on('producerclose', () => {
+    const callId = socketDetails[socket.id];
+    console.log('producer of consumer closed', callId);
+    closeCall(callId);
+  });
+
+  return {
+    id: videoCalls[callId].consumerVideo.id,
+    producerId: videoCalls[callId].producerVideo.id,
+    kind: 'video',
+    rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
+  }
+}
+
+const consumeAudio = async () => {
+  videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
+    producerId: videoCalls[callId].producerAudio.id,
+    rtpCapabilities,
+    paused: true,
+  });
+
+  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+  videoCalls[callId].consumerAudio.on('transportclose', () => {
+    const callId = socketDetails[socket.id];
+    console.log('transport close from consumer', callId);
+    closeCall(callId);
+  });
+
+  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
+  videoCalls[callId].consumerAudio.on('producerclose', () => {
+    const callId = socketDetails[socket.id];
+    console.log('producer of consumer closed', callId);
+    closeCall(callId);
+  });
+  return {
+    id: videoCalls[callId].consumerAudio.id,
+    producerId: videoCalls[callId].producerAudio.id,
+    kind: 'audio',
+    rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
+  }
+}
+
 /*
   - Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
   - It will return parameters, these are required for the client to create the RecvTransport