diff --git a/explainer.md b/explainer.md
index 147455f..9b69d4d 100644
--- a/explainer.md
+++ b/explainer.md
@@ -87,15 +87,14 @@ WebRTC-RtpTransport enables these use cases by enabling applications to:
 ```javascript
 const pc = new RTCPeerConnection({encodedInsertableStreams: true});
 const rtpTransport = pc.createRtpTransport();
-pc.getSenders().forEach((sender) => {
-  pc.createEncodedStreams().readable.
+// Do custom packetization on the first sender
+pc.getSenders()[0].createEncodedStreams().readable.
       pipeThrough(createPacketizingTransformer()).pipeTo(rtpTransport.writable);
-});
 
 function createPacketizingTransformer() {
   return new TransformStream({
     async transform(encodedFrame, controller) {
-      let rtpPackets = myPacketizer.packetize(frame);
+      let rtpPackets = myPacketizer.packetize(encodedFrame);
       rtpPackets.forEach(controller.enqueue);
     }
   });