Merge pull request #1581 from pipecat-ai/voice_agent_ice_servers
Configuring the voice-agent example to wait for all the ice candidates.
This commit is contained in:
@@ -46,6 +46,46 @@ http://localhost:7860
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---
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## WebRTC ICE Servers Configuration
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When implementing WebRTC in your project, **STUN** (Session Traversal Utilities for NAT) and **TURN** (Traversal Using Relays around NAT)
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servers are usually needed in cases where users are behind routers or firewalls.
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In local networks (e.g., testing within the same home or office network), you usually don’t need to configure STUN or TURN servers.
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In such cases, WebRTC can often directly establish peer-to-peer connections without needing to traverse NAT or firewalls.
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### What are STUN and TURN Servers?
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- **STUN Server**: Helps clients discover their public IP address and port when they're behind a NAT (Network Address Translation) device (like a router).
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This allows WebRTC to attempt direct peer-to-peer communication by providing the public-facing IP and port.
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- **TURN Server**: Used as a fallback when direct peer-to-peer communication isn't possible due to strict NATs or firewalls blocking connections.
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The TURN server relays media traffic between peers.
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### Why are ICE Servers Important?
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**ICE (Interactive Connectivity Establishment)** is a framework used by WebRTC to handle network traversal and NAT issues.
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The `iceServers` configuration provides a list of **STUN** and **TURN** servers that WebRTC uses to find the best way to connect two peers.
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### Example Configuration for ICE Servers
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Here’s how you can configure a basic `iceServers` object in WebRTC for testing purposes, using Google's public STUN server:
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```javascript
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const config = {
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iceServers: [
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{
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urls: ["stun:stun.l.google.com:19302"], // Google's public STUN server
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}
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],
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};
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```
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> For testing purposes, you can either use public **STUN** servers (like Google's) or set up your own **TURN** server.
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If you're running your own TURN server, make sure to include your server URL, username, and credential in the configuration.
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---
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### 💡 Notes
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- Ensure all dependencies are installed before running the server.
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- Check the `.env` file for missing configurations.
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@@ -24,27 +24,47 @@
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let connected = false
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let peerConnection = null
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/*const waitForIceGatheringComplete = async (pc) => {
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const waitForIceGatheringComplete = async (pc, timeoutMs = 2000) => {
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if (pc.iceGatheringState === 'complete') return;
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console.log("Waiting for ICE gathering to complete. Current state:", pc.iceGatheringState);
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return new Promise((resolve) => {
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let timeoutId;
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const checkState = () => {
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console.log("icegatheringstatechange:", pc.iceGatheringState);
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if (pc.iceGatheringState === 'complete') {
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pc.removeEventListener('icegatheringstatechange', checkState);
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cleanup();
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resolve();
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}
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};
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const onTimeout = () => {
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console.warn(`ICE gathering timed out after ${timeoutMs} ms.`);
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cleanup();
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resolve();
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};
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const cleanup = () => {
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pc.removeEventListener('icegatheringstatechange', checkState);
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clearTimeout(timeoutId);
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};
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pc.addEventListener('icegatheringstatechange', checkState);
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timeoutId = setTimeout(onTimeout, timeoutMs);
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// Checking the state again to avoid any eventual race condition
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checkState();
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});
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}*/
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};
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const createSmallWebRTCConnection = async (audioTrack) => {
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const pc = new RTCPeerConnection()
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const config = {
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iceServers: [],
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};
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const pc = new RTCPeerConnection(config)
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addPeerConnectionEventListeners(pc)
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pc.ontrack = e => audioEl.srcObject = e.streams[0]
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// SmallWebRTCTransport expects to receive both transceivers
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pc.addTransceiver(audioTrack, { direction: 'sendrecv' })
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pc.addTransceiver('video', { direction: 'sendrecv' })
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await pc.setLocalDescription(await pc.createOffer())
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//await waitForIceGatheringComplete(pc)
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await waitForIceGatheringComplete(pc)
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const offer = pc.localDescription
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const response = await fetch('/api/offer', {
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body: JSON.stringify({ sdp: offer.sdp, type: offer.type}),
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@@ -57,16 +77,37 @@
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}
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const connect = async () => {
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_onConnecting()
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const audioStream = await navigator.mediaDevices.getUserMedia({audio: true})
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peerConnection= await createSmallWebRTCConnection(audioStream.getAudioTracks()[0])
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peerConnection.onconnectionstatechange = () => {
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let connectionState = peerConnection?.connectionState
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}
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const addPeerConnectionEventListeners = (pc) => {
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pc.oniceconnectionstatechange = () => {
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console.log("oniceconnectionstatechange", pc?.iceConnectionState)
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}
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pc.onconnectionstatechange = () => {
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console.log("onconnectionstatechange", pc?.connectionState)
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let connectionState = pc?.connectionState
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if (connectionState === 'connected') {
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_onConnected()
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} else if (connectionState === 'disconnected') {
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_onDisconnected()
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}
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}
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pc.onicecandidate = (event) => {
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if (event.candidate) {
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console.log("New ICE candidate:", event.candidate);
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} else {
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console.log("All ICE candidates have been sent.");
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}
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};
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}
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const _onConnecting = () => {
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statusEl.textContent = "Connecting"
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buttonEl.textContent = "Disconnect"
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connected = true
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}
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const _onConnected = () => {
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