mirror of
https://github.com/suitenumerique/meet.git
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✨(sip) add a sip software simulator to test unified encrypted flow
This commit is contained in:
Thomas Ramé
@@ -0,0 +1,405 @@
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<!DOCTYPE html>
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<!--
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Minimal SIP-via-Janus demo. Replaces the upstream siptest.html bundle with
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a single self-contained page. Everything is auto-driven: page load creates
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the Janus session, attaches to the SIP plugin, registers as guest, places
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the call. The user just needs to type the room PIN on the dialpad.
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Co-located only with janus.js (vendor lib, ~113KB) — no settings.js, no
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navbar/footer, no shared CSS, no logos. nginx serves these two files.
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-->
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<html lang="en">
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<head>
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<meta charset="utf-8">
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<title>SIP Demo</title>
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<style>
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* { box-sizing: border-box; }
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html, body { margin: 0; padding: 0; height: 100%; background: #1b1b1f; color: #e8e8e8;
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font: 14px/1.4 -apple-system, BlinkMacSystemFont, "Segoe UI", sans-serif; }
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#app { display: flex; flex-direction: column; height: 100vh; max-width: 720px; margin: 0 auto; padding: 16px; gap: 12px; }
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h1 { margin: 0; font-size: 18px; font-weight: 600; }
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#status { padding: 8px 12px; border-radius: 6px; background: #2c2c33; font-family: ui-monospace, monospace; font-size: 13px; }
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#video-wrap { position: relative; flex: 1; background: #000; border-radius: 8px; overflow: hidden; min-height: 240px; }
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#remoteVideo { width: 100%; height: 100%; object-fit: contain; }
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#noVideo { position: absolute; inset: 0; display: flex; align-items: center; justify-content: center; color: #555; font-size: 14px; }
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.row { display: flex; gap: 8px; flex-wrap: wrap; }
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button { font: inherit; padding: 8px 14px; border: 0; border-radius: 6px; background: #4a90e2; color: #fff; cursor: pointer; }
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button:hover:not(:disabled) { background: #357ec0; }
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button:disabled { opacity: 0.4; cursor: not-allowed; }
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button.danger { background: #dc3545; }
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button.danger:hover:not(:disabled) { background: #b62733; }
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button.success { background: #28a745; }
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button.success:hover:not(:disabled) { background: #1f8035; }
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.dialpad { display: grid; grid-template-columns: repeat(4, 1fr); gap: 6px; }
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.dialpad button { padding: 14px 0; font-size: 16px; background: #3a3a44; }
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.dialpad button:hover:not(:disabled) { background: #4a4a55; }
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#pin-row input { flex: 1; padding: 8px 12px; border: 1px solid #444; border-radius: 6px; background: #2c2c33; color: #fff; font: inherit; font-family: ui-monospace, monospace; letter-spacing: 0.2em; }
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#log { font-family: ui-monospace, monospace; font-size: 11px; max-height: 120px; overflow-y: auto; background: #15151a; padding: 8px; border-radius: 6px; color: #888; }
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.log-line { white-space: pre-wrap; word-break: break-word; }
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.log-err { color: #ff7373; }
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.disabled { opacity: 0.35; pointer-events: none; transition: opacity 0.2s; }
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</style>
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</head>
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<body>
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<div id="app">
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<h1>SIP demo (browser → Janus → livekit-sip → room)</h1>
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<div id="status">Loading…</div>
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<div id="video-wrap">
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<video id="remoteVideo" autoplay playsinline muted></video>
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<audio id="remoteAudio" autoplay></audio>
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<div id="noVideo">No remote video yet</div>
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</div>
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<div id="pin-row" class="row" style="display:none">
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<input type="text" id="pin" placeholder="Enter PIN then press # on the pad" inputmode="numeric" pattern="[0-9]*">
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<button id="sendPin" class="success">Send PIN</button>
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</div>
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<div id="dialpad" class="dialpad" style="display:none">
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<button data-d="1">1</button><button data-d="2">2</button><button data-d="3">3</button><button data-d="A">A</button>
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<button data-d="4">4</button><button data-d="5">5</button><button data-d="6">6</button><button data-d="B">B</button>
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<button data-d="7">7</button><button data-d="8">8</button><button data-d="9">9</button><button data-d="C">C</button>
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<button data-d="*">*</button><button data-d="0">0</button><button data-d="#">#</button><button data-d="D">D</button>
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</div>
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<div id="controls" class="row">
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<button id="muteBtn" class="danger">🔇 Mic muted</button>
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<button id="cameraBtn" class="success">📷 Camera on</button>
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<button id="speakerBtn" class="success">🔊 Speaker on</button>
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<button id="hangupBtn" class="danger" disabled>Hangup</button>
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<button id="rejoinBtn" style="display:none">Re-join</button>
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</div>
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<details><summary style="cursor:pointer; opacity:0.7; font-size:12px">debug log</summary>
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<div id="log"></div>
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</details>
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</div>
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<!-- webrtc-adapter is required by janus.js (cross-browser WebRTC API
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normalizer). The upstream siptest pulls it from CDN; we do the same.
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Without it Janus.init() silently fails. -->
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<script src="https://cdn.jsdelivr.net/npm/webrtc-adapter@9.0.1/out/adapter.min.js"></script>
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<script src="janus.js"></script>
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<script>
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// ============================================================================
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// Configuration (hard-coded; this is a dev demo)
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// ============================================================================
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const JANUS_SERVER = location.origin + "/janus"; // nginx proxies to janus
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const SIP_PROXY = "sip:sip:5060"; // docker service name
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const SIP_IDENTITY = "sip:demo@example.invalid"; // no registrar, guest
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const SIP_PEER = "sip:phone@sip:5060"; // user-part irrelevant
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// (Direct+PIN rule)
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const DISPLAY_NAME = "test-user";
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const ICE_SERVERS = []; // ICE-TCP via janus.jcfg
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// ============================================================================
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// UI helpers
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// ============================================================================
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const $ = id => document.getElementById(id);
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function setStatus(text) { $("status").textContent = text; log(text); }
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function log(msg, isErr = false) {
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const line = document.createElement("div");
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line.className = "log-line" + (isErr ? " log-err" : "");
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line.textContent = "[" + new Date().toLocaleTimeString() + "] " + msg;
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$("log").appendChild(line);
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$("log").scrollTop = $("log").scrollHeight;
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}
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// ============================================================================
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// Janus state
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// ============================================================================
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let janus = null;
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let sipcall = null;
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let inCall = false;
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// ============================================================================
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// Init: create Janus → session → SIP handle → register → call
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// ============================================================================
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Janus.init({
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debug: false,
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callback: () => {
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if (!Janus.isWebrtcSupported()) {
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setStatus("WebRTC not supported in this browser.");
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return;
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}
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createSession();
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}
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});
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function createSession() {
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setStatus("Connecting to Janus…");
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janus = new Janus({
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server: JANUS_SERVER,
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iceServers: ICE_SERVERS,
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success: () => {
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setStatus("Connected. Attaching SIP plugin…");
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attachSip();
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},
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error: (err) => setStatus("Janus error: " + err),
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destroyed: () => setStatus("Janus session destroyed."),
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});
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}
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function attachSip() {
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janus.attach({
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plugin: "janus.plugin.sip",
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opaqueId: "sip-demo-" + Janus.randomString(8),
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success: (handle) => {
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sipcall = handle;
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window.sipcall = handle; // for ad-hoc console debugging
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registerAsGuest();
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},
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error: (err) => setStatus("Plugin attach error: " + err),
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onmessage: handleMessage,
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onlocaltrack: (track, on) => {
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// We don't render local preview — Meet tab is already showing it.
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},
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onremotetrack: (track, mid, on, meta) => {
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if (!on) return;
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const stream = new MediaStream([track]);
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if (track.kind === "audio") {
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$("remoteAudio").srcObject = stream;
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} else if (track.kind === "video") {
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$("remoteVideo").srcObject = stream;
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$("noVideo").style.display = "none";
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}
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},
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oncleanup: () => { setStatus("Call cleaned up."); endCallUi(); },
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});
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}
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function registerAsGuest() {
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setStatus("Registering as guest…");
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sipcall.send({
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message: {
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request: "register",
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type: "guest",
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proxy: SIP_PROXY,
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username: SIP_IDENTITY,
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display_name: DISPLAY_NAME,
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},
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});
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}
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function placeCall() {
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setStatus("Calling " + SIP_PEER + "…");
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const tracks = [
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{ type: "audio", capture: true, recv: true },
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{ type: "video", capture: true, recv: true },
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];
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sipcall.createOffer({
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tracks: tracks,
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success: (jsep) => {
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sipcall.send({
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message: { request: "call", uri: SIP_PEER, autoaccept_reinvites: false },
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jsep: jsep,
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});
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},
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error: (err) => setStatus("createOffer failed: " + err),
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});
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}
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// ============================================================================
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// Plugin event router
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// ============================================================================
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function handleMessage(msg, jsep) {
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const result = msg && msg.result;
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const event = result && result.event;
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if (event) log("event: " + event);
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if (event === "registered") {
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setStatus("Registered. Placing call…");
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// Small delay to settle the registration
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setTimeout(placeCall, 100);
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}
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else if (event === "calling") {
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setStatus("Calling…");
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}
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else if (event === "accepted") {
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setStatus("Call accepted. You should hear the IVR — enter PIN on the dialpad.");
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startCallUi();
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if (jsep) sipcall.handleRemoteJsep({ jsep: jsep });
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}
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else if (event === "hangup") {
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setStatus("Call ended: " + (result.reason || result.code || ""));
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endCallUi();
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}
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else if (event === "missed_call") {
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setStatus("Missed call.");
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endCallUi();
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}
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else if (event === "registration_failed") {
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setStatus("Registration failed: " + result.code + " " + result.reason, true);
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}
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else if (msg && msg.error) {
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setStatus("Plugin error: " + msg.error, true);
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}
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if (jsep && event !== "accepted") {
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sipcall.handleRemoteJsep({ jsep: jsep });
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}
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}
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// ============================================================================
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// In-call UI: mute, hangup, dialpad, PIN entry
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// ============================================================================
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function startCallUi() {
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inCall = true;
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$("pin-row").style.display = "flex";
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$("pin-row").classList.remove("disabled");
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$("dialpad").style.display = "grid";
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$("dialpad").classList.remove("disabled");
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$("hangupBtn").disabled = false;
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$("rejoinBtn").style.display = "none";
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// Auto-mute mic so two tabs on the same Mac don't feedback-howl.
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if (sipcall.isAudioMuted && !sipcall.isAudioMuted()) {
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sipcall.muteAudio();
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}
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updateMuteButton();
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updateCameraButton();
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updateSpeakerButton();
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}
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function endCallUi() {
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inCall = false;
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$("pin-row").style.display = "none";
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$("pin-row").classList.remove("disabled");
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$("dialpad").style.display = "none";
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$("dialpad").classList.remove("disabled");
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$("hangupBtn").disabled = true;
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$("rejoinBtn").style.display = "inline-block";
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$("remoteVideo").srcObject = null;
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$("remoteAudio").srcObject = null;
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$("noVideo").style.display = "flex";
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}
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function updateMuteButton() {
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const b = $("muteBtn");
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if (!sipcall || !sipcall.isAudioMuted) return;
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if (sipcall.isAudioMuted()) {
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b.textContent = "🔇 Mic muted";
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b.className = "danger";
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} else {
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b.textContent = "🎤 Mic on";
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b.className = "success";
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}
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}
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function updateCameraButton() {
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const b = $("cameraBtn");
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if (!sipcall || !sipcall.isVideoMuted) return;
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if (sipcall.isVideoMuted()) {
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b.textContent = "📷 Camera off";
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b.className = "danger";
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} else {
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b.textContent = "📷 Camera on";
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b.className = "success";
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}
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}
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function updateSpeakerButton() {
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const b = $("speakerBtn");
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const audio = $("remoteAudio");
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if (audio.muted) {
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b.textContent = "🔇 Speaker off";
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b.className = "danger";
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} else {
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b.textContent = "🔊 Speaker on";
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b.className = "success";
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}
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}
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$("muteBtn").addEventListener("click", () => {
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if (!sipcall) return;
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if (sipcall.isAudioMuted()) sipcall.unmuteAudio();
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else sipcall.muteAudio();
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setTimeout(updateMuteButton, 50);
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});
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$("cameraBtn").addEventListener("click", () => {
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if (!sipcall || !sipcall.muteVideo) return;
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if (sipcall.isVideoMuted()) sipcall.unmuteVideo();
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else sipcall.muteVideo();
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setTimeout(updateCameraButton, 50);
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});
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$("speakerBtn").addEventListener("click", () => {
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const audio = $("remoteAudio");
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audio.muted = !audio.muted;
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updateSpeakerButton();
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});
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$("hangupBtn").addEventListener("click", () => {
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if (!sipcall) return;
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sipcall.send({ message: { request: "hangup" } });
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sipcall.hangup();
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});
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$("rejoinBtn").addEventListener("click", () => location.reload());
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// ============================================================================
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// DTMF — try the in-band RTP-event path first (RFC 4733 via the handle's
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// dtmf() method), which is what livekit-sip's IVR actually listens for.
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// Fall back to SIP INFO if RTP DTMF isn't available (older Janus, no
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// telephone-event in SDP, etc.).
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// ============================================================================
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function sendDtmf(digit) {
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if (!sipcall || !inCall) return;
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log("DTMF → " + digit);
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let sent = false;
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try {
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if (typeof sipcall.dtmf === "function") {
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sipcall.dtmf({
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dtmf: { tones: String(digit), duration: 200, gap: 70 },
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success: () => {},
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error: (err) => log("dtmf() error: " + err, true),
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});
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sent = true;
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}
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} catch (e) {
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log("dtmf() threw: " + e, true);
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}
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if (!sent) {
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// Fallback: SIP INFO method
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sipcall.send({
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message: { request: "dtmf_info", digit: String(digit) },
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});
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}
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}
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document.querySelectorAll(".dialpad button").forEach(b => {
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b.addEventListener("click", () => sendDtmf(b.dataset.d));
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});
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$("sendPin").addEventListener("click", () => {
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const pin = $("pin").value.replace(/[^\d#*]/g, "");
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if (!pin) return;
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const tones = (pin.endsWith("#") ? pin : pin + "#");
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log("DTMF batch → " + tones);
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// Send all tones in ONE call. Per-digit looping with setTimeout races
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// the WebRTC DTMF sender's internal queue and drops digits (we lost
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// the trailing "0" before "#" doing it digit-by-digit). The browser's
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// RTCDTMFSender paces them itself given duration/gap.
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try {
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sipcall.dtmf({
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dtmf: { tones: tones, duration: 400, gap: 100 },
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success: () => log("DTMF batch sent"),
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error: (err) => log("dtmf() error: " + err, true),
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});
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} catch (e) {
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log("dtmf() threw, falling back to SIP INFO per digit: " + e, true);
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for (const ch of tones) {
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sipcall.send({ message: { request: "dtmf_info", digit: String(ch) } });
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}
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}
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$("pin").value = "";
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// PIN + dialpad are no longer needed once you've submitted. Gateway
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// either bridges you in (correct PIN) or hangs up with wrong-pin
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// (the hangup event re-resets everything via endCallUi).
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$("pin-row").classList.add("disabled");
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$("dialpad").classList.add("disabled");
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});
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</script>
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</body>
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</html>
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