@sphoro/bugcatch-web-sdk 0.17.0 → 0.18.0

package/dist/bugcatch-call.es.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"bugcatch-call.es.js","sources":["../src/audio.js","../src/live-call.js","../src/incoming-call.js"],"sourcesContent":["// Microphone to µ-law, and back.\n//\n// sphoro.voice speaks G.711 µ-law at 8 kHz in both directions — the format a\n// telephone carrier hands it, which is what makes a browser call and a phone\n// call indistinguishable to everything downstream. A browser has neither: it\n// captures 32-bit floats at whatever rate the sound card runs at. The gap\n// between those two is a codec and a resampler, and neither is available from\n// the platform, so both are here.\n//\n// This file exists rather than the two callers each having a copy. The agent\n// call sends µ-law because that is the whole conversation; a live call sends it\n// because sphoro.voice cannot record or transcribe a conversation it is not in\n// the middle of, and asks each browser for a copy of its own microphone. Two\n// implementations of G.711 in one package would be one implementation and one\n// that is subtly wrong on the day somebody fixes only the first.\n//\n// Nothing is imported to do it. The SDK runs on someone else's page and has no\n// dependencies, and that rule does not bend for audio.\n\n/** The wire's sample rate. Not negotiable — see the note above. */\nexport const WIRE_RATE = 8000;\n\n/**\n * An audio context, at the wire's rate where the browser will give us one.\n *\n * Asking for 8 kHz outright makes the resampler below dead code on most\n * machines. Some browsers refuse a rate their hardware does not like, so the\n * fallback is an ordinary context and the manual path.\n */\nexport function createContext(rate = WIRE_RATE) {\n  const Ctor = window.AudioContext || window.webkitAudioContext;\n  try {\n    return new Ctor({ sampleRate: rate });\n  } catch {\n    return new Ctor();\n  }\n}\n\n/**\n * Start a context that the browser has parked, if it will let us.\n *\n * Resolves either way. There is nothing useful to do with the rejection: it\n * means this browser wants a user gesture we are not inside, and the call is\n * still worth attempting — a context that stays suspended is a silent call,\n * which is what the caller finds out about from the states it is already being\n * sent.\n */\nexport async function resumeContext(context) {\n  if (context.state !== 'suspended') return;\n  try {\n    await context.resume();\n  } catch { /* no activation to spend */ }\n}\n\n/**\n * Drop a signal to the wire rate by averaging each source window.\n *\n * Exported for its own test: the browsers that refuse an 8 kHz context are the\n * ones that need this, and they are exactly the browsers a developer is least\n * likely to be looking at when they change it.\n *\n * Averaging rather than picking every nth sample: plain decimation aliases\n * everything above 4 kHz down into the speech band, and consonants are exactly\n * what lives up there. This is a crude low-pass, but it is the difference\n * between \"sixty\" and \"sixteen\" surviving the trip.\n */\nexport function downsample(input, fromRate, toRate) {\n  const ratio = fromRate / toRate;\n  const out = new Float32Array(Math.floor(input.length / ratio));\n  for (let i = 0; i < out.length; i++) {\n    const start = Math.floor(i * ratio);\n    const end = Math.min(Math.floor((i + 1) * ratio), input.length);\n    let sum = 0;\n    for (let j = start; j < end; j++) sum += input[j];\n    out[i] = end > start ? sum / (end - start) : 0;\n  }\n  return out;\n}\n\n/** G.711 µ-law: the bias and clip the standard specifies. */\nconst MU_BIAS = 0x84;\nconst MU_CLIP = 32635;\n\n/**\n * Float samples to µ-law bytes, ready to put on the wire.\n *\n * Exported for the codec test rather than for callers: this file is not the\n * package's entry point, and a hand-rolled G.711 that nothing checks is a\n * silent call nobody can debug from the outside.\n */\nexport function encodeMuLaw(samples) {\n  const out = new Uint8Array(samples.length);\n  for (let i = 0; i < samples.length; i++) {\n    // Clamped before scaling: a float outside [-1, 1] is legal in Web Audio and\n    // would wrap round to the opposite sign as an integer.\n    const clamped = Math.max(-1, Math.min(1, samples[i]));\n    let sample = Math.round(clamped * 32767);\n\n    const sign = sample < 0 ? 0x80 : 0;\n    if (sign) sample = -sample;\n    if (sample > MU_CLIP) sample = MU_CLIP;\n    sample += MU_BIAS;\n\n    let exponent = 7;\n    for (let mask = 0x4000; (sample & mask) === 0 && exponent > 0; exponent--, mask >>= 1);\n    const mantissa = (sample >> (exponent + 3)) & 0x0F;\n    out[i] = ~(sign | (exponent << 4) | mantissa) & 0xFF;\n  }\n  return out;\n}\n\n/** µ-law bytes back to float samples, ready for an AudioBuffer. */\nexport function decodeMuLaw(bytes) {\n  const out = new Float32Array(bytes.length);\n  for (let i = 0; i < bytes.length; i++) {\n    const byte = ~bytes[i] & 0xFF;\n    const sign = byte & 0x80;\n    const exponent = (byte >> 4) & 0x07;\n    const mantissa = byte & 0x0F;\n    const magnitude = (((mantissa << 3) + MU_BIAS) << exponent) - MU_BIAS;\n    out[i] = (sign ? -magnitude : magnitude) / 32768;\n  }\n  return out;\n}\n\n/**\n * How much audio rides in one frame.\n *\n * Measured in samples, and the context above runs at 8 kHz wherever the browser\n * allows it — so this is 64 ms of speech, not the 20-odd it would be at\n * hardware rate. That distinction is the whole reason for the number: at 2048\n * the same code sends a quarter of a second per frame, which is a quarter of a\n * second added to the start of every turn and to every barge-in, and it sounds\n * exactly like an agent that is slow to notice you talking.\n *\n * Not smaller than this. ScriptProcessorNode runs on the main thread, and a\n * block it cannot fill in time is a block it drops — which is a gap in the\n * middle of a word rather than a delay before one. Powers of two only.\n */\nexport const CAPTURE_BLOCK = 512;\n\n/**\n * Tap a microphone and hand each block over as µ-law bytes.\n *\n * ScriptProcessorNode is deprecated in favour of AudioWorklet, and is used\n * anyway. A worklet's processor has to be loaded from a URL, which on this SDK\n * means a blob: URL — and a blob: worklet is refused outright by the\n * Content-Security-Policy of a great many of the sites this code runs on. A\n * deprecated node that works on a customer's page beats a modern one that is\n * blocked by their headers.\n *\n * Returns a `stop` that lets go of the whole chain. The microphone track is\n * deliberately *not* stopped by it: on a live call the same track is carrying\n * the conversation, and stopping it to end a recording would end the call.\n *\n * @param {AudioContext} context\n * @param {MediaStream} stream\n * @param {(bytes: Uint8Array) => void} onFrame\n * @param {number} [wireRate]\n * @returns {{stop: () => void}}\n */\nexport function tapMicrophone(context, stream, onFrame, wireRate = WIRE_RATE) {\n  const source = context.createMediaStreamSource(stream);\n  const capture = context.createScriptProcessor(CAPTURE_BLOCK, 1, 1);\n\n  capture.onaudioprocess = (event) => {\n    const input = event.inputBuffer.getChannelData(0);\n    const frames = context.sampleRate === wireRate\n      ? input\n      : downsample(input, context.sampleRate, wireRate);\n    onFrame(encodeMuLaw(frames));\n  };\n\n  source.connect(capture);\n  // A ScriptProcessorNode does not run unless it is connected to a destination,\n  // even when nothing is meant to come out of it. Zero gain is what keeps the\n  // speaker from hearing their own voice played back at them.\n  const mute = context.createGain();\n  mute.gain.value = 0;\n  capture.connect(mute);\n  mute.connect(context.destination);\n\n  return {\n    stop() {\n      capture.onaudioprocess = null;\n      try { source.disconnect(); } catch { /* already gone */ }\n      try { capture.disconnect(); } catch { /* already gone */ }\n      try { mute.disconnect(); } catch { /* already gone */ }\n    },\n  };\n}\n","// Two people, talking directly.\n//\n// The agent call next door is a microphone, a socket and a codec: the browser\n// sends µ-law up to sphoro.voice, a model answers, and every sample passes\n// through a server. This is the opposite arrangement. The audio here goes\n// straight from one browser to the other over WebRTC, and sphoro.voice's part is\n// only to introduce them — it passes an offer, an answer and a handful of ICE\n// candidates between the two sockets and then has nothing left to do.\n//\n// Which means there is no codec in this file, no resampler, and no audio path\n// at all. The browser's own WebRTC stack owns those, and it is better at them\n// than anything hand-written: Opus at whatever rate the two ends agree on,\n// echo cancellation, jitter buffering and packet loss concealment, none of\n// which the µ-law path can have because a telephone carrier cannot receive\n// them. A live call sounds better than the agent call, and costs nothing per\n// minute, because nobody is in the middle of it.\n//\n// # Shared with the dashboard\n//\n// Both ends of a live call run this file: the reporter's widget and the member\n// who answers in the portal, which imports it from here rather than keeping a\n// second copy. The two sides differ by one flag — who makes the offer — and it\n// arrives from the server rather than being decided here, so the two ends\n// cannot both sit waiting for the other to go first.\n//\n// # The one exception: the copy that gets kept\n//\n// An agent that is set to record or transcribe cannot do either from here —\n// the audio it would need never reaches it. So sphoro.voice asks instead: it\n// names, in `session.created`, what this call is keeping and in what format,\n// and each browser uplinks a copy of *its own* microphone over the signaling\n// socket it already has open. That copy is the only way a recording or a\n// transcript of a peer-to-peer call can exist at all, and it is why the codec\n// this file said it did not have is imported below.\n//\n// It is a copy and not the call. The conversation still runs browser to\n// browser; an uplink that stalls or is refused costs a recording, never a call.\n//\n// Nothing outside the package is imported to do any of it. The SDK runs on\n// someone else's page and has no dependencies, and that rule does not bend for\n// a peer connection.\n\nimport { createContext, resumeContext, tapMicrophone } from './audio.js';\n\n/**\n * sphoro.voice's realtime protocol, sent as the WebSocket subprotocol. The same\n * one the agent call speaks — a bridged call is a different conversation over\n * the same envelope, not a different protocol.\n */\nconst SUBPROTOCOL = 'voiceai.realtime.v1';\n\n/**\n * Where a browser looks to find its way to the other one, when the server names\n * nowhere.\n *\n * The server almost always does name somewhere — it is configured per\n * deployment, because a network behind symmetric NAT needs a TURN relay and\n * that costs money. This is the floor: a public STUN server, which is enough\n * for most home and office networks and free.\n */\nconst FALLBACK_ICE = [{ urls: 'stun:stun.l.google.com:19302' }];\n\n/**\n * Ask for the microphone.\n *\n * Separate from joining, and deliberately. The permission prompt takes as long\n * as the person takes to read it, and both ends have something they should not\n * do until it is answered: the reporter should not make a room full of\n * dashboards ring before knowing they can speak, and whoever answers should not\n * claim a call they then cannot take. So both ask first and act second.\n *\n * @returns {Promise<MediaStream>}\n */\nexport function openMicrophone() {\n  return navigator.mediaDevices.getUserMedia({\n    audio: {\n      // Both ends are on speakers as often as not. Without cancellation each\n      // hears themselves back through the other, half a second late, which is\n      // the single most effective way to make a conversation impossible.\n      echoCancellation: true,\n      noiseSuppression: true,\n      autoGainControl: true,\n      channelCount: 1,\n    },\n  });\n}\n\n/** Whether this browser can be on a live call at all. */\nexport function liveCallSupported() {\n  return typeof window !== 'undefined'\n    && typeof window.RTCPeerConnection === 'function'\n    && Boolean(navigator.mediaDevices?.getUserMedia);\n}\n\n/**\n * Join a live call and stay on it until somebody hangs up.\n *\n * @param {object} opts\n * @param {string} opts.realtimeUrl  From the API, carrying its own credential\n *   and its own seat. Never log it and never build one by hand.\n * @param {MediaStream} [opts.stream]  A microphone already opened by\n *   `openMicrophone`. One is opened here if not, which is the wrong order for\n *   anything that had a decision to make first.\n * @param {string} [opts.ringsUntil]  ISO-8601, on the caller's side only: when\n *   the ringing stops. The browser is the thing that decides it has, because it\n *   is the one client guaranteed to still be listening at that moment.\n * @param {(state: string, detail?: object) => void} [opts.onState]\n *   'ringing' | 'connecting' | 'connected' | 'reconnecting' | 'ended' | 'missed'\n *   | 'failed'. 'reconnecting' is not a failure and never ends a call — see\n *   `routeChanged`.\n * @param {(blocked: boolean) => void} [opts.onAudioBlocked]  The browser refused\n *   to play the other person, or has since started. Its own callback rather\n *   than a state, because it is orthogonal to every one of them: the call is\n *   connected and working, and one end cannot hear it. See `playRemote`.\n * @param {(live: boolean) => void} [opts.onMicrophone]  Whether this person's\n *   microphone is still reaching the call. Also orthogonal, and for the same\n *   reason: the call is up, the other end is audible, and nothing this person\n *   says is going anywhere. See `watchMicrophone`.\n * @returns {Promise<{hangUp: () => void, setMuted: (muted: boolean) => boolean,\n *   isMuted: () => boolean}>}\n */\nexport async function joinLiveCall({\n  realtimeUrl, stream, ringsUntil, onState, onAudioBlocked, onMicrophone,\n}) {\n  const emit = (state, detail) => { try { onState?.(state, detail); } catch { /* theirs */ } };\n  const sayBlocked = (blocked) => {\n    try { onAudioBlocked?.(blocked); } catch { /* theirs */ }\n  };\n  const sayMicrophone = (live) => {\n    try { onMicrophone?.(live); } catch { /* theirs */ }\n  };\n\n  if (!liveCallSupported()) {\n    const err = new Error('[BugCatch] this browser cannot make live calls.');\n    emit('failed', { reason: 'unsupported' });\n    throw err;\n  }\n\n  let microphone = stream;\n  if (!microphone) {\n    try {\n      microphone = await openMicrophone();\n    } catch (err) {\n      emit('failed', { reason: 'microphone', error: String(err?.name || err) });\n      throw err;\n    }\n  }\n\n  let socket;\n  try {\n    socket = await openSocket(realtimeUrl);\n  } catch (err) {\n    stopStream(microphone);\n    emit('failed', { reason: 'connect' });\n    throw err;\n  }\n\n  let peer = null;\n  let remoteAudio = null;\n  let finished = false;\n  let ringTimer = null;\n  // What sphoro.voice said this call is keeping, and the tap feeding it. Both\n  // stay null on the ordinary call, which keeps nothing and uplinks nothing.\n  let capture = null;\n  let uplink = null;\n  // Whether the browser is currently refusing to play the other person, and the\n  // teardown for the gesture listeners armed to try again. See playRemote.\n  let blocked = false;\n  let disarmGesture = null;\n  /** Teardown for the microphone watch, armed once the peer connection is built. */\n  let unwatchMicrophone = null;\n  // Claimed synchronously, before the first await below, because the seat that\n  // answers is told the call has started twice — once in its own\n  // `session.created`, which says somebody is already waiting, and again in the\n  // `peer.joined` that goes to both ends a moment later. Two starts that both\n  // check `uplink` before either has set it are two taps on one microphone, and\n  // two copies of every frame interleaved into one recording.\n  let starting = false;\n  // Candidates that arrived before there was anywhere to put them.\n  //\n  // The offer and the first candidates are sent within a millisecond of each\n  // other and arrive in that order, but `setRemoteDescription` is a promise —\n  // so a candidate can be in hand while the description it belongs to is still\n  // being applied. Adding one then throws, and the call comes up with a worse\n  // route than it could have had, or none.\n  const pending = [];\n\n  // --- surviving a change of network ------------------------------------------------\n  //\n  // A phone call moves. Somebody walks out of the office and the handset hands the\n  // connection from Wi-Fi to cellular; a lift takes the signal for four seconds; a laptop\n  // wakes on a different access point. Every one of those changes the local address the\n  // call was negotiated against, and WebRTC's answer to all of them is the same: the ICE\n  // agent reports `disconnected`, and unless somebody restarts it, `failed` a moment later.\n  //\n  // Before this, `failed` ended the call and `disconnected` was not handled at all — so a\n  // call that would have recovered on its own in two seconds was hung up on, and one that\n  // needed a restart was hung up on with no attempt made. On a desk that is rare. On the\n  // phone this app also ships as, it is what happens when you stand up.\n  //\n  // What replaces it is a grace period, then an ICE restart, then a deadline. Each is\n  // there for a distinct reason:\n  //\n  //   the grace   `disconnected` is frequently transient — a couple of lost packets on a\n  //               route that is still fine. Restarting immediately would tear down a\n  //               working connection to rebuild the same one, and take audio with it.\n  //\n  //   the restart Renegotiation from scratch over the signaling socket, which is still\n  //               open: it never depended on the media route. This is the part that\n  //               actually crosses a network change.\n  //\n  //   the deadline Reconnecting forever is its own failure. Somebody staring at a card\n  //               that says \"reconnecting\" for five minutes has been abandoned more\n  //               thoroughly than somebody told the call dropped.\n  //\n  // Only one side offers, and it is the side the server nominated at the start. Both ends\n  // see the drop, so if the two sides both restarted there would be two offers crossing —\n  // a glare condition WebRTC resolves badly. The other side asks instead, over the\n  // signaling socket, and its request is best-effort: the offerer's own detection is what\n  // this actually rests on.\n\n  /** Which side was told to make the offer. Set from `peer.joined`, for both ends. */\n  let shouldOffer = false;\n  /** Whether the two ends have ever been connected. A drop before that is a failed setup. */\n  let connectedOnce = false;\n  let reconnecting = false;\n  let recoverTimer = null;\n  let giveUpTimer = null;\n  let restarts = 0;\n\n  /** How long `disconnected` is allowed to resolve itself before a restart is attempted. */\n  const RECOVER_GRACE_MS = 2000;\n  /** How long between one restart and the next, when the first did not take. */\n  const RECOVER_RETRY_MS = 5000;\n  /** How long a call may spend off the air in total before it is called dropped. */\n  const RECOVER_DEADLINE_MS = 30000;\n  const MAX_RESTARTS = 4;\n\n  /** Everything ends here, whatever ended it, exactly once. */\n  const finish = (state, detail) => {\n    if (finished) return;\n    finished = true;\n    if (ringTimer) { clearTimeout(ringTimer); ringTimer = null; }\n    // Before the peer is closed, and before the stream is stopped. Closing a connection\n    // moves it to `closed` and stopping a track fires `ended`, and both would otherwise\n    // reach handlers that would report a dropped call and a dead microphone to somebody\n    // who has just hung up.\n    clearRecovery();\n    unwatchMicrophone?.();\n    unwatchMicrophone = null;\n    stopUplink();\n    // Before anything else is torn down: these are listeners on `window`, and a\n    // call that left them behind would keep retrying playback on an element\n    // that is about to be removed — once per click, for the rest of the session.\n    disarmGesture?.();\n    disarmGesture = null;\n    try { peer?.close(); } catch { /* already gone */ }\n    stopStream(microphone);\n    if (remoteAudio) {\n      remoteAudio.srcObject = null;\n      remoteAudio.remove();\n      remoteAudio = null;\n    }\n    try {\n      if (socket.readyState === WebSocket.OPEN) socket.close(1000, 'call over');\n    } catch { /* already closing */ }\n    emit(state, detail);\n  };\n\n  const send = (message) => {\n    if (socket.readyState === WebSocket.OPEN) socket.send(JSON.stringify(message));\n  };\n\n  /**\n   * Start sending sphoro.voice a copy of this person's microphone.\n   *\n   * Only when it asked, and only once both people are on the call. The server\n   * throws away anything that arrives while the phone is still ringing — a\n   * recording is timed from the moment the second person joined, and a minute\n   * of somebody waiting alone at the head of the file is a timeline that\n   * disagrees with the call's own duration. Sending it anyway would just be\n   * bandwidth spent on audio that gets dropped, on the same network carrying\n   * the call.\n   *\n   * Every failure here is survivable and none of them ends the call. A browser\n   * with no Web Audio, a context the autoplay policy will not start, a format\n   * this version does not know how to produce — each costs the recording and\n   * leaves two people talking, which is the right way round. It is said out\n   * loud, though: an agent set to record whose calls silently arrive with no\n   * audio is the failure that is impossible to diagnose from the outside.\n   */\n  const startUplink = async () => {\n    if (starting || uplink || finished || !capture) return;\n    if (!capture.audio && !capture.transcript) return;\n    starting = true;\n\n    // The server names the format rather than assuming one, so a client that\n    // does not recognise it sends nothing rather than sending µ-law into a\n    // recording that expected something else — which would store a file of\n    // noise and read as a working recording to everything downstream.\n    const encoding = capture.encoding || 'mulaw';\n    if (encoding !== 'mulaw') {\n      warn(`this call is kept as ${encoding}, which this SDK cannot produce.`);\n      return;\n    }\n\n\n    const rate = Number(capture.sample_rate) || 8000;\n    try {\n      const context = createContext(rate);\n      // A context built without a user gesture starts suspended, and a\n      // suspended context's ScriptProcessorNode never fires — so nothing would\n      // be uplinked and nothing would say why. Both ends granted microphone\n      // access a moment ago, which is the activation this spends.\n      await resumeContext(context);\n      if (finished) {\n        try { context.close(); } catch { /* already closed */ }\n        return;\n      }\n\n      const tap = tapMicrophone(context, microphone, (bytes) => {\n        // Binary, not the base64 envelope the protocol also accepts: this is a\n        // second copy of audio going up a network that is already carrying the\n        // call, and base64 would make it a third larger for nothing.\n        if (socket.readyState === WebSocket.OPEN) socket.send(bytes);\n      }, rate);\n\n      uplink = { context, tap };\n    } catch (err) {\n      warn(`could not send the copy this call is keeping: ${String(err?.message || err)}`);\n    }\n  };\n\n  /**\n   * Stop sending it.\n   *\n   * The microphone track is deliberately left alone — `finish` stops the whole\n   * stream a moment later, and stopping it here would cut the conversation off\n   * to end a recording.\n   */\n  function stopUplink() {\n    starting = false;\n    if (!uplink) return;\n    const { context, tap } = uplink;\n    uplink = null;\n    try { tap.stop(); } catch { /* already gone */ }\n    try { context.close(); } catch { /* already closed */ }\n  }\n\n  /**\n   * Play the other person, and do something about it when the browser says no.\n   *\n   * The refusal used to be swallowed: `play().catch(() => {})`, on the reasoning\n   * that both ends had granted microphone access a moment earlier and that\n   * satisfies every autoplay policy. It does not always. A call answered from a\n   * notification arrives with no user gesture on the page at all — the tap\n   * happened on a system notification, which is not this document — and inside\n   * an Android WebView that is exactly the case that fails.\n   *\n   * What that produced is the worst shape a bug can have: the call connects,\n   * every state says `connected`, both ends are billed for it, and one person\n   * hears nothing. Nothing is logged, because the rejection was caught and\n   * dropped, so there is no evidence anywhere that it happened.\n   *\n   * So now it is reported and it is retried. The retry is armed on the next\n   * gesture anywhere on the page, which is the one thing that reliably lifts an\n   * autoplay block, and it is armed in capture so a handler calling\n   * `stopPropagation` cannot hide the gesture from it. Whoever is on the call is\n   * about to touch something — the mute button, the hang-up button, the report\n   * they are talking about — and the sound comes up when they do, with the\n   * caller free to say so on screen in the meantime.\n   */\n  function playRemote() {\n    if (!remoteAudio || finished) return;\n    let attempt;\n    try {\n      attempt = remoteAudio.play?.();\n    } catch {\n      // A synchronous throw, which some engines do instead of rejecting.\n      markBlocked();\n      return;\n    }\n    if (!attempt || typeof attempt.then !== 'function') {\n      // No promise: an older engine, or a stub. Nothing to wait on and nothing\n      // to report — it either made a sound or it did not.\n      markPlaying();\n      return;\n    }\n    attempt.then(markPlaying, markBlocked);\n  }\n\n  function markPlaying() {\n    disarmGesture?.();\n    disarmGesture = null;\n    if (!blocked) return;\n    blocked = false;\n    sayBlocked(false);\n  }\n\n  function markBlocked() {\n    if (finished) return;\n    if (!blocked) {\n      blocked = true;\n      sayBlocked(true);\n    }\n    armGesture();\n  }\n\n  /** Try again on the next thing the person touches. Armed once. */\n  function armGesture() {\n    if (disarmGesture) return;\n    // Guarded rather than assumed: this file is also loaded by a host page's\n    // bundler into environments that define a partial `window`, and a retry\n    // that cannot be armed is a quieter failure than one that throws inside a\n    // promise nobody is awaiting.\n    if (typeof window === 'undefined' || typeof window.addEventListener !== 'function') return;\n    const retry = () => playRemote();\n    const events = ['pointerdown', 'keydown', 'touchstart'];\n    for (const name of events) {\n      window.addEventListener(name, retry, { capture: true, passive: true });\n    }\n    disarmGesture = () => {\n      for (const name of events) {\n        try { window.removeEventListener(name, retry, { capture: true }); } catch { /* gone */ }\n      }\n    };\n  }\n\n  /**\n   * Build the peer connection.\n   *\n   * Made on `session.created` rather than lazily on the first signal, because\n   * the ICE gathering it starts is the slowest part of connecting and the whole\n   * of it can happen while the phone is still ringing. Waiting until somebody\n   * answers would add that delay to the exact moment a person says hello, which\n   * is the one moment on this call where it is audible.\n   */\n  const build = (iceServers) => {\n    peer = new RTCPeerConnection({ iceServers: iceServers?.length ? iceServers : FALLBACK_ICE });\n\n    for (const track of microphone.getAudioTracks()) peer.addTrack(track, microphone);\n\n    // Armed here rather than at the top, so a call that never got a peer connection has\n    // nothing to tear down — and so the watch covers exactly the span in which being\n    // unheard is a problem worth telling somebody about.\n    unwatchMicrophone = watchMicrophone();\n\n    peer.onicecandidate = (event) => {\n      if (!event.candidate) return;\n      send({ type: 'signal', signal: { kind: 'candidate', payload: event.candidate.toJSON() } });\n    };\n\n    peer.ontrack = (event) => {\n      const [incoming] = event.streams;\n      if (!incoming) return;\n      // An element rather than the Web Audio API. Audio routed through a\n      // MediaStreamAudioSourceNode loses the browser's own echo cancellation on\n      // several engines, and this call has a live microphone next to it.\n      if (!remoteAudio) {\n        remoteAudio = document.createElement('audio');\n        remoteAudio.autoplay = true;\n        // Inline, for a WebView that would otherwise hand a media element to a\n        // full-screen player — which on a call is a black rectangle over the\n        // dashboard.\n        remoteAudio.playsInline = true;\n        remoteAudio.setAttribute?.('playsinline', '');\n        // Off-screen rather than display:none — a hidden media element is\n        // treated as invisible by some autoplay heuristics and refused.\n        remoteAudio.style.cssText = 'position:fixed;width:0;height:0;opacity:0;pointer-events:none';\n        document.body.appendChild(remoteAudio);\n      }\n      remoteAudio.srcObject = incoming;\n      playRemote();\n    };\n\n    peer.onconnectionstatechange = routeChanged;\n    // The same signal, read the other way, for the engines that do not give a reliable\n    // `connectionState`. WebKit — which is every browser on iOS and the WebView this app\n    // ships inside there — has historically fired `iceconnectionstatechange` and not the\n    // aggregate one. Both are wired to the same handler, which reads whichever of the two\n    // the engine actually maintains, so a duplicate event costs one comparison.\n    peer.oniceconnectionstatechange = routeChanged;\n  };\n\n  /**\n   * What the media route is doing, and what to do about it.\n   *\n   * Reads whichever state the engine keeps up to date, and treats `completed` — which only\n   * the ICE-level state has — as connected, because it is: it means a route was found and\n   * checking has stopped.\n   */\n  function routeChanged() {\n    if (!peer || finished) return;\n    const state = peer.connectionState || peer.iceConnectionState;\n\n    if (state === 'connected' || state === 'completed') {\n      connectedOnce = true;\n      restarts = 0;\n      clearRecovery();\n      // Emitted whether or not this is a reconnection. The caller renders one state, and a\n      // card that came back from \"reconnecting\" has to be told the call is up again.\n      reconnecting = false;\n      emit('connected');\n      return;\n    }\n\n    if (state === 'disconnected') {\n      // Before the two ends ever met, a `disconnected` is ICE still working — it goes\n      // there between checking one candidate pair and the next. Only `failed` is final at\n      // that stage, and it is handled below.\n      if (!connectedOnce) return;\n      enterRecovery(RECOVER_GRACE_MS);\n      return;\n    }\n\n    if (state === 'failed') {\n      if (!connectedOnce) {\n        // The two browsers never found a route to each other at all. Almost always a\n        // network that needs a TURN relay the deployment has not configured; reported as\n        // its own reason so that is diagnosable from a support ticket rather than from a\n        // packet capture. A restart would try the same candidates again.\n        finish('failed', { reason: 'ice' });\n        return;\n      }\n      // A route that worked and then stopped. That is a network change, and a restart is\n      // exactly the remedy — no grace, because `failed` is not going to resolve itself.\n      enterRecovery(0);\n    }\n  }\n\n  /** Say the call is off the air, and line up an attempt to bring it back. */\n  function enterRecovery(delay) {\n    if (finished) return;\n    if (!reconnecting) {\n      reconnecting = true;\n      emit('reconnecting');\n      giveUpTimer = setTimeout(() => finish('failed', { reason: 'network' }), RECOVER_DEADLINE_MS);\n    }\n    // One attempt in flight at a time. `disconnected` and `failed` both arrive on the way\n    // down, and two timers would mean two offers crossing on one connection.\n    if (recoverTimer) return;\n    recoverTimer = setTimeout(recover, delay);\n  }\n\n  function clearRecovery() {\n    if (recoverTimer) { clearTimeout(recoverTimer); recoverTimer = null; }\n    if (giveUpTimer) { clearTimeout(giveUpTimer); giveUpTimer = null; }\n  }\n\n  /**\n   * Renegotiate the media route over the signaling socket, which never left.\n   *\n   * `createOffer({ iceRestart: true })` rather than `restartIce()`: the latter is the\n   * modern spelling and does the same thing by firing `negotiationneeded`, but it is\n   * missing from engines this SDK is expected to run in and the explicit form works\n   * everywhere. It also keeps the offer on the same path as the original one, so there is a\n   * single place where an offer is made.\n   */\n  async function recover() {\n    recoverTimer = null;\n    if (finished || !peer) return;\n    // It came back on its own while the grace period was running, which is the common case\n    // and the reason the grace period exists.\n    const state = peer.connectionState || peer.iceConnectionState;\n    if (state === 'connected' || state === 'completed') return;\n\n    if (!shouldOffer) {\n      // Not this side's to restart. Ask, and keep asking — the offerer sees the same drop\n      // and is very likely already doing it, so this is the belt to that braces.\n      send({ type: 'signal', signal: { kind: 'restart', payload: {} } });\n      recoverTimer = setTimeout(recover, RECOVER_RETRY_MS);\n      return;\n    }\n\n    if (restarts >= MAX_RESTARTS) return; // the deadline is what ends it\n    restarts += 1;\n    await restartIce();\n    recoverTimer = setTimeout(recover, RECOVER_RETRY_MS);\n  }\n\n  /** The restart itself, also reachable from the other side's request. */\n  async function restartIce() {\n    if (finished || !peer) return;\n    try {\n      const description = await peer.createOffer({ iceRestart: true });\n      await peer.setLocalDescription(description);\n      send({ type: 'signal', signal: { kind: 'offer', payload: peer.localDescription.toJSON() } });\n    } catch {\n      // Nothing to do here but let the next attempt or the deadline decide. Ending the\n      // call on one refused renegotiation would throw away the retries that follow.\n    }\n  }\n\n  /**\n   * Whether this person is still being heard.\n   *\n   * A track's `muted` is not the mute button — that is `enabled`, which this file sets and\n   * therefore already knows about. `muted` is the *source* stopping: the OS handed the\n   * microphone to an incoming phone call, a headset was unplugged, or — the case this app\n   * meets daily — Android revoked microphone access because the app went to the background.\n   *\n   * Which produces the same shape as every other bug on this path: the call is up, the\n   * other person is audible, both ends are billed, and nothing this person says goes\n   * anywhere. They find out when the other person says \"hello? are you there?\".\n   *\n   * The loss is reported on a short delay and the recovery immediately. Several engines\n   * flick `muted` on for a few frames while a track settles, and a card that said \"your\n   * microphone was cut off\" for 200ms at the start of every call would be noise — while a\n   * microphone that came back and was still reported lost is the error saying the wrong\n   * thing.\n   */\n  function watchMicrophone() {\n    const SETTLE_MS = 600;\n    let settle = null;\n    let lost = false;\n\n    const say = (live) => {\n      if (settle) { clearTimeout(settle); settle = null; }\n      if (live) {\n        if (!lost) return;\n        lost = false;\n        sayMicrophone(true);\n        return;\n      }\n      if (lost) return;\n      settle = setTimeout(() => {\n        settle = null;\n        if (finished) return;\n        lost = true;\n        sayMicrophone(false);\n      }, SETTLE_MS);\n    };\n\n    for (const track of microphone.getAudioTracks()) {\n      track.onmute = () => say(false);\n      track.onunmute = () => say(true);\n      // An ended track never comes back — the device is gone. Said at once rather than\n      // after the settle, because there is nothing to wait for.\n      track.onended = () => {\n        if (settle) { clearTimeout(settle); settle = null; }\n        if (lost) return;\n        lost = true;\n        sayMicrophone(false);\n      };\n    }\n    return () => {\n      if (settle) { clearTimeout(settle); settle = null; }\n      for (const track of microphone.getAudioTracks()) {\n        track.onmute = null;\n        track.onunmute = null;\n        track.onended = null;\n      }\n    };\n  }\n\n  const offer = async () => {\n    try {\n      const description = await peer.createOffer();\n      await peer.setLocalDescription(description);\n      send({ type: 'signal', signal: { kind: 'offer', payload: peer.localDescription.toJSON() } });\n    } catch {\n      finish('failed', { reason: 'negotiation' });\n    }\n  };\n\n  const accept = async (payload) => {\n    try {\n      await peer.setRemoteDescription(new RTCSessionDescription(payload));\n      const answer = await peer.createAnswer();\n      await peer.setLocalDescription(answer);\n      send({ type: 'signal', signal: { kind: 'answer', payload: peer.localDescription.toJSON() } });\n      await drainCandidates();\n    } catch {\n      finish('failed', { reason: 'negotiation' });\n    }\n  };\n\n  const drainCandidates = async () => {\n    while (pending.length) {\n      const candidate = pending.shift();\n      try {\n        await peer.addIceCandidate(new RTCIceCandidate(candidate));\n      } catch {\n        // One unusable candidate is not a failed call — there are usually\n        // several, and the connection needs only one of them to work.\n      }\n    }\n  };\n\n  /**\n   * One message at a time, in the order they arrived.\n   *\n   * Handling is asynchronous — `setRemoteDescription` and `createAnswer` are\n   * promises — and the socket does not wait. Without this chain a candidate\n   * arriving mid-answer is examined while the description it belongs to is\n   * still being applied, so it goes on the pending list a moment after the\n   * pending list was drained, and stays there for the rest of the call.\n   */\n  let handling = Promise.resolve();\n  socket.onmessage = (event) => {\n    handling = handling.then(() => handle(event)).catch(() => { /* one bad frame */ });\n  };\n\n  const handle = async (event) => {\n    if (typeof event.data !== 'string') return;\n    let message;\n    try { message = JSON.parse(event.data); } catch { return; }\n\n    switch (message.type) {\n      case 'session.created': {\n        build(message.data?.ice_servers);\n        // What this call keeps, and in what format. Held rather than acted on:\n        // nothing is uplinked until somebody is actually on the other end.\n        capture = message.data?.capture || null;\n        // Ringing, until somebody is on the other side. Whoever answers finds\n        // the caller already waiting and skips straight past it — the\n        // `peer.joined` that follows a moment later says the same thing, and\n        // this is what keeps the answerer's first frame from being silence.\n        if (message.data?.peer_present) {\n          emit('connecting');\n          void startUplink();\n          break;\n        }\n        emit('ringing');\n        armRingTimer();\n        break;\n      }\n\n      case 'peer.joined': {\n        if (ringTimer) { clearTimeout(ringTimer); ringTimer = null; }\n        emit('connecting');\n        // Told, not inferred. Both ends learn somebody arrived in the same\n        // frame, and only one of them is asked to go first.\n        //\n        // Remembered as well as acted on: it is also the answer to \"whose job is it to\n        // renegotiate when the network changes\", and that question is asked later, when\n        // the frame that answered it is long gone. See `recover`.\n        shouldOffer = Boolean(message.data?.should_offer);\n        if (shouldOffer) await offer();\n        // After the offer, and not awaited: opening an audio context is\n        // milliseconds, but they would be milliseconds added to the moment\n        // somebody says hello, which is the one place on this call where they\n        // are audible.\n        void startUplink();\n        break;\n      }\n\n      case 'signal': {\n        if (!peer) break;\n        const { kind, payload } = message.data || {};\n        if (kind === 'offer') {\n          await accept(payload);\n        } else if (kind === 'answer') {\n          try {\n            await peer.setRemoteDescription(new RTCSessionDescription(payload));\n            await drainCandidates();\n          } catch {\n            finish('failed', { reason: 'negotiation' });\n          }\n        } else if (kind === 'candidate') {\n          if (peer.remoteDescription) {\n            try { await peer.addIceCandidate(new RTCIceCandidate(payload)); } catch { /* see above */ }\n          } else {\n            pending.push(payload);\n          }\n        } else if (kind === 'restart') {\n          // The other side lost the route and cannot renegotiate itself. Honoured only by\n          // the side that was nominated to offer, so a request arriving at both ends of a\n          // mis-signalled call cannot produce two offers crossing.\n          if (shouldOffer) await restartIce();\n        }\n        break;\n      }\n\n      case 'peer.left':\n        finish('ended', { reason: 'peer_left' });\n        break;\n\n      case 'error':\n        finish('failed', { reason: 'server', detail: message.error?.detail });\n        break;\n\n      case 'session.ended':\n        finish('ended', { reason: 'server' });\n        break;\n\n      default:\n        // Every message the agent conversation has — audio deltas, transcripts,\n        // state changes. None of them happen on a bridged call, and one\n        // arriving is not worth ending a working call over.\n        break;\n    }\n  };\n\n  socket.onclose = () => finish('ended', { reason: 'disconnected' });\n  socket.onerror = () => finish('failed', { reason: 'socket' });\n\n  /**\n   * Stop waiting.\n   *\n   * The deadline comes from the server, which wrote it down when the call was\n   * created — so a browser with a slow clock waits a little longer rather than\n   * hanging up on somebody mid-answer, and the server refuses a give-up that\n   * arrives early in any case.\n   */\n  function armRingTimer() {\n    if (!ringsUntil) return;\n    const remaining = new Date(ringsUntil).getTime() - Date.now();\n    if (!Number.isFinite(remaining)) return;\n    ringTimer = setTimeout(() => finish('missed'), Math.max(0, remaining));\n  }\n\n  return {\n    hangUp() {\n      // Said out loud before the socket closes. A close on its own would end\n      // the call too — sphoro.voice ends a bridge the moment one side leaves —\n      // but saying so first is what distinguishes hanging up from a tunnel.\n      send({ type: 'session.end' });\n      finish('ended', { reason: 'hung_up' });\n    },\n    setMuted(muted) {\n      for (const track of microphone.getAudioTracks()) track.enabled = !muted;\n      return muted;\n    },\n    isMuted() {\n      return microphone.getAudioTracks().every((track) => !track.enabled);\n    },\n  };\n}\n\n/** Open the socket, resolving only once it is actually open. */\nfunction openSocket(url) {\n  return new Promise((resolve, reject) => {\n    let socket;\n    try {\n      socket = new WebSocket(url, SUBPROTOCOL);\n    } catch (err) {\n      reject(err);\n      return;\n    }\n    socket.onopen = () => {\n      socket.onopen = null;\n      socket.onerror = null;\n      resolve(socket);\n    };\n    socket.onerror = () => {\n      socket.onopen = null;\n      socket.onerror = null;\n      reject(new Error('[BugCatch] could not connect the call.'));\n    };\n  });\n}\n\n/**\n * Say something went wrong with the copy, without ending anything.\n *\n * `console.warn` and not a state: every caller of this file renders the states\n * to somebody who is mid-conversation, and \"the recording did not start\" is not\n * news to the two people talking. It is news to whoever turned recording on,\n * and the console is where they will look.\n */\nfunction warn(message) {\n  try { console.warn(`[BugCatch] ${message}`); } catch { /* no console */ }\n}\n\nfunction stopStream(stream) {\n  for (const track of stream?.getTracks?.() ?? []) {\n    try { track.stop(); } catch { /* already stopped */ }\n  }\n}\n","// The reporter's telephone.\n//\n// Everything else this SDK draws is a consequence of something the person on\n// the page did: they pressed the widget, they filed a report, they asked to\n// talk it through. This is the one thing that arrives unasked — a member of the\n// team has a question about a report and is ringing to ask it — and that makes\n// it the piece with the most rules around it.\n//\n// # What it is allowed to do\n//\n// Appear, and make a noise. That is the whole of it. No microphone is opened,\n// no audio moves and nothing is connected until the person presses Answer, and\n// if they press nothing at all it takes itself away when the ringing stops. The\n// card is an offer; the SDK does not put a customer on a call because somebody\n// at the vendor clicked a button.\n//\n// # And what it says\n//\n// A name, a product and what the call is about, because a card that says\n// \"someone is calling about a report\" is indistinguishable from the thing every\n// bank tells people to hang up on. And whether the call would be recorded,\n// stated before Answer rather than after — on a peer-to-peer call the copy that\n// gets kept is uplinked by this browser, so telling somebody afterwards is\n// telling them about something that already happened.\n//\n// # The noise\n//\n// Synthesised, for the reason the dashboard's is: a ringtone as an audio file\n// is a binary in a bundle that runs on somebody else's site, an extra request\n// on their page, and a thing that can go missing behind a cache header. Four\n// oscillators and a table of numbers cost none of that.\n//\n// It is also softer and shorter than the dashboard's. A phone on a support\n// desk is meant to interrupt; this one is ringing in the middle of a stranger's\n// shopping cart, and a page that suddenly plays a loud telephone at somebody is\n// a page they close. If the browser refuses to play it at all — no interaction\n// yet, which is its rule and not ours — the card rings silently rather than not\n// appearing.\n\n/**\n * A quiet double-pulse, once every three seconds.\n *\n * Two notes rather than one so it reads as a telephone rather than as an alert,\n * and a fifth apart because that interval is the one people hear as \"ringing\"\n * across every phone system anybody has used.\n */\nconst RING = {\n  cycleMs: 3000,\n  peak: 0.045,\n  bursts: [\n    { at: 0, seconds: 0.4, hz: [440, 660] },\n    { at: 0.55, seconds: 0.4, hz: [440, 660] },\n  ],\n};\n\n/**\n * The same question asked of the browser rather than of ourselves.\n *\n * Chrome and Edge implement this and are the browsers that enforce the policy\n * in the first place, so where it matters it answers. It is the recovery for a\n * host page that swallowed every gesture before we saw it — better to make a\n * quiet noise the person expects than to stay silent because of somebody else's\n * event handler.\n */\nfunction activatedByBrowser() {\n  try {\n    return Boolean(navigator.userActivation?.hasBeenActive);\n  } catch {\n    return false;\n  }\n}\n\n/**\n * Start ringing, and give back the way to stop.\n *\n * Returns a no-op teardown on every browser that will not do it, which is a\n * real and ordinary case: a page nobody has clicked on yet has no audio\n * permission, and the correct behaviour there is a card that appears in silence\n * rather than no card.\n */\nfunction startRinging(touched) {\n  if (!touched() && !activatedByBrowser()) return () => {};\n\n  const Ctx = typeof window !== 'undefined' && (window.AudioContext || window.webkitAudioContext);\n  if (!Ctx) return () => {};\n\n  let context;\n  try {\n    context = new Ctx();\n  } catch {\n    return () => {};\n  }\n\n  let stopped = false;\n  let timer = null;\n  const voices = [];\n\n  const burst = ({ at, seconds, hz }) => {\n    const start = context.currentTime + at;\n    const gain = context.createGain();\n    gain.connect(context.destination);\n    // Shaped rather than switched. A square edge on a sine is a click, and a\n    // click every three seconds is more irritating than the tone it introduces.\n    gain.gain.setValueAtTime(0, start);\n    gain.gain.linearRampToValueAtTime(RING.peak, start + 0.02);\n    gain.gain.setValueAtTime(RING.peak, start + seconds - 0.06);\n    gain.gain.linearRampToValueAtTime(0, start + seconds);\n\n    for (const frequency of hz) {\n      const osc = context.createOscillator();\n      osc.type = 'sine';\n      osc.frequency.value = frequency;\n      osc.connect(gain);\n      osc.start(start);\n      osc.stop(start + seconds + 0.02);\n      voices.push(osc);\n    }\n  };\n\n  const cycle = () => {\n    if (stopped) return;\n    try {\n      RING.bursts.forEach(burst);\n    } catch { /* the card is still on screen, which is the part that matters */ }\n  };\n\n  const begin = () => {\n    if (stopped) return;\n    cycle();\n    timer = setInterval(cycle, RING.cycleMs);\n  };\n\n  if (context.state === 'running') begin();\n  else context.resume().then(begin).catch(() => {});\n\n  return () => {\n    stopped = true;\n    if (timer) clearInterval(timer);\n    // Stopped rather than left to finish. Answered, this is cut off by the\n    // microphone opening a fraction of a second later, and a ring still sounding\n    // over the first word of a conversation is the thing people apologise for.\n    for (const osc of voices) {\n      try { osc.stop(); } catch { /* already stopped on its own schedule */ }\n    }\n    try { context.close(); } catch { /* nothing left to close */ }\n  };\n}\n\n/** Seconds left until a deadline, floored at zero. */\nfunction secondsLeft(ringsUntil) {\n  const until = Date.parse(ringsUntil || '');\n  if (Number.isNaN(until)) return 0;\n  return Math.max(0, Math.round((until - Date.now()) / 1000));\n}\n\nconst BUTTON = [\n  'flex:1',\n  'padding:9px 12px',\n  'border:0',\n  'border-radius:8px',\n  'font:600 13px/1.4 inherit',\n  'cursor:pointer',\n].join(';');\n\n/**\n * Put the card on screen and ring until something happens to it.\n *\n * @param {object} opts\n * @param {object} opts.call  The `incomingCall` object off the update stream —\n *   `callId`, `from`, `project`, `reference`, `ringsUntil`, `recorded`.\n * @param {string} opts.accent  The integration's own colour, for Answer.\n * @param {string} [opts.position]  Same two words the toast takes.\n * @param {() => void} opts.onAnswer   Pressed Answer.\n * @param {() => void} opts.onDecline  Pressed Decline.\n * @param {() => void} [opts.onExpire] Nobody pressed anything in time.\n * @param {() => boolean} [opts.touched]  Whether anybody has touched the page\n *   yet, which is when a sound may be made (gesture.js). Asked of the caller\n *   because this file is fetched when a call rings, and could not have been\n *   listening for the clicks that came before.\n * @returns {(reason?: string) => void} Take it away. Called with no reason on\n *   an ordinary dismissal; the caller passes one only for its own logging.\n */\nexport function showIncomingCall({\n  call, accent = '#5b6ef5', position = 'bottom-right',\n  onAnswer, onDecline, onExpire, touched = () => false,\n}) {\n  if (typeof document === 'undefined' || !document.body) return () => {};\n\n  const [vertical, horizontal] = String(position).split('-');\n  const el = document.createElement('div');\n  el.setAttribute('role', 'alertdialog');\n  el.setAttribute('aria-live', 'assertive');\n  el.setAttribute('aria-label', `${call.from || 'Support'} is calling about ${call.reference}`);\n  el.style.cssText = [\n    'position:fixed',\n    `${vertical === 'top' ? 'top' : 'bottom'}:20px`,\n    `${horizontal === 'left' ? 'left' : 'right'}:20px`,\n    // Above the update toast, which uses the maximum and may be on screen at\n    // the same moment — a status line must not sit on top of a ringing phone.\n    'z-index:2147483647',\n    'width:300px',\n    'max-width:calc(100vw - 40px)',\n    'padding:14px',\n    'border-radius:12px',\n    'background:#1b1e26',\n    'color:#eceef2',\n    'font:13px/1.5 -apple-system,BlinkMacSystemFont,\"Segoe UI\",sans-serif',\n    'box-shadow:0 18px 44px rgba(0,0,0,0.5)',\n    `border:1px solid ${accent}`,\n    'opacity:0',\n    'transform:translateY(8px)',\n    'transition:opacity .2s ease,transform .2s ease',\n  ].join(';');\n\n  const heading = document.createElement('div');\n  heading.style.cssText = 'font-weight:600;font-size:14px;margin-bottom:2px';\n  // textContent throughout: every string on this card came from a person typing\n  // it, and it is being written into somebody else's page.\n  heading.textContent = `${call.from || 'Support'} is calling`;\n\n  const about = document.createElement('div');\n  about.style.cssText = 'color:#a4abba;margin-bottom:10px';\n  about.textContent = call.project\n    ? `About your report ${call.reference} · ${call.project}`\n    : `About your report ${call.reference}`;\n\n  const countdown = document.createElement('div');\n  countdown.style.cssText = 'color:#7d8598;font-size:12px;margin-bottom:10px';\n\n  const answer = document.createElement('button');\n  answer.type = 'button';\n  answer.textContent = 'Answer';\n  answer.style.cssText = `${BUTTON};background:${accent};color:#fff`;\n\n  const decline = document.createElement('button');\n  decline.type = 'button';\n  decline.textContent = 'Not now';\n  decline.style.cssText = `${BUTTON};background:#2b2f3a;color:#c8cedb`;\n\n  const actions = document.createElement('div');\n  actions.style.cssText = 'display:flex;gap:8px';\n  actions.append(answer, decline);\n\n  el.append(heading, about, countdown);\n  if (call.recorded) {\n    const kept = 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package/dist/bugcatch-call.umd.js

@@ -0,0 +1,2 @@
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package/dist/bugcatch-call.umd.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"bugcatch-call.umd.js","sources":["../src/audio.js","../src/live-call.js","../src/incoming-call.js","../src/chunks.js","../src/umd/call.js"],"sourcesContent":["// Microphone to µ-law, and back.\n//\n// sphoro.voice speaks G.711 µ-law at 8 kHz in both directions — the format a\n// telephone carrier hands it, which is what makes a browser call and a phone\n// call indistinguishable to everything downstream. A browser has neither: it\n// captures 32-bit floats at whatever rate the sound card runs at. The gap\n// between those two is a codec and a resampler, and neither is available from\n// the platform, so both are here.\n//\n// This file exists rather than the two callers each having a copy. The agent\n// call sends µ-law because that is the whole conversation; a live call sends it\n// because sphoro.voice cannot record or transcribe a conversation it is not in\n// the middle of, and asks each browser for a copy of its own microphone. Two\n// implementations of G.711 in one package would be one implementation and one\n// that is subtly wrong on the day somebody fixes only the first.\n//\n// Nothing is imported to do it. The SDK runs on someone else's page and has no\n// dependencies, and that rule does not bend for audio.\n\n/** The wire's sample rate. Not negotiable — see the note above. */\nexport const WIRE_RATE = 8000;\n\n/**\n * An audio context, at the wire's rate where the browser will give us one.\n *\n * Asking for 8 kHz outright makes the resampler below dead code on most\n * machines. Some browsers refuse a rate their hardware does not like, so the\n * fallback is an ordinary context and the manual path.\n */\nexport function createContext(rate = WIRE_RATE) {\n  const Ctor = window.AudioContext || window.webkitAudioContext;\n  try {\n    return new Ctor({ sampleRate: rate });\n  } catch {\n    return new Ctor();\n  }\n}\n\n/**\n * Start a context that the browser has parked, if it will let us.\n *\n * Resolves either way. There is nothing useful to do with the rejection: it\n * means this browser wants a user gesture we are not inside, and the call is\n * still worth attempting — a context that stays suspended is a silent call,\n * which is what the caller finds out about from the states it is already being\n * sent.\n */\nexport async function resumeContext(context) {\n  if (context.state !== 'suspended') return;\n  try {\n    await context.resume();\n  } catch { /* no activation to spend */ }\n}\n\n/**\n * Drop a signal to the wire rate by averaging each source window.\n *\n * Exported for its own test: the browsers that refuse an 8 kHz context are the\n * ones that need this, and they are exactly the browsers a developer is least\n * likely to be looking at when they change it.\n *\n * Averaging rather than picking every nth sample: plain decimation aliases\n * everything above 4 kHz down into the speech band, and consonants are exactly\n * what lives up there. This is a crude low-pass, but it is the difference\n * between \"sixty\" and \"sixteen\" surviving the trip.\n */\nexport function downsample(input, fromRate, toRate) {\n  const ratio = fromRate / toRate;\n  const out = new Float32Array(Math.floor(input.length / ratio));\n  for (let i = 0; i < out.length; i++) {\n    const start = Math.floor(i * ratio);\n    const end = Math.min(Math.floor((i + 1) * ratio), input.length);\n    let sum = 0;\n    for (let j = start; j < end; j++) sum += input[j];\n    out[i] = end > start ? sum / (end - start) : 0;\n  }\n  return out;\n}\n\n/** G.711 µ-law: the bias and clip the standard specifies. */\nconst MU_BIAS = 0x84;\nconst MU_CLIP = 32635;\n\n/**\n * Float samples to µ-law bytes, ready to put on the wire.\n *\n * Exported for the codec test rather than for callers: this file is not the\n * package's entry point, and a hand-rolled G.711 that nothing checks is a\n * silent call nobody can debug from the outside.\n */\nexport function encodeMuLaw(samples) {\n  const out = new Uint8Array(samples.length);\n  for (let i = 0; i < samples.length; i++) {\n    // Clamped before scaling: a float outside [-1, 1] is legal in Web Audio and\n    // would wrap round to the opposite sign as an integer.\n    const clamped = Math.max(-1, Math.min(1, samples[i]));\n    let sample = Math.round(clamped * 32767);\n\n    const sign = sample < 0 ? 0x80 : 0;\n    if (sign) sample = -sample;\n    if (sample > MU_CLIP) sample = MU_CLIP;\n    sample += MU_BIAS;\n\n    let exponent = 7;\n    for (let mask = 0x4000; (sample & mask) === 0 && exponent > 0; exponent--, mask >>= 1);\n    const mantissa = (sample >> (exponent + 3)) & 0x0F;\n    out[i] = ~(sign | (exponent << 4) | mantissa) & 0xFF;\n  }\n  return out;\n}\n\n/** µ-law bytes back to float samples, ready for an AudioBuffer. */\nexport function decodeMuLaw(bytes) {\n  const out = new Float32Array(bytes.length);\n  for (let i = 0; i < bytes.length; i++) {\n    const byte = ~bytes[i] & 0xFF;\n    const sign = byte & 0x80;\n    const exponent = (byte >> 4) & 0x07;\n    const mantissa = byte & 0x0F;\n    const magnitude = (((mantissa << 3) + MU_BIAS) << exponent) - MU_BIAS;\n    out[i] = (sign ? -magnitude : magnitude) / 32768;\n  }\n  return out;\n}\n\n/**\n * How much audio rides in one frame.\n *\n * Measured in samples, and the context above runs at 8 kHz wherever the browser\n * allows it — so this is 64 ms of speech, not the 20-odd it would be at\n * hardware rate. That distinction is the whole reason for the number: at 2048\n * the same code sends a quarter of a second per frame, which is a quarter of a\n * second added to the start of every turn and to every barge-in, and it sounds\n * exactly like an agent that is slow to notice you talking.\n *\n * Not smaller than this. ScriptProcessorNode runs on the main thread, and a\n * block it cannot fill in time is a block it drops — which is a gap in the\n * middle of a word rather than a delay before one. Powers of two only.\n */\nexport const CAPTURE_BLOCK = 512;\n\n/**\n * Tap a microphone and hand each block over as µ-law bytes.\n *\n * ScriptProcessorNode is deprecated in favour of AudioWorklet, and is used\n * anyway. A worklet's processor has to be loaded from a URL, which on this SDK\n * means a blob: URL — and a blob: worklet is refused outright by the\n * Content-Security-Policy of a great many of the sites this code runs on. A\n * deprecated node that works on a customer's page beats a modern one that is\n * blocked by their headers.\n *\n * Returns a `stop` that lets go of the whole chain. The microphone track is\n * deliberately *not* stopped by it: on a live call the same track is carrying\n * the conversation, and stopping it to end a recording would end the call.\n *\n * @param {AudioContext} context\n * @param {MediaStream} stream\n * @param {(bytes: Uint8Array) => void} onFrame\n * @param {number} [wireRate]\n * @returns {{stop: () => void}}\n */\nexport function tapMicrophone(context, stream, onFrame, wireRate = WIRE_RATE) {\n  const source = context.createMediaStreamSource(stream);\n  const capture = context.createScriptProcessor(CAPTURE_BLOCK, 1, 1);\n\n  capture.onaudioprocess = (event) => {\n    const input = event.inputBuffer.getChannelData(0);\n    const frames = context.sampleRate === wireRate\n      ? input\n      : downsample(input, context.sampleRate, wireRate);\n    onFrame(encodeMuLaw(frames));\n  };\n\n  source.connect(capture);\n  // A ScriptProcessorNode does not run unless it is connected to a destination,\n  // even when nothing is meant to come out of it. Zero gain is what keeps the\n  // speaker from hearing their own voice played back at them.\n  const mute = context.createGain();\n  mute.gain.value = 0;\n  capture.connect(mute);\n  mute.connect(context.destination);\n\n  return {\n    stop() {\n      capture.onaudioprocess = null;\n      try { source.disconnect(); } catch { /* already gone */ }\n      try { capture.disconnect(); } catch { /* already gone */ }\n      try { mute.disconnect(); } catch { /* already gone */ }\n    },\n  };\n}\n","// Two people, talking directly.\n//\n// The agent call next door is a microphone, a socket and a codec: the browser\n// sends µ-law up to sphoro.voice, a model answers, and every sample passes\n// through a server. This is the opposite arrangement. The audio here goes\n// straight from one browser to the other over WebRTC, and sphoro.voice's part is\n// only to introduce them — it passes an offer, an answer and a handful of ICE\n// candidates between the two sockets and then has nothing left to do.\n//\n// Which means there is no codec in this file, no resampler, and no audio path\n// at all. The browser's own WebRTC stack owns those, and it is better at them\n// than anything hand-written: Opus at whatever rate the two ends agree on,\n// echo cancellation, jitter buffering and packet loss concealment, none of\n// which the µ-law path can have because a telephone carrier cannot receive\n// them. A live call sounds better than the agent call, and costs nothing per\n// minute, because nobody is in the middle of it.\n//\n// # Shared with the dashboard\n//\n// Both ends of a live call run this file: the reporter's widget and the member\n// who answers in the portal, which imports it from here rather than keeping a\n// second copy. The two sides differ by one flag — who makes the offer — and it\n// arrives from the server rather than being decided here, so the two ends\n// cannot both sit waiting for the other to go first.\n//\n// # The one exception: the copy that gets kept\n//\n// An agent that is set to record or transcribe cannot do either from here —\n// the audio it would need never reaches it. So sphoro.voice asks instead: it\n// names, in `session.created`, what this call is keeping and in what format,\n// and each browser uplinks a copy of *its own* microphone over the signaling\n// socket it already has open. That copy is the only way a recording or a\n// transcript of a peer-to-peer call can exist at all, and it is why the codec\n// this file said it did not have is imported below.\n//\n// It is a copy and not the call. The conversation still runs browser to\n// browser; an uplink that stalls or is refused costs a recording, never a call.\n//\n// Nothing outside the package is imported to do any of it. The SDK runs on\n// someone else's page and has no dependencies, and that rule does not bend for\n// a peer connection.\n\nimport { createContext, resumeContext, tapMicrophone } from './audio.js';\n\n/**\n * sphoro.voice's realtime protocol, sent as the WebSocket subprotocol. The same\n * one the agent call speaks — a bridged call is a different conversation over\n * the same envelope, not a different protocol.\n */\nconst SUBPROTOCOL = 'voiceai.realtime.v1';\n\n/**\n * Where a browser looks to find its way to the other one, when the server names\n * nowhere.\n *\n * The server almost always does name somewhere — it is configured per\n * deployment, because a network behind symmetric NAT needs a TURN relay and\n * that costs money. This is the floor: a public STUN server, which is enough\n * for most home and office networks and free.\n */\nconst FALLBACK_ICE = [{ urls: 'stun:stun.l.google.com:19302' }];\n\n/**\n * Ask for the microphone.\n *\n * Separate from joining, and deliberately. The permission prompt takes as long\n * as the person takes to read it, and both ends have something they should not\n * do until it is answered: the reporter should not make a room full of\n * dashboards ring before knowing they can speak, and whoever answers should not\n * claim a call they then cannot take. So both ask first and act second.\n *\n * @returns {Promise<MediaStream>}\n */\nexport function openMicrophone() {\n  return navigator.mediaDevices.getUserMedia({\n    audio: {\n      // Both ends are on speakers as often as not. Without cancellation each\n      // hears themselves back through the other, half a second late, which is\n      // the single most effective way to make a conversation impossible.\n      echoCancellation: true,\n      noiseSuppression: true,\n      autoGainControl: true,\n      channelCount: 1,\n    },\n  });\n}\n\n/** Whether this browser can be on a live call at all. */\nexport function liveCallSupported() {\n  return typeof window !== 'undefined'\n    && typeof window.RTCPeerConnection === 'function'\n    && Boolean(navigator.mediaDevices?.getUserMedia);\n}\n\n/**\n * Join a live call and stay on it until somebody hangs up.\n *\n * @param {object} opts\n * @param {string} opts.realtimeUrl  From the API, carrying its own credential\n *   and its own seat. Never log it and never build one by hand.\n * @param {MediaStream} [opts.stream]  A microphone already opened by\n *   `openMicrophone`. One is opened here if not, which is the wrong order for\n *   anything that had a decision to make first.\n * @param {string} [opts.ringsUntil]  ISO-8601, on the caller's side only: when\n *   the ringing stops. The browser is the thing that decides it has, because it\n *   is the one client guaranteed to still be listening at that moment.\n * @param {(state: string, detail?: object) => void} [opts.onState]\n *   'ringing' | 'connecting' | 'connected' | 'reconnecting' | 'ended' | 'missed'\n *   | 'failed'. 'reconnecting' is not a failure and never ends a call — see\n *   `routeChanged`.\n * @param {(blocked: boolean) => void} [opts.onAudioBlocked]  The browser refused\n *   to play the other person, or has since started. Its own callback rather\n *   than a state, because it is orthogonal to every one of them: the call is\n *   connected and working, and one end cannot hear it. See `playRemote`.\n * @param {(live: boolean) => void} [opts.onMicrophone]  Whether this person's\n *   microphone is still reaching the call. Also orthogonal, and for the same\n *   reason: the call is up, the other end is audible, and nothing this person\n *   says is going anywhere. See `watchMicrophone`.\n * @returns {Promise<{hangUp: () => void, setMuted: (muted: boolean) => boolean,\n *   isMuted: () => boolean}>}\n */\nexport async function joinLiveCall({\n  realtimeUrl, stream, ringsUntil, onState, onAudioBlocked, onMicrophone,\n}) {\n  const emit = (state, detail) => { try { onState?.(state, detail); } catch { /* theirs */ } };\n  const sayBlocked = (blocked) => {\n    try { onAudioBlocked?.(blocked); } catch { /* theirs */ }\n  };\n  const sayMicrophone = (live) => {\n    try { onMicrophone?.(live); } catch { /* theirs */ }\n  };\n\n  if (!liveCallSupported()) {\n    const err = new Error('[BugCatch] this browser cannot make live calls.');\n    emit('failed', { reason: 'unsupported' });\n    throw err;\n  }\n\n  let microphone = stream;\n  if (!microphone) {\n    try {\n      microphone = await openMicrophone();\n    } catch (err) {\n      emit('failed', { reason: 'microphone', error: String(err?.name || err) });\n      throw err;\n    }\n  }\n\n  let socket;\n  try {\n    socket = await openSocket(realtimeUrl);\n  } catch (err) {\n    stopStream(microphone);\n    emit('failed', { reason: 'connect' });\n    throw err;\n  }\n\n  let peer = null;\n  let remoteAudio = null;\n  let finished = false;\n  let ringTimer = null;\n  // What sphoro.voice said this call is keeping, and the tap feeding it. Both\n  // stay null on the ordinary call, which keeps nothing and uplinks nothing.\n  let capture = null;\n  let uplink = null;\n  // Whether the browser is currently refusing to play the other person, and the\n  // teardown for the gesture listeners armed to try again. See playRemote.\n  let blocked = false;\n  let disarmGesture = null;\n  /** Teardown for the microphone watch, armed once the peer connection is built. */\n  let unwatchMicrophone = null;\n  // Claimed synchronously, before the first await below, because the seat that\n  // answers is told the call has started twice — once in its own\n  // `session.created`, which says somebody is already waiting, and again in the\n  // `peer.joined` that goes to both ends a moment later. Two starts that both\n  // check `uplink` before either has set it are two taps on one microphone, and\n  // two copies of every frame interleaved into one recording.\n  let starting = false;\n  // Candidates that arrived before there was anywhere to put them.\n  //\n  // The offer and the first candidates are sent within a millisecond of each\n  // other and arrive in that order, but `setRemoteDescription` is a promise —\n  // so a candidate can be in hand while the description it belongs to is still\n  // being applied. Adding one then throws, and the call comes up with a worse\n  // route than it could have had, or none.\n  const pending = [];\n\n  // --- surviving a change of network ------------------------------------------------\n  //\n  // A phone call moves. Somebody walks out of the office and the handset hands the\n  // connection from Wi-Fi to cellular; a lift takes the signal for four seconds; a laptop\n  // wakes on a different access point. Every one of those changes the local address the\n  // call was negotiated against, and WebRTC's answer to all of them is the same: the ICE\n  // agent reports `disconnected`, and unless somebody restarts it, `failed` a moment later.\n  //\n  // Before this, `failed` ended the call and `disconnected` was not handled at all — so a\n  // call that would have recovered on its own in two seconds was hung up on, and one that\n  // needed a restart was hung up on with no attempt made. On a desk that is rare. On the\n  // phone this app also ships as, it is what happens when you stand up.\n  //\n  // What replaces it is a grace period, then an ICE restart, then a deadline. Each is\n  // there for a distinct reason:\n  //\n  //   the grace   `disconnected` is frequently transient — a couple of lost packets on a\n  //               route that is still fine. Restarting immediately would tear down a\n  //               working connection to rebuild the same one, and take audio with it.\n  //\n  //   the restart Renegotiation from scratch over the signaling socket, which is still\n  //               open: it never depended on the media route. This is the part that\n  //               actually crosses a network change.\n  //\n  //   the deadline Reconnecting forever is its own failure. Somebody staring at a card\n  //               that says \"reconnecting\" for five minutes has been abandoned more\n  //               thoroughly than somebody told the call dropped.\n  //\n  // Only one side offers, and it is the side the server nominated at the start. Both ends\n  // see the drop, so if the two sides both restarted there would be two offers crossing —\n  // a glare condition WebRTC resolves badly. The other side asks instead, over the\n  // signaling socket, and its request is best-effort: the offerer's own detection is what\n  // this actually rests on.\n\n  /** Which side was told to make the offer. Set from `peer.joined`, for both ends. */\n  let shouldOffer = false;\n  /** Whether the two ends have ever been connected. A drop before that is a failed setup. */\n  let connectedOnce = false;\n  let reconnecting = false;\n  let recoverTimer = null;\n  let giveUpTimer = null;\n  let restarts = 0;\n\n  /** How long `disconnected` is allowed to resolve itself before a restart is attempted. */\n  const RECOVER_GRACE_MS = 2000;\n  /** How long between one restart and the next, when the first did not take. */\n  const RECOVER_RETRY_MS = 5000;\n  /** How long a call may spend off the air in total before it is called dropped. */\n  const RECOVER_DEADLINE_MS = 30000;\n  const MAX_RESTARTS = 4;\n\n  /** Everything ends here, whatever ended it, exactly once. */\n  const finish = (state, detail) => {\n    if (finished) return;\n    finished = true;\n    if (ringTimer) { clearTimeout(ringTimer); ringTimer = null; }\n    // Before the peer is closed, and before the stream is stopped. Closing a connection\n    // moves it to `closed` and stopping a track fires `ended`, and both would otherwise\n    // reach handlers that would report a dropped call and a dead microphone to somebody\n    // who has just hung up.\n    clearRecovery();\n    unwatchMicrophone?.();\n    unwatchMicrophone = null;\n    stopUplink();\n    // Before anything else is torn down: these are listeners on `window`, and a\n    // call that left them behind would keep retrying playback on an element\n    // that is about to be removed — once per click, for the rest of the session.\n    disarmGesture?.();\n    disarmGesture = null;\n    try { peer?.close(); } catch { /* already gone */ }\n    stopStream(microphone);\n    if (remoteAudio) {\n      remoteAudio.srcObject = null;\n      remoteAudio.remove();\n      remoteAudio = null;\n    }\n    try {\n      if (socket.readyState === WebSocket.OPEN) socket.close(1000, 'call over');\n    } catch { /* already closing */ }\n    emit(state, detail);\n  };\n\n  const send = (message) => {\n    if (socket.readyState === WebSocket.OPEN) socket.send(JSON.stringify(message));\n  };\n\n  /**\n   * Start sending sphoro.voice a copy of this person's microphone.\n   *\n   * Only when it asked, and only once both people are on the call. The server\n   * throws away anything that arrives while the phone is still ringing — a\n   * recording is timed from the moment the second person joined, and a minute\n   * of somebody waiting alone at the head of the file is a timeline that\n   * disagrees with the call's own duration. Sending it anyway would just be\n   * bandwidth spent on audio that gets dropped, on the same network carrying\n   * the call.\n   *\n   * Every failure here is survivable and none of them ends the call. A browser\n   * with no Web Audio, a context the autoplay policy will not start, a format\n   * this version does not know how to produce — each costs the recording and\n   * leaves two people talking, which is the right way round. It is said out\n   * loud, though: an agent set to record whose calls silently arrive with no\n   * audio is the failure that is impossible to diagnose from the outside.\n   */\n  const startUplink = async () => {\n    if (starting || uplink || finished || !capture) return;\n    if (!capture.audio && !capture.transcript) return;\n    starting = true;\n\n    // The server names the format rather than assuming one, so a client that\n    // does not recognise it sends nothing rather than sending µ-law into a\n    // recording that expected something else — which would store a file of\n    // noise and read as a working recording to everything downstream.\n    const encoding = capture.encoding || 'mulaw';\n    if (encoding !== 'mulaw') {\n      warn(`this call is kept as ${encoding}, which this SDK cannot produce.`);\n      return;\n    }\n\n\n    const rate = Number(capture.sample_rate) || 8000;\n    try {\n      const context = createContext(rate);\n      // A context built without a user gesture starts suspended, and a\n      // suspended context's ScriptProcessorNode never fires — so nothing would\n      // be uplinked and nothing would say why. Both ends granted microphone\n      // access a moment ago, which is the activation this spends.\n      await resumeContext(context);\n      if (finished) {\n        try { context.close(); } catch { /* already closed */ }\n        return;\n      }\n\n      const tap = tapMicrophone(context, microphone, (bytes) => {\n        // Binary, not the base64 envelope the protocol also accepts: this is a\n        // second copy of audio going up a network that is already carrying the\n        // call, and base64 would make it a third larger for nothing.\n        if (socket.readyState === WebSocket.OPEN) socket.send(bytes);\n      }, rate);\n\n      uplink = { context, tap };\n    } catch (err) {\n      warn(`could not send the copy this call is keeping: ${String(err?.message || err)}`);\n    }\n  };\n\n  /**\n   * Stop sending it.\n   *\n   * The microphone track is deliberately left alone — `finish` stops the whole\n   * stream a moment later, and stopping it here would cut the conversation off\n   * to end a recording.\n   */\n  function stopUplink() {\n    starting = false;\n    if (!uplink) return;\n    const { context, tap } = uplink;\n    uplink = null;\n    try { tap.stop(); } catch { /* already gone */ }\n    try { context.close(); } catch { /* already closed */ }\n  }\n\n  /**\n   * Play the other person, and do something about it when the browser says no.\n   *\n   * The refusal used to be swallowed: `play().catch(() => {})`, on the reasoning\n   * that both ends had granted microphone access a moment earlier and that\n   * satisfies every autoplay policy. It does not always. A call answered from a\n   * notification arrives with no user gesture on the page at all — the tap\n   * happened on a system notification, which is not this document — and inside\n   * an Android WebView that is exactly the case that fails.\n   *\n   * What that produced is the worst shape a bug can have: the call connects,\n   * every state says `connected`, both ends are billed for it, and one person\n   * hears nothing. Nothing is logged, because the rejection was caught and\n   * dropped, so there is no evidence anywhere that it happened.\n   *\n   * So now it is reported and it is retried. The retry is armed on the next\n   * gesture anywhere on the page, which is the one thing that reliably lifts an\n   * autoplay block, and it is armed in capture so a handler calling\n   * `stopPropagation` cannot hide the gesture from it. Whoever is on the call is\n   * about to touch something — the mute button, the hang-up button, the report\n   * they are talking about — and the sound comes up when they do, with the\n   * caller free to say so on screen in the meantime.\n   */\n  function playRemote() {\n    if (!remoteAudio || finished) return;\n    let attempt;\n    try {\n      attempt = remoteAudio.play?.();\n    } catch {\n      // A synchronous throw, which some engines do instead of rejecting.\n      markBlocked();\n      return;\n    }\n    if (!attempt || typeof attempt.then !== 'function') {\n      // No promise: an older engine, or a stub. Nothing to wait on and nothing\n      // to report — it either made a sound or it did not.\n      markPlaying();\n      return;\n    }\n    attempt.then(markPlaying, markBlocked);\n  }\n\n  function markPlaying() {\n    disarmGesture?.();\n    disarmGesture = null;\n    if (!blocked) return;\n    blocked = false;\n    sayBlocked(false);\n  }\n\n  function markBlocked() {\n    if (finished) return;\n    if (!blocked) {\n      blocked = true;\n      sayBlocked(true);\n    }\n    armGesture();\n  }\n\n  /** Try again on the next thing the person touches. Armed once. */\n  function armGesture() {\n    if (disarmGesture) return;\n    // Guarded rather than assumed: this file is also loaded by a host page's\n    // bundler into environments that define a partial `window`, and a retry\n    // that cannot be armed is a quieter failure than one that throws inside a\n    // promise nobody is awaiting.\n    if (typeof window === 'undefined' || typeof window.addEventListener !== 'function') return;\n    const retry = () => playRemote();\n    const events = ['pointerdown', 'keydown', 'touchstart'];\n    for (const name of events) {\n      window.addEventListener(name, retry, { capture: true, passive: true });\n    }\n    disarmGesture = () => {\n      for (const name of events) {\n        try { window.removeEventListener(name, retry, { capture: true }); } catch { /* gone */ }\n      }\n    };\n  }\n\n  /**\n   * Build the peer connection.\n   *\n   * Made on `session.created` rather than lazily on the first signal, because\n   * the ICE gathering it starts is the slowest part of connecting and the whole\n   * of it can happen while the phone is still ringing. Waiting until somebody\n   * answers would add that delay to the exact moment a person says hello, which\n   * is the one moment on this call where it is audible.\n   */\n  const build = (iceServers) => {\n    peer = new RTCPeerConnection({ iceServers: iceServers?.length ? iceServers : FALLBACK_ICE });\n\n    for (const track of microphone.getAudioTracks()) peer.addTrack(track, microphone);\n\n    // Armed here rather than at the top, so a call that never got a peer connection has\n    // nothing to tear down — and so the watch covers exactly the span in which being\n    // unheard is a problem worth telling somebody about.\n    unwatchMicrophone = watchMicrophone();\n\n    peer.onicecandidate = (event) => {\n      if (!event.candidate) return;\n      send({ type: 'signal', signal: { kind: 'candidate', payload: event.candidate.toJSON() } });\n    };\n\n    peer.ontrack = (event) => {\n      const [incoming] = event.streams;\n      if (!incoming) return;\n      // An element rather than the Web Audio API. Audio routed through a\n      // MediaStreamAudioSourceNode loses the browser's own echo cancellation on\n      // several engines, and this call has a live microphone next to it.\n      if (!remoteAudio) {\n        remoteAudio = document.createElement('audio');\n        remoteAudio.autoplay = true;\n        // Inline, for a WebView that would otherwise hand a media element to a\n        // full-screen player — which on a call is a black rectangle over the\n        // dashboard.\n        remoteAudio.playsInline = true;\n        remoteAudio.setAttribute?.('playsinline', '');\n        // Off-screen rather than display:none — a hidden media element is\n        // treated as invisible by some autoplay heuristics and refused.\n        remoteAudio.style.cssText = 'position:fixed;width:0;height:0;opacity:0;pointer-events:none';\n        document.body.appendChild(remoteAudio);\n      }\n      remoteAudio.srcObject = incoming;\n      playRemote();\n    };\n\n    peer.onconnectionstatechange = routeChanged;\n    // The same signal, read the other way, for the engines that do not give a reliable\n    // `connectionState`. WebKit — which is every browser on iOS and the WebView this app\n    // ships inside there — has historically fired `iceconnectionstatechange` and not the\n    // aggregate one. Both are wired to the same handler, which reads whichever of the two\n    // the engine actually maintains, so a duplicate event costs one comparison.\n    peer.oniceconnectionstatechange = routeChanged;\n  };\n\n  /**\n   * What the media route is doing, and what to do about it.\n   *\n   * Reads whichever state the engine keeps up to date, and treats `completed` — which only\n   * the ICE-level state has — as connected, because it is: it means a route was found and\n   * checking has stopped.\n   */\n  function routeChanged() {\n    if (!peer || finished) return;\n    const state = peer.connectionState || peer.iceConnectionState;\n\n    if (state === 'connected' || state === 'completed') {\n      connectedOnce = true;\n      restarts = 0;\n      clearRecovery();\n      // Emitted whether or not this is a reconnection. The caller renders one state, and a\n      // card that came back from \"reconnecting\" has to be told the call is up again.\n      reconnecting = false;\n      emit('connected');\n      return;\n    }\n\n    if (state === 'disconnected') {\n      // Before the two ends ever met, a `disconnected` is ICE still working — it goes\n      // there between checking one candidate pair and the next. Only `failed` is final at\n      // that stage, and it is handled below.\n      if (!connectedOnce) return;\n      enterRecovery(RECOVER_GRACE_MS);\n      return;\n    }\n\n    if (state === 'failed') {\n      if (!connectedOnce) {\n        // The two browsers never found a route to each other at all. Almost always a\n        // network that needs a TURN relay the deployment has not configured; reported as\n        // its own reason so that is diagnosable from a support ticket rather than from a\n        // packet capture. A restart would try the same candidates again.\n        finish('failed', { reason: 'ice' });\n        return;\n      }\n      // A route that worked and then stopped. That is a network change, and a restart is\n      // exactly the remedy — no grace, because `failed` is not going to resolve itself.\n      enterRecovery(0);\n    }\n  }\n\n  /** Say the call is off the air, and line up an attempt to bring it back. */\n  function enterRecovery(delay) {\n    if (finished) return;\n    if (!reconnecting) {\n      reconnecting = true;\n      emit('reconnecting');\n      giveUpTimer = setTimeout(() => finish('failed', { reason: 'network' }), RECOVER_DEADLINE_MS);\n    }\n    // One attempt in flight at a time. `disconnected` and `failed` both arrive on the way\n    // down, and two timers would mean two offers crossing on one connection.\n    if (recoverTimer) return;\n    recoverTimer = setTimeout(recover, delay);\n  }\n\n  function clearRecovery() {\n    if (recoverTimer) { clearTimeout(recoverTimer); recoverTimer = null; }\n    if (giveUpTimer) { clearTimeout(giveUpTimer); giveUpTimer = null; }\n  }\n\n  /**\n   * Renegotiate the media route over the signaling socket, which never left.\n   *\n   * `createOffer({ iceRestart: true })` rather than `restartIce()`: the latter is the\n   * modern spelling and does the same thing by firing `negotiationneeded`, but it is\n   * missing from engines this SDK is expected to run in and the explicit form works\n   * everywhere. It also keeps the offer on the same path as the original one, so there is a\n   * single place where an offer is made.\n   */\n  async function recover() {\n    recoverTimer = null;\n    if (finished || !peer) return;\n    // It came back on its own while the grace period was running, which is the common case\n    // and the reason the grace period exists.\n    const state = peer.connectionState || peer.iceConnectionState;\n    if (state === 'connected' || state === 'completed') return;\n\n    if (!shouldOffer) {\n      // Not this side's to restart. Ask, and keep asking — the offerer sees the same drop\n      // and is very likely already doing it, so this is the belt to that braces.\n      send({ type: 'signal', signal: { kind: 'restart', payload: {} } });\n      recoverTimer = setTimeout(recover, RECOVER_RETRY_MS);\n      return;\n    }\n\n    if (restarts >= MAX_RESTARTS) return; // the deadline is what ends it\n    restarts += 1;\n    await restartIce();\n    recoverTimer = setTimeout(recover, RECOVER_RETRY_MS);\n  }\n\n  /** The restart itself, also reachable from the other side's request. */\n  async function restartIce() {\n    if (finished || !peer) return;\n    try {\n      const description = await peer.createOffer({ iceRestart: true });\n      await peer.setLocalDescription(description);\n      send({ type: 'signal', signal: { kind: 'offer', payload: peer.localDescription.toJSON() } });\n    } catch {\n      // Nothing to do here but let the next attempt or the deadline decide. Ending the\n      // call on one refused renegotiation would throw away the retries that follow.\n    }\n  }\n\n  /**\n   * Whether this person is still being heard.\n   *\n   * A track's `muted` is not the mute button — that is `enabled`, which this file sets and\n   * therefore already knows about. `muted` is the *source* stopping: the OS handed the\n   * microphone to an incoming phone call, a headset was unplugged, or — the case this app\n   * meets daily — Android revoked microphone access because the app went to the background.\n   *\n   * Which produces the same shape as every other bug on this path: the call is up, the\n   * other person is audible, both ends are billed, and nothing this person says goes\n   * anywhere. They find out when the other person says \"hello? are you there?\".\n   *\n   * The loss is reported on a short delay and the recovery immediately. Several engines\n   * flick `muted` on for a few frames while a track settles, and a card that said \"your\n   * microphone was cut off\" for 200ms at the start of every call would be noise — while a\n   * microphone that came back and was still reported lost is the error saying the wrong\n   * thing.\n   */\n  function watchMicrophone() {\n    const SETTLE_MS = 600;\n    let settle = null;\n    let lost = false;\n\n    const say = (live) => {\n      if (settle) { clearTimeout(settle); settle = null; }\n      if (live) {\n        if (!lost) return;\n        lost = false;\n        sayMicrophone(true);\n        return;\n      }\n      if (lost) return;\n      settle = setTimeout(() => {\n        settle = null;\n        if (finished) return;\n        lost = true;\n        sayMicrophone(false);\n      }, SETTLE_MS);\n    };\n\n    for (const track of microphone.getAudioTracks()) {\n      track.onmute = () => say(false);\n      track.onunmute = () => say(true);\n      // An ended track never comes back — the device is gone. Said at once rather than\n      // after the settle, because there is nothing to wait for.\n      track.onended = () => {\n        if (settle) { clearTimeout(settle); settle = null; }\n        if (lost) return;\n        lost = true;\n        sayMicrophone(false);\n      };\n    }\n    return () => {\n      if (settle) { clearTimeout(settle); settle = null; }\n      for (const track of microphone.getAudioTracks()) {\n        track.onmute = null;\n        track.onunmute = null;\n        track.onended = null;\n      }\n    };\n  }\n\n  const offer = async () => {\n    try {\n      const description = await peer.createOffer();\n      await peer.setLocalDescription(description);\n      send({ type: 'signal', signal: { kind: 'offer', payload: peer.localDescription.toJSON() } });\n    } catch {\n      finish('failed', { reason: 'negotiation' });\n    }\n  };\n\n  const accept = async (payload) => {\n    try {\n      await peer.setRemoteDescription(new RTCSessionDescription(payload));\n      const answer = await peer.createAnswer();\n      await peer.setLocalDescription(answer);\n      send({ type: 'signal', signal: { kind: 'answer', payload: peer.localDescription.toJSON() } });\n      await drainCandidates();\n    } catch {\n      finish('failed', { reason: 'negotiation' });\n    }\n  };\n\n  const drainCandidates = async () => {\n    while (pending.length) {\n      const candidate = pending.shift();\n      try {\n        await peer.addIceCandidate(new RTCIceCandidate(candidate));\n      } catch {\n        // One unusable candidate is not a failed call — there are usually\n        // several, and the connection needs only one of them to work.\n      }\n    }\n  };\n\n  /**\n   * One message at a time, in the order they arrived.\n   *\n   * Handling is asynchronous — `setRemoteDescription` and `createAnswer` are\n   * promises — and the socket does not wait. Without this chain a candidate\n   * arriving mid-answer is examined while the description it belongs to is\n   * still being applied, so it goes on the pending list a moment after the\n   * pending list was drained, and stays there for the rest of the call.\n   */\n  let handling = Promise.resolve();\n  socket.onmessage = (event) => {\n    handling = handling.then(() => handle(event)).catch(() => { /* one bad frame */ });\n  };\n\n  const handle = async (event) => {\n    if (typeof event.data !== 'string') return;\n    let message;\n    try { message = JSON.parse(event.data); } catch { return; }\n\n    switch (message.type) {\n      case 'session.created': {\n        build(message.data?.ice_servers);\n        // What this call keeps, and in what format. Held rather than acted on:\n        // nothing is uplinked until somebody is actually on the other end.\n        capture = message.data?.capture || null;\n        // Ringing, until somebody is on the other side. Whoever answers finds\n        // the caller already waiting and skips straight past it — the\n        // `peer.joined` that follows a moment later says the same thing, and\n        // this is what keeps the answerer's first frame from being silence.\n        if (message.data?.peer_present) {\n          emit('connecting');\n          void startUplink();\n          break;\n        }\n        emit('ringing');\n        armRingTimer();\n        break;\n      }\n\n      case 'peer.joined': {\n        if (ringTimer) { clearTimeout(ringTimer); ringTimer = null; }\n        emit('connecting');\n        // Told, not inferred. Both ends learn somebody arrived in the same\n        // frame, and only one of them is asked to go first.\n        //\n        // Remembered as well as acted on: it is also the answer to \"whose job is it to\n        // renegotiate when the network changes\", and that question is asked later, when\n        // the frame that answered it is long gone. See `recover`.\n        shouldOffer = Boolean(message.data?.should_offer);\n        if (shouldOffer) await offer();\n        // After the offer, and not awaited: opening an audio context is\n        // milliseconds, but they would be milliseconds added to the moment\n        // somebody says hello, which is the one place on this call where they\n        // are audible.\n        void startUplink();\n        break;\n      }\n\n      case 'signal': {\n        if (!peer) break;\n        const { kind, payload } = message.data || {};\n        if (kind === 'offer') {\n          await accept(payload);\n        } else if (kind === 'answer') {\n          try {\n            await peer.setRemoteDescription(new RTCSessionDescription(payload));\n            await drainCandidates();\n          } catch {\n            finish('failed', { reason: 'negotiation' });\n          }\n        } else if (kind === 'candidate') {\n          if (peer.remoteDescription) {\n            try { await peer.addIceCandidate(new RTCIceCandidate(payload)); } catch { /* see above */ }\n          } else {\n            pending.push(payload);\n          }\n        } else if (kind === 'restart') {\n          // The other side lost the route and cannot renegotiate itself. Honoured only by\n          // the side that was nominated to offer, so a request arriving at both ends of a\n          // mis-signalled call cannot produce two offers crossing.\n          if (shouldOffer) await restartIce();\n        }\n        break;\n      }\n\n      case 'peer.left':\n        finish('ended', { reason: 'peer_left' });\n        break;\n\n      case 'error':\n        finish('failed', { reason: 'server', detail: message.error?.detail });\n        break;\n\n      case 'session.ended':\n        finish('ended', { reason: 'server' });\n        break;\n\n      default:\n        // Every message the agent conversation has — audio deltas, transcripts,\n        // state changes. None of them happen on a bridged call, and one\n        // arriving is not worth ending a working call over.\n        break;\n    }\n  };\n\n  socket.onclose = () => finish('ended', { reason: 'disconnected' });\n  socket.onerror = () => finish('failed', { reason: 'socket' });\n\n  /**\n   * Stop waiting.\n   *\n   * The deadline comes from the server, which wrote it down when the call was\n   * created — so a browser with a slow clock waits a little longer rather than\n   * hanging up on somebody mid-answer, and the server refuses a give-up that\n   * arrives early in any case.\n   */\n  function armRingTimer() {\n    if (!ringsUntil) return;\n    const remaining = new Date(ringsUntil).getTime() - Date.now();\n    if (!Number.isFinite(remaining)) return;\n    ringTimer = setTimeout(() => finish('missed'), Math.max(0, remaining));\n  }\n\n  return {\n    hangUp() {\n      // Said out loud before the socket closes. 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Asked of the caller\n *   because this file is fetched when a call rings, and could not have been\n *   listening for the clicks that came before.\n * @returns {(reason?: string) => void} Take it away. Called with no reason on\n *   an ordinary dismissal; the caller passes one only for its own logging.\n */\nexport function showIncomingCall({\n  call, accent = '#5b6ef5', position = 'bottom-right',\n  onAnswer, onDecline, onExpire, touched = () => false,\n}) {\n  if (typeof document === 'undefined' || !document.body) return () => {};\n\n  const [vertical, horizontal] = String(position).split('-');\n  const el = document.createElement('div');\n  el.setAttribute('role', 'alertdialog');\n  el.setAttribute('aria-live', 'assertive');\n  el.setAttribute('aria-label', `${call.from || 'Support'} is calling about ${call.reference}`);\n  el.style.cssText = [\n    'position:fixed',\n    `${vertical === 'top' ? 'top' : 'bottom'}:20px`,\n    `${horizontal === 'left' ? 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