Adding a section explaining about ice servers.

examples/p2p-webrtc/voice-agent/README.md

@@ -46,6 +46,46 @@ http://localhost:7860
 
 ---
 
+## WebRTC ICE Servers Configuration
+
+When implementing WebRTC in your project, **STUN** (Session Traversal Utilities for NAT) and **TURN** (Traversal Using Relays around NAT) 
+servers are usually needed in cases where users are behind routers or firewalls.
+
+In local networks (e.g., testing within the same home or office network), you usually don’t need to configure STUN or TURN servers. 
+In such cases, WebRTC can often directly establish peer-to-peer connections without needing to traverse NAT or firewalls.
+
+### What are STUN and TURN Servers?
+
+- **STUN Server**: Helps clients discover their public IP address and port when they're behind a NAT (Network Address Translation) device (like a router). 
+This allows WebRTC to attempt direct peer-to-peer communication by providing the public-facing IP and port.
+  
+- **TURN Server**: Used as a fallback when direct peer-to-peer communication isn't possible due to strict NATs or firewalls blocking connections. 
+The TURN server relays media traffic between peers.
+
+### Why are ICE Servers Important?
+
+**ICE (Interactive Connectivity Establishment)** is a framework used by WebRTC to handle network traversal and NAT issues. 
+The `iceServers` configuration provides a list of **STUN** and **TURN** servers that WebRTC uses to find the best way to connect two peers. 
+
+### Example Configuration for ICE Servers
+
+Here’s how you can configure a basic `iceServers` object in WebRTC for testing purposes, using Google's public STUN server:
+
+```javascript
+const config = {
+  iceServers: [
+    {
+      urls: ["stun:stun.l.google.com:19302"], // Google's public STUN server
+    }
+  ],
+};
+```
+
+> For testing purposes, you can either use public **STUN** servers (like Google's) or set up your own **TURN** server. 
+If you're running your own TURN server, make sure to include your server URL, username, and credential in the configuration.
+
+---
+
 ### 💡 Notes
 - Ensure all dependencies are installed before running the server.
 - Check the `.env` file for missing configurations.

examples/p2p-webrtc/voice-agent/index.html

@@ -55,7 +55,7 @@
 
         const createSmallWebRTCConnection = async (audioTrack) => {
             const config = {
-              iceServers: [{ urls: ["stun:stun.l.google.com:19302"] }],
+              iceServers: [],
             };
             const pc = new RTCPeerConnection(config)
             addPeerConnectionEventListeners(pc)