diff --git a/webrtc/signalling/Protocol.md b/webrtc/signalling/Protocol.md
index 1af892eee3..18ef5fa430 100644
--- a/webrtc/signalling/Protocol.md
+++ b/webrtc/signalling/Protocol.md
@@ -1,26 +1,26 @@
 # Terminology
 
-### Client
+## Client
 
 A GStreamer-based application
 
-### Browser
+## Browser
 
 A JS application that runs in the browser and uses built-in browser webrtc APIs
 
-### Peer
+## Peer
  
 Any webrtc-using application that can participate in a call
 
-### Signalling server
+## Signalling server
 
 Basic websockets server implemented in Python that manages the peers list and shovels data between peers
 
-# Overview
+### Overview
 
 This is a basic protocol for doing 1-1 audio+video calls between a gstreamer app and a JS app in a browser.
 
-# Peer registration and calling
+### Peer registration and calling
 
 Peers must register with the signalling server before a call can be initiated. The server connection should stay open as long as the peer is available or in a call.
 
@@ -38,7 +38,7 @@ This protocol builds upon https://github.com/shanet/WebRTC-Example/
 * Closure of the server connection means the call has ended; either because the other peer ended it or went away
 * To end the call, disconnect from the server. You may reconnect again whenever you wish.
 
-# Negotiation
+### Negotiation
 
 Once a call has been setup with the signalling server, the peers must negotiate SDP and ICE candidates with each other.