[{"data":1,"prerenderedAt":-1},["ShallowReactive",2],{"academy-blogs-en-1-1-all-ep116-realtime-voice-video-webrtc-websocket-go-all--*":3,"academy-blog-translations-kgoilbfmq2b2kt1":89},{"data":4,"page":77,"perPage":77,"totalItems":77,"totalPages":77},[5],{"alt":6,"collectionId":7,"collectionName":8,"content":9,"cover_image":10,"cover_image_path":11,"created":12,"created_by":13,"expand":14,"id":83,"keywords":84,"locale":59,"published_at":85,"scheduled_at":13,"school_blog":81,"short_description":86,"slug":87,"status":79,"title":6,"updated":88,"updated_by":13,"views":82},"EP.116 Real-time Voice\u002FVideo Communication with WebRTC + WebSocket Signaling in Go","sclblg987654321","school_blog_translations","\u003Cp>Text-based chat over WebSocket is just the beginning. Modern communication systems must support real-time voice and video features just like your favorite meeting apps.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Cp>And the global standard for real-time media transmission is WebRTC. But here’s the catch: WebRTC cannot initiate the connection by itself. It needs a mechanism called Signaling, and in this tutorial, we’ll use WebSocket as our signaling server.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Cp>This article will walk you through everything you need to know: From PeerConnection → ICE Candidates → Signaling → actual audio\u002Fvideo streaming all with real, production-grade Go code.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Ch2>🔊 What is WebRTC?\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cp>WebRTC is a framework for real-time peer-to-peer communication directly between browsers.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Ch3>✨ Key features:\u003C\u002Fh3>\u003Cul>\u003Cli>Low-latency audio\u002Fvideo (less than 100ms)\u003C\u002Fli>\u003Cli>True peer-to-peer connection (no central server for media)\u003C\u002Fli>\u003Cli>Secure by default (encrypted media streams)\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Ch2>🤔 What about WebSocket?\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cfigure class=\"table\">\u003Ctable>\u003Cthead>\u003Ctr>\u003Cth>Feature\u003C\u002Fth>\u003Cth>WebSocket\u003C\u002Fth>\u003Cth>WebRTC\u003C\u002Fth>\u003C\u002Ftr>\u003C\u002Fthead>\u003Ctbody>\u003Ctr>\u003Ctd>Real-time messaging\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>File transfer\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>Audio\u002FVideo streaming\u003C\u002Ftd>\u003Ctd>❌ Not ideal\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>Peer-to-peer connection\u003C\u002Ftd>\u003Ctd>❌\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>Sub-50ms latency\u003C\u002Ftd>\u003Ctd>❌\u003C\u002Ftd>\u003Ctd>✅\u003C\u002Ftd>\u003C\u002Ftr>\u003C\u002Ftbody>\u003C\u002Ftable>\u003C\u002Ffigure>\u003Cp> \u003C\u002Fp>\u003Cp>WebRTC is purpose-built for real-time media transmission.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Ch2>🛰 Components of WebRTC\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cul>\u003Cli>RTCPeerConnection\u003Cbr>Handles encrypted media connection between peers.\u003C\u002Fli>\u003Cli>ICE Candidate\u003Cbr>Determines possible network paths (LAN, Public IP, TURN server, etc.)\u003C\u002Fli>\u003Cli>SDP (Session Description Protocol)\u003Cbr>Describes media details (codec, bitrate, etc.)\u003C\u002Fli>\u003Cli>Signaling Server\u003Cbr>Responsible for exchanging messages like offer, answer, and ICE candidates between clients.\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Cp>⚠️ WebRTC does not include signaling you need to implement it yourself.\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Ch2>🔄 Using WebSocket as Signaling Server\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cp>Connection flow:\u003C\u002Fp>\u003Col>\u003Cli>👤 Client A creates an \u003Ccode inline=\"\">offer\u003C\u002Fcode> and sends it via WebSocket\u003C\u002Fli>\u003Cli>🔁 Server forwards the offer to Client B\u003C\u002Fli>\u003Cli>👤 Client B replies with an \u003Ccode inline=\"\">answer\u003C\u002Fcode>\u003C\u002Fli>\u003Cli>🔄 Both clients exchange ICE candidates\u003C\u002Fli>\u003Cli>✅ Once handshake is complete → media streaming begins!\u003C\u002Fli>\u003C\u002Fol>\u003Cp> \u003C\u002Fp>\u003Ch2>🧪 Go Example: WebSocket Signaling Server\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cpre>\u003Ccode class=\"language-plaintext language-go\">type SignalMessage struct {\n    Type string `json:\"type\"` \u002F\u002F offer, answer, ice\n    From string `json:\"from\"`\n    To   string `json:\"to\"`\n    Data json.RawMessage `json:\"data\"`\n}\n\nvar clients = make(map[string]*websocket.Conn)\n\nfunc signalingHandler(c *fiber.Ctx) error {\n    conn, err := upgrader.Upgrade(c.Context(), nil)\n    if err != nil { return err }\n\n    clientID := c.Query(\"id\")\n    clients[clientID] = conn\n\n    for {\n        _, msg, err := conn.ReadMessage()\n        if err != nil {\n            delete(clients, clientID)\n            return nil\n        }\n\n        var signal SignalMessage\n        json.Unmarshal(msg, &signal)\n\n        if target, ok := clients[signal.To]; ok {\n            target.WriteMessage(websocket.TextMessage, msg)\n        }\n    }\n}\n\u003C\u002Fcode>\u003C\u002Fpre>\u003Cp> \u003C\u002Fp>\u003Ch2>🎧 Client-side Code (JavaScript)\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Ch3>1. Create PeerConnection\u003C\u002Fh3>\u003Cpre>\u003Ccode class=\"language-plaintext language-js\">const pc = new RTCPeerConnection({\n  iceServers: [{ urls: \"stun:stun.l.google.com:19302\" }]\n});\n\npc.onicecandidate = event => {\n  if (event.candidate) {\n    socket.send(JSON.stringify({\n      type: \"ice\",\n      to: remoteUser,\n      data: event.candidate\n    }));\n  }\n};\n\u003C\u002Fcode>\u003C\u002Fpre>\u003Cp> \u003C\u002Fp>\u003Ch3>2. Start Call and Send Offer\u003C\u002Fh3>\u003Cpre>\u003Ccode class=\"language-plaintext language-js\">async function startCall() {\n  const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });\n  stream.getTracks().forEach(track => pc.addTrack(track, stream));\n\n  const offer = await pc.createOffer();\n  await pc.setLocalDescription(offer);\n\n  socket.send(JSON.stringify({\n    type: \"offer\",\n    to: remoteUser,\n    data: offer\n  }));\n}\n\u003C\u002Fcode>\u003C\u002Fpre>\u003Cp> \u003C\u002Fp>\u003Ch2>🔐 Using TURN Server (When P2P Fails)\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cp>In some scenarios, P2P connection might fail due to:\u003C\u002Fp>\u003Cul>\u003Cli>NAT restrictions\u003C\u002Fli>\u003Cli>Firewalls blocking ports\u003C\u002Fli>\u003Cli>Mobile networks\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Cp>In such cases, you’ll need a TURN server to relay the media Examples:\u003C\u002Fp>\u003Cul>\u003Cli>Coturn\u003C\u002Fli>\u003Cli>Twilio TURN Service\u003C\u002Fli>\u003Cli>Google Cloud WebRTC Infra\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Ch2>⚙️ Production-grade Best Practices\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cfigure class=\"table\">\u003Ctable>\u003Cthead>\u003Ctr>\u003Cth>Security Concern\u003C\u002Fth>\u003Cth>Recommended Strategy\u003C\u002Fth>\u003C\u002Ftr>\u003C\u002Fthead>\u003Ctbody>\u003Ctr>\u003Ctd>Token Reuse Prevention\u003C\u002Ftd>\u003Ctd>Use JWT with Device Binding\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>Stranger Calls\u003C\u002Ftd>\u003Ctd>Use Room-based Permission\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>DDOS \u002F Abuse\u003C\u002Ftd>\u003Ctd>Implement Rate Limits\u003C\u002Ftd>\u003C\u002Ftr>\u003Ctr>\u003Ctd>Media Relay\u003C\u002Ftd>\u003Ctd>Use Secure TURN (with auth)\u003C\u002Ftd>\u003C\u002Ftr>\u003C\u002Ftbody>\u003C\u002Ftable>\u003C\u002Ffigure>\u003Cp> \u003C\u002Fp>\u003Ch2>🚀 Challenge: Build Your First Video Call\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cp>✅ Implement a working prototype using:\u003C\u002Fp>\u003Cul>\u003Cli>WebSocket → Signaling\u003C\u002Fli>\u003Cli>WebRTC → P2P connection\u003C\u002Fli>\u003Cli>ICE Candidates exchange\u003C\u002Fli>\u003Cli>Media streaming\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Cp>Start with 1-on-1 calls → Expand to group call later!\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Chr>\u003Cp> \u003C\u002Fp>\u003Ch2>🔮 Coming Up Next: WebSocket for IoT Device Networks\u003C\u002Fh2>\u003Cp> \u003C\u002Fp>\u003Cp>In the next episode, we’ll explore building a real-time IoT network with WebSocket:\u003C\u002Fp>\u003Cul>\u003Cli>Manage thousands of devices (sensors, robots, smart home)\u003C\u002Fli>\u003Cli>Handle device auth & secure communication\u003C\u002Fli>\u003Cli>Optimize bandwidth for constrained networks\u003C\u002Fli>\u003C\u002Ful>\u003Cp> \u003C\u002Fp>\u003Cp>If you’ve made it this far you’re ready for real-time engineering at scale! 🎯\u003C\u002Fp>\u003Cp> \u003C\u002Fp>\u003Cdiv class=\"raw-html-embed\">\u003Cdiv style=\"margin:0 0 6px 0; font-weight:700;\">Read more:\u003C\u002Fdiv>\n\u003Cul style=\"list-style:none; padding:0; margin:0; line-height:1.4;\">\n  \u003Cli style=\"margin:0;\">\u003Ca href=\"\u002Fen\u002Fblogs\u002Fcategories\u002FGolang\" title=\"Golang The Series\">Golang The Series\u003C\u002Fa>\u003C\u002Fli>\n  \u003Cli style=\"margin:0;\">\u003Ca href=\"\u002Fen\u002Fblogs\u002Fcategories\u002FJS2GO\" title=\"JS2GO\">JS2GO\u003C\u002Fa>\u003C\u002Fli>\n  \u003Cli style=\"margin:0;\">\u003Ca href=\"\u002Fen\u002Fblogs\u002Fcategories\u002FTailwind%20CSS\" title=\"Tailwind CSS\">Tailwind CSS\u003C\u002Fa>\u003C\u002Fli>\n\u003C\u002Ful>\u003C\u002Fdiv>\u003Cp> \u003C\u002Fp>\u003Cdiv class=\"raw-html-embed\">\n  \u003Cp style=\"margin:0 0 6px 0;\">\u003Cstrong>Follow Us:\u003C\u002Fstrong>\u003C\u002Fp>\n  \u003Cul style=\"list-style:none; padding:0; margin:0; line-height: 0.4;\">\n    \u003Cli style=\"display:flex; align-items:center; gap:6px; margin:0;\">\n      \n      \u003Csvg width=\"16\" height=\"16\" viewBox=\"0 0 24 24\" fill=\"#1877F2\" aria-hidden=\"true\">\n        \u003Cpath d=\"M22 12.07C22 6.48 17.52 2 11.93 2S2 6.48 2 12.07c0 5 3.66 9.14 8.44 9.93v-7.02H7.9v-2.91h2.54V9.41c0-2.5 1.49-3.88 3.77-3.88 1.09 0 2.24.2 2.24.2v2.46h-1.26c-1.24 0-1.63.77-1.63 1.56v1.87h2.78l-.44 2.91h-2.34V22c4.78-.79 8.44-4.93 8.44-9.93Z\">\u003C\u002Fpath>\n      \u003C\u002Fsvg>\n      \u003Ca href=\"https:\u002F\u002Fwww.facebook.com\u002Fsuperdev.academy.th\" target=\"_blank\" rel=\"nofollow noopener\" title=\"Follow Superdev Academy on Facebook\">Facebook: Superdev Academy\u003C\u002Fa>\n    \u003C\u002Fli>\n\n    \u003Cli style=\"display:flex; align-items:center; gap:6px; margin:0;\">\n      \n      \u003Csvg 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08:33:53.210Z","59ty92ns80w_15oc1implw.png","https:\u002F\u002Ftwsme-r2.tumwebsme.com\u002Fsclcatblg987654321\u002Fwqxt7ag2gn7xcmk\u002F59ty92ns80w_15oc1implw.png",{"en":76,"th":76},"Golang The Series",1,"2026-03-16 04:39:38.440Z","published","2026-04-25 02:32:15.470Z","kgoilbfmq2b2kt1",216,"xkmzgoufvuwfzdv",[20,25,30,35,40,45,50],"2025-12-15 01:58:24.825Z","Learn how to build real-time video calls and voice chat using WebRTC and WebSocket Signaling in Go, including production-ready architecture, code examples, and best practices.","ep116-realtime-voice-video-webrtc-websocket-go","2026-04-25 02:47:49.171Z",{"en":87}]