@guava-ai/guava-sdk 0.37.0 → 0.38.0

package/README.md

@@ -1,7 +1,7 @@
 # Guava Typescript SDK
 [![NPM Version](https://img.shields.io/npm/v/%40guava-ai%2Fguava-sdk)](https://www.npmjs.com/package/@guava-ai/guava-sdk)
 
-This library allows you to build Guava voice agents using TypeScript or Javascript. Currently only NodeJS is officially supported as a runtime.
+This library allows you to build Guava voice agents using TypeScript or Javascript. Currently only NodeJS is officially supported as a runtime for the main SDK. The WebRTC portion of the SDK can be used from the browser.
 
 ## Documentation
 
@@ -40,4 +40,46 @@ $ yarn add @guava-ai/guava-sdk
 $ pnpm add @guava-ai/guava-sdk
 ```
 
-The SDK can be used with Javascript, but TypeScript is highly recommended.
+The SDK can be used with Javascript, but TypeScript is highly recommended.
+
+## Using WebRTC from the Browser
+
+To call your agent from the browser with a bundler, install the SDK as above and import from `@guava-ai/guava-sdk/webrtc`.
+
+```typescript
+import * as Guava from "@guava-ai/guava-sdk/webrtc";
+
+const call = new Guava.Call({ webrtcCode: "..." });
+await call.start();
+```
+
+You can also load a prebuilt bundle from a CDN.
+
+```html
+<script
+  src="https://cdn.jsdelivr.net/npm/@guava-ai/guava-sdk@0.38.0/dist/guava-webrtc.min.js"
+  integrity="sha384-6MN0aBFnGuNmEmoeTtbYR4z0Irx2sJ26LYWjUOUyrKlAnAugcZRQOIpnpNz3zAo5"
+  crossorigin="anonymous"
+></script>
+<script>
+  const call = new Guava.Call({ webrtcCode: "..." });
+  call.start();
+</script>
+```
+
+Or as an ES module:
+
+```html
+<script type="importmap">
+  {
+    "integrity": {
+      "https://cdn.jsdelivr.net/npm/@guava-ai/guava-sdk@0.38.0/dist/guava-webrtc.esm.min.js": "sha384-qnseWbldkzYVfnszRGXZuEDW2mg62vnYWLJpAk8wjfFdt+g0qBhvU4b1wnYVvkMJ"
+    }
+  }
+</script>
+<script type="module">
+  import * as Guava from "https://cdn.jsdelivr.net/npm/@guava-ai/guava-sdk@0.38.0/dist/guava-webrtc.esm.min.js";
+  const call = new Guava.Call({ webrtcCode: "..." });
+  await call.start();
+</script>
+```

package/dist/guava-webrtc.esm.min.js

@@ -0,0 +1,2 @@
+var m=Object.defineProperty;var v=(s,e,t)=>e in s?m(s,e,{enumerable:!0,configurable:!0,writable:!0,value:t}):s[e]=t;var n=(s,e,t)=>v(s,typeof e!="symbol"?e+"":e,t);function l(s,e){return s*(60/e)}var h=class{constructor(){n(this,"context",null);n(this,"intervalId",null);n(this,"volume",.15);n(this,"onSeconds",2);n(this,"periodSeconds",6);n(this,"startDelay",.3)}start(){if(this.context)return;let e=new AudioContext,t=e.createGain();t.gain.value=0,t.connect(e.destination);for(let a of[440,480]){let c=e.createOscillator();c.frequency.value=a,c.connect(t),c.start()}let i=e.currentTime+this.startDelay,r=()=>{t.gain.setValueAtTime(this.volume,i),t.gain.setValueAtTime(0,i+this.onSeconds),i+=this.periodSeconds};r(),this.intervalId=setInterval(r,this.periodSeconds*1e3),this.context=e}stop(){this.context&&(this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null),this.context.close(),this.context=null)}},d=class{constructor(){n(this,"context",null);n(this,"masterGain",null);n(this,"intervalId",null);n(this,"running",!1);n(this,"pulseCount",0);n(this,"nextBarStart",0);n(this,"volume",.25);n(this,"bpm",120);n(this,"beatsPerBar",4);n(this,"startDelay",.3)}start(){if(this.running)return;this.running=!0,this.pulseCount=0,this.context=new AudioContext,this.masterGain=this.context.createGain(),this.masterGain.gain.value=this.volume,this.masterGain.connect(this.context.destination),this.nextBarStart=this.context.currentTime+this.startDelay,this.playPulse(),this.playPulse();let e=l(this.beatsPerBar,this.bpm);this.intervalId=setInterval(()=>{this.playPulse()},e*1e3)}noteToFrequency(e){let t=e.match(/^([A-G])(#|b)?(-?\d+)$/);if(!t)throw new Error(`Invalid note: ${e}`);let[,i,r,a]=t,c={C:0,D:2,E:4,F:5,G:7,A:9,B:11},u=r==="#"?1:r==="b"?-1:0,_=(Number(a)+1)*12+c[i]+u;return 440*Math.pow(2,(_-69)/12)}stop(){if(!this.running)return;this.running=!1,this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null);let e=this.context,t=this.masterGain;if(!e||!t)return;let i=e.currentTime;t.gain.setValueAtTime(t.gain.value,i),t.gain.linearRampToValueAtTime(0,i+.15),setTimeout(()=>{e.close()},200),this.context=null,this.masterGain=null}playPulse(){if(!this.context||!this.masterGain)return;let e=this.nextBarStart,t=r=>e+l(r,this.bpm),i=l(1,this.bpm);this.pulseCount%4===0||this.pulseCount%4===2?(this.playNote("C5",t(0),i),this.playNote("E5",t(.5),i),this.playNote("G4",t(1),i)):this.pulseCount%4===3?(this.playNote("E5",t(0),i),this.playNote("G5",t(.5),i),this.playNote("C5",t(1),i)):(this.playNote("E5",t(0),i),this.playNote("G4",t(.5),i),this.playNote("C5",t(1),i)),this.pulseCount++,this.nextBarStart=t(this.beatsPerBar)}playNote(e,t,i){if(!this.context||!this.masterGain)return;let r=this.context.createOscillator(),a=this.context.createGain();r.type="sine",r.frequency.value=this.noteToFrequency(e),a.gain.setValueAtTime(0,t),a.gain.linearRampToValueAtTime(.7,t+.05),a.gain.exponentialRampToValueAtTime(.001,t+i),r.connect(a),a.connect(this.masterGain),r.start(t),r.stop(t+i+.01),r.onended=()=>{r.disconnect(),a.disconnect()}}};var o=class extends Error{constructor(t,i,r){super(i,r);n(this,"code");this.name="CallError",this.code=t}},C=[{urls:["stun:stun.l.google.com:19302","stun:stun.sipgate.net:10000"]}],f={"http:":"ws:","https:":"wss:","ws:":"ws:","wss:":"wss:"},p=class{constructor({webrtcCode:e,baseUrl:t="https://app.goguava.ai/",connectingTone:i=new d}){n(this,"_state","not-started");n(this,"_error");n(this,"_stateHandlers",new Set);n(this,"_remoteStreamHandlers",new Set);n(this,"_webrtcCode");n(this,"_sockUrl");n(this,"_remoteAudio");n(this,"_connectingTone");n(this,"_sock");n(this,"_localStream");n(this,"_peerConnection");n(this,"_remoteStream");n(this,"_iceServers");n(this,"_pendingStart");this._webrtcCode=e,i!=="disabled"&&(this._connectingTone=i),this._sockUrl=new URL("/webrtc/",t);let r=f[this._sockUrl.protocol];if(r===void 0)throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);this._sockUrl.protocol=r,this._remoteAudio=document.createElement("audio"),this._remoteAudio.autoplay=!0}get state(){return this._state}get error(){return this._error}get remoteStream(){return this._remoteStream}onStateChange(e){return this._stateHandlers.add(e),()=>{this._stateHandlers.delete(e)}}onRemoteStream(e){return this._remoteStreamHandlers.add(e),()=>{this._remoteStreamHandlers.delete(e)}}_setState(e){if(e!==this._state){this._state=e,e==="connecting"?this._connectingTone?.start():this._connectingTone?.stop();for(let t of this._stateHandlers)t(e)}}async start(){if(this._state!=="not-started")throw new Error("start() was already called.");this._setState("starting"),document.body.appendChild(this._remoteAudio);let e=setTimeout(()=>{this._setState("waiting-for-permission")},500);try{this._localStream=await navigator.mediaDevices.getUserMedia({audio:!0,video:!1})}catch(i){let r=new o("MICROPHONE_ACCESS_FAILED","Failed to access microphone.",{cause:i});throw this._end(r),r}finally{clearTimeout(e)}if(this.state==="ended"){this._teardown();return}this._setState("connecting");let t=new Promise((i,r)=>{this._pendingStart={resolve:i,reject:r}});this._sock=new WebSocket(this._sockUrl),this._sock.onopen=()=>{this._send({type:"auth",body:this._webrtcCode})},this._sock.onmessage=i=>{this._onMessage(JSON.parse(i.data))},this._sock.onclose=()=>{this._end(new o("CONNECTION_FAILED","Signaling connection closed."))},await t}async hangup(){if(this._state==="not-started")throw new Error("hangup() called before start().");this._state!=="ended"&&(this._send({type:"hangup"}),this._end())}_send(e){this._sock!==void 0&&this._sock.readyState===WebSocket.OPEN&&this._sock.send(JSON.stringify(e))}_onMessage(e){switch(e.type){case"auth":if(e.body!=="OK"){this._end(new o("AUTHENTICATION_FAILED",e.reason||"Authentication failed."));return}this._iceServers=e.ice_servers,this._send({type:"connect",body:{}}),this._connectPeer();break;case"desc":if(!this._peerConnection){this._end(new o("PROTOCOL_ERROR","Received desc before the peer connection was created."));return}this._peerConnection.setRemoteDescription(e.body).catch(t=>{this._end(new o("CONNECTION_FAILED","Failed to set remote description.",{cause:t}))});break;case"candidate":if(!this._peerConnection){this._end(new o("PROTOCOL_ERROR","Received candidate before the peer connection was created."));return}this._peerConnection.addIceCandidate(e.body).catch(t=>{console.warn("Failed to add ICE candidate:",t)});break;case"hangup":this._state==="connected"?this._end():this._end(new o("REMOTE_HANGUP","The remote hung up before the call connected."));break;case"error":this._end(new o("SERVER_ERROR",e.body||"Unknown server error."));break;case"transfer":this._end(new o("PROTOCOL_ERROR","Transfers are not supported."));break;default:console.warn("Unknown signaling message:",e)}}async _connectPeer(){if(this._localStream){this._peerConnection=new RTCPeerConnection({iceServers:this._iceServers??C}),this._peerConnection.onicecandidate=e=>{e.candidate&&this._send({type:"candidate",body:e.candidate})},this._peerConnection.ontrack=e=>{this._remoteStream=new MediaStream([e.track]),this._remoteAudio.srcObject=this._remoteStream;for(let t of this._remoteStreamHandlers)t(this._remoteStream)},this._peerConnection.onconnectionstatechange=()=>{this._peerConnection!==void 0&&(this._peerConnection.connectionState==="connected"?(this._setState("connected"),this._pendingStart!==void 0&&(this._pendingStart.resolve(),this._pendingStart=void 0)):this._peerConnection.connectionState==="failed"&&this._end(new o("CONNECTION_FAILED","Peer connection failed.")))};for(let e of this._localStream.getAudioTracks())this._peerConnection.addTrack(e);try{await this._peerConnection.setLocalDescription()}catch(e){this._end(new o("CONNECTION_FAILED","Failed to create offer.",{cause:e}));return}this.state!=="ended"&&this._send({type:"desc",body:this._peerConnection.localDescription})}}_end(e){this._state!=="ended"&&(this._error=e,this._teardown(),this._setState("ended"),this._pendingStart!==void 0&&(e?this._pendingStart.reject(e):this._pendingStart.resolve(),this._pendingStart=void 0))}_teardown(){this._sock&&(this._sock.onclose=null,this._sock.close(),this._sock=void 0),this._peerConnection!==void 0&&(this._peerConnection.close(),this._peerConnection=void 0);for(let e of this._localStream?.getTracks()??[])e.stop();this._localStream=void 0,this._remoteStream=void 0,this._remoteAudio.srcObject=null,this._remoteAudio.remove()}};export{p as Call,o as CallError,d as JingleTone,h as RingbackTone};
+//# sourceMappingURL=guava-webrtc.esm.min.js.map

package/dist/guava-webrtc.esm.min.js.map

@@ -0,0 +1,7 @@
+{
+  "version": 3,
+  "sources": ["../src/webrtc/tones.ts", "../src/webrtc/index.ts"],
+  "sourcesContent": ["// Seconds from the start of bar 0 to the given 0-indexed beat.\nfunction beatTime(beat: number, bpm: number): number {\n  return beat * (60 / bpm);\n}\n\nexport interface ConnectingTone {\n  start(): void;\n  stop(): void;\n}\n\n// North American ringback: 440 + 480 Hz, 2s on, 4s off.\nexport class RingbackTone implements ConnectingTone {\n  private context: AudioContext | null = null;\n  private intervalId: ReturnType<typeof setInterval> | null = null;\n\n  private readonly volume = 0.15;\n  private readonly onSeconds = 2;\n  private readonly periodSeconds = 6;\n  // Delay before the first ring, so near-instant connections stay silent.\n  private readonly startDelay = 0.3;\n\n  start(): void {\n    if (this.context) return;\n\n    const context = new AudioContext();\n    const gain = context.createGain();\n    gain.gain.value = 0;\n    gain.connect(context.destination);\n\n    for (const frequency of [440, 480]) {\n      const oscillator = context.createOscillator();\n      oscillator.frequency.value = frequency;\n      oscillator.connect(gain);\n      oscillator.start();\n    }\n\n    let nextRing = context.currentTime + this.startDelay;\n    const ring = () => {\n      gain.gain.setValueAtTime(this.volume, nextRing);\n      gain.gain.setValueAtTime(0, nextRing + this.onSeconds);\n      nextRing += this.periodSeconds;\n    };\n    ring();\n    this.intervalId = setInterval(ring, this.periodSeconds * 1000);\n    this.context = context;\n  }\n\n  stop(): void {\n    if (!this.context) return;\n\n    if (this.intervalId) {\n      clearInterval(this.intervalId);\n      this.intervalId = null;\n    }\n    // Closing the context stops the oscillators.\n    void this.context.close();\n    this.context = null;\n  }\n}\n\nexport class JingleTone implements ConnectingTone {\n  private context: AudioContext | null = null;\n  private masterGain: GainNode | null = null;\n  private intervalId: ReturnType<typeof setInterval> | null = null;\n  private running = false;\n  private pulseCount = 0;\n  // Audio-clock time of the next bar to schedule.\n  private nextBarStart = 0;\n\n  // Configuration\n  private readonly volume = 0.25;\n  private readonly bpm = 120;\n  private readonly beatsPerBar = 4;\n  // Delay before the first note, so near-instant connections stay silent.\n  private readonly startDelay = 0.3;\n\n  start(): void {\n    if (this.running) return;\n\n    this.running = true;\n    this.pulseCount = 0;\n    this.context = new AudioContext();\n    this.masterGain = this.context.createGain();\n    this.masterGain.gain.value = this.volume;\n    this.masterGain.connect(this.context.destination);\n    this.nextBarStart = this.context.currentTime + this.startDelay;\n\n    // Stay one bar ahead so timer jitter doesn't shift the beat.\n    this.playPulse();\n    this.playPulse();\n\n    const barLength = beatTime(this.beatsPerBar, this.bpm);\n    this.intervalId = setInterval(() => {\n      this.playPulse();\n    }, barLength * 1000);\n  }\n\n  private noteToFrequency(note: string): number {\n    const match = note.match(/^([A-G])(#|b)?(-?\\d+)$/);\n\n    if (!match) {\n      throw new Error(`Invalid note: ${note}`);\n    }\n\n    const [, name, accidental, octave] = match;\n\n    const semitones: Record<string, number> = {\n      C: 0,\n      D: 2,\n      E: 4,\n      F: 5,\n      G: 7,\n      A: 9,\n      B: 11,\n    };\n\n    const offset = accidental === \"#\" ? 1 : accidental === \"b\" ? -1 : 0;\n    const midi = (Number(octave) + 1) * 12 + semitones[name] + offset;\n\n    return 440 * Math.pow(2, (midi - 69) / 12);\n  }\n\n  stop(): void {\n    if (!this.running) return;\n\n    this.running = false;\n\n    if (this.intervalId) {\n      clearInterval(this.intervalId);\n      this.intervalId = null;\n    }\n\n    const context = this.context;\n    const gain = this.masterGain;\n\n    if (!context || !gain) return;\n\n    // Smooth fade-out\n    const now = context.currentTime;\n    gain.gain.setValueAtTime(gain.gain.value, now);\n    gain.gain.linearRampToValueAtTime(0, now + 0.15);\n\n    setTimeout(() => {\n      void context.close();\n    }, 200);\n\n    this.context = null;\n    this.masterGain = null;\n  }\n\n  private playPulse(): void {\n    if (!this.context || !this.masterGain) return;\n\n    const bar = this.nextBarStart;\n    const beat = (n: number) => bar + beatTime(n, this.bpm);\n    const beatLength = beatTime(1, this.bpm);\n\n    // Notes on beats 1-3, silence on beat 4.\n    if (this.pulseCount % 4 === 0 || this.pulseCount % 4 === 2) {\n      this.playNote(\"C5\", beat(0), beatLength);\n      this.playNote(\"E5\", beat(0.5), beatLength);\n      this.playNote(\"G4\", beat(1), beatLength);\n    } else if (this.pulseCount % 4 === 3) {\n      this.playNote(\"E5\", beat(0), beatLength);\n      this.playNote(\"G5\", beat(0.5), beatLength);\n      this.playNote(\"C5\", beat(1), beatLength);\n    } else {\n      this.playNote(\"E5\", beat(0), beatLength);\n      this.playNote(\"G4\", beat(0.5), beatLength);\n      this.playNote(\"C5\", beat(1), beatLength);\n    }\n    this.pulseCount++;\n    this.nextBarStart = beat(this.beatsPerBar);\n  }\n\n  private playNote(note: string, startTime: number, duration: number): void {\n    if (!this.context || !this.masterGain) return;\n\n    const oscillator = this.context.createOscillator();\n    const envelope = this.context.createGain();\n\n    oscillator.type = \"sine\";\n    oscillator.frequency.value = this.noteToFrequency(note);\n\n    envelope.gain.setValueAtTime(0, startTime);\n    envelope.gain.linearRampToValueAtTime(0.7, startTime + 0.05);\n    envelope.gain.exponentialRampToValueAtTime(0.001, startTime + duration);\n\n    oscillator.connect(envelope);\n    envelope.connect(this.masterGain);\n\n    oscillator.start(startTime);\n    oscillator.stop(startTime + duration + 0.01);\n\n    oscillator.onended = () => {\n      oscillator.disconnect();\n      envelope.disconnect();\n    };\n  }\n}\n", "import { type ConnectingTone, JingleTone } from \"./tones.ts\";\n\nexport { type ConnectingTone, JingleTone, RingbackTone } from \"./tones.ts\";\n\nexport type CallState =\n  | \"not-started\" // start() hasn't been called yet.\n  | \"starting\" // Starting the call. Do not play a Ringback yet.\n  | \"waiting-for-permission\" // We're waiting for the Microphone permission prompt. You may show a reminder UI here.\n  | \"connecting\" // The call is now connection to the remote. This is where you can play a ringback or other connecting tone.\n  | \"connected\" // The call is now connected and the audio channel is established.\n  | \"ended\"; // The call has ended.\n\nexport type CallErrorCode =\n  | \"MICROPHONE_ACCESS_FAILED\"\n  | \"CONNECTION_FAILED\"\n  | \"AUTHENTICATION_FAILED\"\n  | \"PROTOCOL_ERROR\" // The server sent an unexpected message.\n  | \"SERVER_ERROR\" // The server sent an error message.\n  | \"REMOTE_HANGUP\"; // The remote hung up before the call connected.\n\nexport class CallError extends Error {\n  readonly code: CallErrorCode;\n\n  constructor(code: CallErrorCode, message: string, options?: ErrorOptions) {\n    super(message, options);\n    this.name = \"CallError\";\n    this.code = code;\n  }\n}\n\n// Used if the server doesn't send ICE servers with the auth reply.\nconst DEFAULT_ICE_SERVERS: RTCIceServer[] = [\n  { urls: [\"stun:stun.l.google.com:19302\", \"stun:stun.sipgate.net:10000\"] },\n];\n\nconst SOCKET_PROTOCOLS: Record<string, string> = {\n  \"http:\": \"ws:\",\n  \"https:\": \"wss:\",\n  \"ws:\": \"ws:\",\n  \"wss:\": \"wss:\",\n};\n\n// Messages received on the signaling socket. See junkswitch/junkswitch/webrtcsocket.py.\ntype SignalingMessage =\n  | {\n      type: \"auth\";\n      body: \"OK\" | \"FAIL\";\n      reason?: string;\n      ice_servers?: RTCIceServer[];\n    }\n  | { type: \"desc\"; body: RTCSessionDescriptionInit }\n  | { type: \"candidate\"; body: RTCIceCandidateInit }\n  | { type: \"hangup\" }\n  | { type: \"error\"; body?: string }\n  | { type: \"transfer\" };\n\nexport class Call {\n  private _state: CallState = \"not-started\";\n  private _error?: CallError;\n  private readonly _stateHandlers = new Set<(state: CallState) => void>();\n  private readonly _remoteStreamHandlers = new Set<(stream: MediaStream) => void>();\n  private readonly _webrtcCode: string;\n  private readonly _sockUrl: URL;\n  private readonly _remoteAudio: HTMLAudioElement;\n  private readonly _connectingTone?: ConnectingTone;\n  private _sock?: WebSocket;\n  private _localStream?: MediaStream;\n  private _peerConnection?: RTCPeerConnection;\n  private _remoteStream?: MediaStream;\n  private _iceServers?: RTCIceServer[];\n  // Settles the promise returned by start().\n  private _pendingStart?: { resolve: () => void; reject: (error: CallError) => void };\n\n  constructor({\n    webrtcCode,\n    baseUrl = \"https://app.goguava.ai/\",\n    connectingTone = new JingleTone(),\n  }: {\n    webrtcCode: string;\n    baseUrl?: string;\n    // Played while the call is in the \"connecting\" state.\n    connectingTone?: ConnectingTone | \"disabled\";\n  }) {\n    this._webrtcCode = webrtcCode;\n    if (connectingTone !== \"disabled\") {\n      this._connectingTone = connectingTone;\n    }\n\n    this._sockUrl = new URL(\"/webrtc/\", baseUrl);\n    const sockProtocol = SOCKET_PROTOCOLS[this._sockUrl.protocol];\n    if (sockProtocol === undefined) {\n      throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);\n    }\n    this._sockUrl.protocol = sockProtocol;\n\n    this._remoteAudio = document.createElement(\"audio\");\n    this._remoteAudio.autoplay = true;\n  }\n\n  get state(): CallState {\n    return this._state;\n  }\n\n  // Set if the call ended due to a failure.\n  get error(): CallError | undefined {\n    return this._error;\n  }\n\n  // Set once the remote audio track arrives, which happens before the call reaches \"connected\".\n  get remoteStream(): MediaStream | undefined {\n    return this._remoteStream;\n  }\n\n  // Returns a function that unregisters the handler.\n  onStateChange(handler: (state: CallState) => void): () => void {\n    this._stateHandlers.add(handler);\n    return () => {\n      this._stateHandlers.delete(handler);\n    };\n  }\n\n  // Fires whenever the remote stream is replaced, including the first time it arrives.\n  // Returns a function that unregisters the handler.\n  onRemoteStream(handler: (stream: MediaStream) => void): () => void {\n    this._remoteStreamHandlers.add(handler);\n    return () => {\n      this._remoteStreamHandlers.delete(handler);\n    };\n  }\n\n  private _setState(state: CallState) {\n    if (state === this._state) {\n      return;\n    }\n    this._state = state;\n    if (state === \"connecting\") {\n      this._connectingTone?.start();\n    } else {\n      this._connectingTone?.stop();\n    }\n    for (const handler of this._stateHandlers) {\n      handler(state);\n    }\n  }\n\n  async start() {\n    if (this._state !== \"not-started\") {\n      throw new Error(`start() was already called.`);\n    }\n    this._setState(\"starting\");\n\n    // Add the audio element.\n    document.body.appendChild(this._remoteAudio);\n\n    // We start by requesting device permissions first. The idea is that, after permission has been granted,\n    // it makes sence to play a ringback, but it doesn't make sense to do so during the permission prompt.\n\n    const permissionTimer = setTimeout(() => {\n      this._setState(\"waiting-for-permission\");\n    }, 500);\n\n    try {\n      this._localStream = await navigator.mediaDevices.getUserMedia({ audio: true, video: false });\n    } catch (error) {\n      const callError = new CallError(\"MICROPHONE_ACCESS_FAILED\", \"Failed to access microphone.\", {\n        cause: error,\n      });\n      this._end(callError);\n      throw callError;\n    } finally {\n      clearTimeout(permissionTimer);\n    }\n\n    // hangup() was called during the permission prompt.\n    if (this.state === \"ended\") {\n      this._teardown();\n      return;\n    }\n\n    // Now open and authenticate the signaling socket.\n    this._setState(\"connecting\"); // UI can start playing rinback tones here.\n    const started = new Promise<void>((resolve, reject) => {\n      this._pendingStart = { resolve, reject };\n    });\n\n    this._sock = new WebSocket(this._sockUrl);\n    this._sock.onopen = () => {\n      this._send({ type: \"auth\", body: this._webrtcCode });\n    };\n    this._sock.onmessage = (event) => {\n      this._onMessage(JSON.parse(event.data) as SignalingMessage);\n    };\n    this._sock.onclose = () => {\n      this._end(new CallError(\"CONNECTION_FAILED\", \"Signaling connection closed.\"));\n    };\n\n    await started;\n  }\n\n  async hangup() {\n    if (this._state === \"not-started\") {\n      throw new Error(`hangup() called before start().`);\n    }\n    if (this._state === \"ended\") {\n      return;\n    }\n    this._send({ type: \"hangup\" });\n    this._end();\n  }\n\n  private _send(message: { type: string; body?: unknown }) {\n    if (this._sock !== undefined && this._sock.readyState === WebSocket.OPEN) {\n      this._sock.send(JSON.stringify(message));\n    }\n  }\n\n  private _onMessage(message: SignalingMessage) {\n    switch (message.type) {\n      case \"auth\":\n        if (message.body !== \"OK\") {\n          this._end(\n            new CallError(\"AUTHENTICATION_FAILED\", message.reason || \"Authentication failed.\"),\n          );\n          return;\n        }\n        this._iceServers = message.ice_servers;\n        this._send({ type: \"connect\", body: {} });\n        void this._connectPeer();\n        break;\n      case \"desc\":\n        if (!this._peerConnection) {\n          this._end(\n            new CallError(\n              \"PROTOCOL_ERROR\",\n              \"Received desc before the peer connection was created.\",\n            ),\n          );\n          return;\n        }\n        this._peerConnection.setRemoteDescription(message.body).catch((error) => {\n          this._end(\n            new CallError(\"CONNECTION_FAILED\", \"Failed to set remote description.\", {\n              cause: error,\n            }),\n          );\n        });\n        break;\n      case \"candidate\":\n        if (!this._peerConnection) {\n          this._end(\n            new CallError(\n              \"PROTOCOL_ERROR\",\n              \"Received candidate before the peer connection was created.\",\n            ),\n          );\n          return;\n        }\n        this._peerConnection.addIceCandidate(message.body).catch((error) => {\n          console.warn(\"Failed to add ICE candidate:\", error);\n        });\n        break;\n      case \"hangup\":\n        if (this._state === \"connected\") {\n          this._end();\n        } else {\n          this._end(\n            new CallError(\"REMOTE_HANGUP\", \"The remote hung up before the call connected.\"),\n          );\n        }\n        break;\n      case \"error\":\n        this._end(new CallError(\"SERVER_ERROR\", message.body || \"Unknown server error.\"));\n        break;\n      case \"transfer\":\n        this._end(new CallError(\"PROTOCOL_ERROR\", \"Transfers are not supported.\"));\n        break;\n      default:\n        console.warn(\"Unknown signaling message:\", message);\n    }\n  }\n\n  private async _connectPeer() {\n    if (!this._localStream) {\n      return;\n    }\n    this._peerConnection = new RTCPeerConnection({\n      iceServers: this._iceServers ?? DEFAULT_ICE_SERVERS,\n    });\n    this._peerConnection.onicecandidate = (event) => {\n      if (event.candidate) {\n        this._send({ type: \"candidate\", body: event.candidate });\n      }\n    };\n    this._peerConnection.ontrack = (event) => {\n      this._remoteStream = new MediaStream([event.track]);\n      this._remoteAudio.srcObject = this._remoteStream;\n      for (const handler of this._remoteStreamHandlers) {\n        handler(this._remoteStream);\n      }\n    };\n    this._peerConnection.onconnectionstatechange = () => {\n      if (this._peerConnection === undefined) {\n        return;\n      }\n      // \"disconnected\" is often transient, so wait for \"failed\" before ending the call.\n      if (this._peerConnection.connectionState === \"connected\") {\n        this._setState(\"connected\");\n        if (this._pendingStart !== undefined) {\n          this._pendingStart.resolve();\n          this._pendingStart = undefined;\n        }\n      } else if (this._peerConnection.connectionState === \"failed\") {\n        this._end(new CallError(\"CONNECTION_FAILED\", \"Peer connection failed.\"));\n      }\n    };\n    for (const track of this._localStream.getAudioTracks()) {\n      this._peerConnection.addTrack(track);\n    }\n\n    try {\n      await this._peerConnection.setLocalDescription();\n    } catch (error) {\n      this._end(new CallError(\"CONNECTION_FAILED\", \"Failed to create offer.\", { cause: error }));\n      return;\n    }\n\n    // The call may have ended while creating the offer.\n    if (this.state === \"ended\") {\n      return;\n    }\n    this._send({ type: \"desc\", body: this._peerConnection.localDescription });\n  }\n\n  private _end(error?: CallError) {\n    if (this._state === \"ended\") {\n      return;\n    }\n    this._error = error;\n    this._teardown();\n    this._setState(\"ended\");\n\n    // If start() is still pending, it rejects on failure and resolves on hangup().\n    if (this._pendingStart !== undefined) {\n      if (error) {\n        this._pendingStart.reject(error);\n      } else {\n        this._pendingStart.resolve();\n      }\n      this._pendingStart = undefined;\n    }\n  }\n\n  private _teardown() {\n    if (this._sock) {\n      // Don't fire close handlers for a socket we closed ourselves.\n      this._sock.onclose = null;\n      this._sock.close();\n      this._sock = undefined;\n    }\n    if (this._peerConnection !== undefined) {\n      this._peerConnection.close();\n      this._peerConnection = undefined;\n    }\n    for (const track of this._localStream?.getTracks() ?? []) {\n      track.stop();\n    }\n    this._localStream = undefined;\n    this._remoteStream = undefined;\n    this._remoteAudio.srcObject = null;\n    this._remoteAudio.remove();\n  }\n}\n"],
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+  "names": ["beatTime", "beat", "bpm", "RingbackTone", "__publicField", "context", "gain", "frequency", "oscillator", "nextRing", "ring", "JingleTone", "barLength", "note", "match", "name", "accidental", "octave", "semitones", "offset", "midi", "now", "bar", "n", "beatLength", "startTime", "duration", "envelope", "CallError", "code", "message", "options", "__publicField", "DEFAULT_ICE_SERVERS", "SOCKET_PROTOCOLS", "Call", "webrtcCode", "baseUrl", "connectingTone", "JingleTone", "sockProtocol", "handler", "state", "permissionTimer", "error", "callError", "started", "resolve", "reject", "event", "track"]
+}

package/dist/guava-webrtc.min.js

@@ -0,0 +1,2 @@
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d=e.createOscillator();d.frequency.value=a,d.connect(t),d.start()}let n=e.currentTime+this.startDelay,r=()=>{t.gain.setValueAtTime(this.volume,n),t.gain.setValueAtTime(0,n+this.onSeconds),n+=this.periodSeconds};r(),this.intervalId=setInterval(r,this.periodSeconds*1e3),this.context=e}stop(){this.context&&(this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null),this.context.close(),this.context=null)}},c=class{constructor(){i(this,"context",null);i(this,"masterGain",null);i(this,"intervalId",null);i(this,"running",!1);i(this,"pulseCount",0);i(this,"nextBarStart",0);i(this,"volume",.25);i(this,"bpm",120);i(this,"beatsPerBar",4);i(this,"startDelay",.3)}start(){if(this.running)return;this.running=!0,this.pulseCount=0,this.context=new AudioContext,this.masterGain=this.context.createGain(),this.masterGain.gain.value=this.volume,this.masterGain.connect(this.context.destination),this.nextBarStart=this.context.currentTime+this.startDelay,this.playPulse(),this.playPulse();let e=p(this.beatsPerBar,this.bpm);this.intervalId=setInterval(()=>{this.playPulse()},e*1e3)}noteToFrequency(e){let t=e.match(/^([A-G])(#|b)?(-?\d+)$/);if(!t)throw new Error(`Invalid note: ${e}`);let[,n,r,a]=t,d={C:0,D:2,E:4,F:5,G:7,A:9,B:11},_=r==="#"?1:r==="b"?-1:0,m=(Number(a)+1)*12+d[n]+_;return 440*Math.pow(2,(m-69)/12)}stop(){if(!this.running)return;this.running=!1,this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null);let e=this.context,t=this.masterGain;if(!e||!t)return;let n=e.currentTime;t.gain.setValueAtTime(t.gain.value,n),t.gain.linearRampToValueAtTime(0,n+.15),setTimeout(()=>{e.close()},200),this.context=null,this.masterGain=null}playPulse(){if(!this.context||!this.masterGain)return;let e=this.nextBarStart,t=r=>e+p(r,this.bpm),n=p(1,this.bpm);this.pulseCount%4===0||this.pulseCount%4===2?(this.playNote("C5",t(0),n),this.playNote("E5",t(.5),n),this.playNote("G4",t(1),n)):this.pulseCount%4===3?(this.playNote("E5",t(0),n),this.playNote("G5",t(.5),n),this.playNote("C5",t(1),n)):(this.playNote("E5",t(0),n),this.playNote("G4",t(.5),n),this.playNote("C5",t(1),n)),this.pulseCount++,this.nextBarStart=t(this.beatsPerBar)}playNote(e,t,n){if(!this.context||!this.masterGain)return;let r=this.context.createOscillator(),a=this.context.createGain();r.type="sine",r.frequency.value=this.noteToFrequency(e),a.gain.setValueAtTime(0,t),a.gain.linearRampToValueAtTime(.7,t+.05),a.gain.exponentialRampToValueAtTime(.001,t+n),r.connect(a),a.connect(this.masterGain),r.start(t),r.stop(t+n+.01),r.onended=()=>{r.disconnect(),a.disconnect()}}};var s=class extends Error{constructor(t,n,r){super(n,r);i(this,"code");this.name="CallError",this.code=t}},b=[{urls:["stun:stun.l.google.com:19302","stun:stun.sipgate.net:10000"]}],E={"http:":"ws:","https:":"wss:","ws:":"ws:","wss:":"wss:"},u=class{constructor({webrtcCode:e,baseUrl:t="https://app.goguava.ai/",connectingTone:n=new c}){i(this,"_state","not-started");i(this,"_error");i(this,"_stateHandlers",new Set);i(this,"_remoteStreamHandlers",new Set);i(this,"_webrtcCode");i(this,"_sockUrl");i(this,"_remoteAudio");i(this,"_connectingTone");i(this,"_sock");i(this,"_localStream");i(this,"_peerConnection");i(this,"_remoteStream");i(this,"_iceServers");i(this,"_pendingStart");this._webrtcCode=e,n!=="disabled"&&(this._connectingTone=n),this._sockUrl=new URL("/webrtc/",t);let r=E[this._sockUrl.protocol];if(r===void 0)throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);this._sockUrl.protocol=r,this._remoteAudio=document.createElement("audio"),this._remoteAudio.autoplay=!0}get state(){return this._state}get error(){return this._error}get remoteStream(){return this._remoteStream}onStateChange(e){return this._stateHandlers.add(e),()=>{this._stateHandlers.delete(e)}}onRemoteStream(e){return this._remoteStreamHandlers.add(e),()=>{this._remoteStreamHandlers.delete(e)}}_setState(e){if(e!==this._state){this._state=e,e==="connecting"?this._connectingTone?.start():this._connectingTone?.stop();for(let t of this._stateHandlers)t(e)}}async start(){if(this._state!=="not-started")throw new Error("start() was already called.");this._setState("starting"),document.body.appendChild(this._remoteAudio);let e=setTimeout(()=>{this._setState("waiting-for-permission")},500);try{this._localStream=await navigator.mediaDevices.getUserMedia({audio:!0,video:!1})}catch(n){let r=new s("MICROPHONE_ACCESS_FAILED","Failed to access microphone.",{cause:n});throw this._end(r),r}finally{clearTimeout(e)}if(this.state==="ended"){this._teardown();return}this._setState("connecting");let t=new Promise((n,r)=>{this._pendingStart={resolve:n,reject:r}});this._sock=new WebSocket(this._sockUrl),this._sock.onopen=()=>{this._send({type:"auth",body:this._webrtcCode})},this._sock.onmessage=n=>{this._onMessage(JSON.parse(n.data))},this._sock.onclose=()=>{this._end(new s("CONNECTION_FAILED","Signaling connection closed."))},await t}async hangup(){if(this._state==="not-started")throw new Error("hangup() called before start().");this._state!=="ended"&&(this._send({type:"hangup"}),this._end())}_send(e){this._sock!==void 0&&this._sock.readyState===WebSocket.OPEN&&this._sock.send(JSON.stringify(e))}_onMessage(e){switch(e.type){case"auth":if(e.body!=="OK"){this._end(new s("AUTHENTICATION_FAILED",e.reason||"Authentication failed."));return}this._iceServers=e.ice_servers,this._send({type:"connect",body:{}}),this._connectPeer();break;case"desc":if(!this._peerConnection){this._end(new s("PROTOCOL_ERROR","Received desc before the peer connection was created."));return}this._peerConnection.setRemoteDescription(e.body).catch(t=>{this._end(new s("CONNECTION_FAILED","Failed to set remote description.",{cause:t}))});break;case"candidate":if(!this._peerConnection){this._end(new s("PROTOCOL_ERROR","Received candidate before the peer connection was created."));return}this._peerConnection.addIceCandidate(e.body).catch(t=>{console.warn("Failed to add ICE candidate:",t)});break;case"hangup":this._state==="connected"?this._end():this._end(new s("REMOTE_HANGUP","The remote hung up before the call connected."));break;case"error":this._end(new s("SERVER_ERROR",e.body||"Unknown server error."));break;case"transfer":this._end(new s("PROTOCOL_ERROR","Transfers are not supported."));break;default:console.warn("Unknown signaling message:",e)}}async _connectPeer(){if(this._localStream){this._peerConnection=new RTCPeerConnection({iceServers:this._iceServers??b}),this._peerConnection.onicecandidate=e=>{e.candidate&&this._send({type:"candidate",body:e.candidate})},this._peerConnection.ontrack=e=>{this._remoteStream=new MediaStream([e.track]),this._remoteAudio.srcObject=this._remoteStream;for(let t of this._remoteStreamHandlers)t(this._remoteStream)},this._peerConnection.onconnectionstatechange=()=>{this._peerConnection!==void 0&&(this._peerConnection.connectionState==="connected"?(this._setState("connected"),this._pendingStart!==void 0&&(this._pendingStart.resolve(),this._pendingStart=void 0)):this._peerConnection.connectionState==="failed"&&this._end(new s("CONNECTION_FAILED","Peer connection failed.")))};for(let e of this._localStream.getAudioTracks())this._peerConnection.addTrack(e);try{await this._peerConnection.setLocalDescription()}catch(e){this._end(new s("CONNECTION_FAILED","Failed to create offer.",{cause:e}));return}this.state!=="ended"&&this._send({type:"desc",body:this._peerConnection.localDescription})}}_end(e){this._state!=="ended"&&(this._error=e,this._teardown(),this._setState("ended"),this._pendingStart!==void 0&&(e?this._pendingStart.reject(e):this._pendingStart.resolve(),this._pendingStart=void 0))}_teardown(){this._sock&&(this._sock.onclose=null,this._sock.close(),this._sock=void 0),this._peerConnection!==void 0&&(this._peerConnection.close(),this._peerConnection=void 0);for(let e of this._localStream?.getTracks()??[])e.stop();this._localStream=void 0,this._remoteStream=void 0,this._remoteAudio.srcObject=null,this._remoteAudio.remove()}};return T(w);})();
+//# sourceMappingURL=guava-webrtc.min.js.map

package/dist/guava-webrtc.min.js.map

@@ -0,0 +1,7 @@
+{
+  "version": 3,
+  "sources": ["../src/webrtc/index.ts", "../src/webrtc/tones.ts"],
+  "sourcesContent": ["import { type ConnectingTone, JingleTone } from \"./tones.ts\";\n\nexport { type ConnectingTone, JingleTone, RingbackTone } from \"./tones.ts\";\n\nexport type CallState =\n  | \"not-started\" // start() hasn't been called yet.\n  | \"starting\" // Starting the call. Do not play a Ringback yet.\n  | \"waiting-for-permission\" // We're waiting for the Microphone permission prompt. You may show a reminder UI here.\n  | \"connecting\" // The call is now connection to the remote. This is where you can play a ringback or other connecting tone.\n  | \"connected\" // The call is now connected and the audio channel is established.\n  | \"ended\"; // The call has ended.\n\nexport type CallErrorCode =\n  | \"MICROPHONE_ACCESS_FAILED\"\n  | \"CONNECTION_FAILED\"\n  | \"AUTHENTICATION_FAILED\"\n  | \"PROTOCOL_ERROR\" // The server sent an unexpected message.\n  | \"SERVER_ERROR\" // The server sent an error message.\n  | \"REMOTE_HANGUP\"; // The remote hung up before the call connected.\n\nexport class CallError extends Error {\n  readonly code: CallErrorCode;\n\n  constructor(code: CallErrorCode, message: string, options?: ErrorOptions) {\n    super(message, options);\n    this.name = \"CallError\";\n    this.code = code;\n  }\n}\n\n// Used if the server doesn't send ICE servers with the auth reply.\nconst DEFAULT_ICE_SERVERS: RTCIceServer[] = [\n  { urls: [\"stun:stun.l.google.com:19302\", \"stun:stun.sipgate.net:10000\"] },\n];\n\nconst SOCKET_PROTOCOLS: Record<string, string> = {\n  \"http:\": \"ws:\",\n  \"https:\": \"wss:\",\n  \"ws:\": \"ws:\",\n  \"wss:\": \"wss:\",\n};\n\n// Messages received on the signaling socket. See junkswitch/junkswitch/webrtcsocket.py.\ntype SignalingMessage =\n  | {\n      type: \"auth\";\n      body: \"OK\" | \"FAIL\";\n      reason?: string;\n      ice_servers?: RTCIceServer[];\n    }\n  | { type: \"desc\"; body: RTCSessionDescriptionInit }\n  | { type: \"candidate\"; body: RTCIceCandidateInit }\n  | { type: \"hangup\" }\n  | { type: \"error\"; body?: string }\n  | { type: \"transfer\" };\n\nexport class Call {\n  private _state: CallState = \"not-started\";\n  private _error?: CallError;\n  private readonly _stateHandlers = new Set<(state: CallState) => void>();\n  private readonly _remoteStreamHandlers = new Set<(stream: MediaStream) => void>();\n  private readonly _webrtcCode: string;\n  private readonly _sockUrl: URL;\n  private readonly _remoteAudio: HTMLAudioElement;\n  private readonly _connectingTone?: ConnectingTone;\n  private _sock?: WebSocket;\n  private _localStream?: MediaStream;\n  private _peerConnection?: RTCPeerConnection;\n  private _remoteStream?: MediaStream;\n  private _iceServers?: RTCIceServer[];\n  // Settles the promise returned by start().\n  private _pendingStart?: { resolve: () => void; reject: (error: CallError) => void };\n\n  constructor({\n    webrtcCode,\n    baseUrl = \"https://app.goguava.ai/\",\n    connectingTone = new JingleTone(),\n  }: {\n    webrtcCode: string;\n    baseUrl?: string;\n    // Played while the call is in the \"connecting\" state.\n    connectingTone?: ConnectingTone | \"disabled\";\n  }) {\n    this._webrtcCode = webrtcCode;\n    if (connectingTone !== \"disabled\") {\n      this._connectingTone = connectingTone;\n    }\n\n    this._sockUrl = new URL(\"/webrtc/\", baseUrl);\n    const sockProtocol = SOCKET_PROTOCOLS[this._sockUrl.protocol];\n    if (sockProtocol === undefined) {\n      throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);\n    }\n    this._sockUrl.protocol = sockProtocol;\n\n    this._remoteAudio = document.createElement(\"audio\");\n    this._remoteAudio.autoplay = true;\n  }\n\n  get state(): CallState {\n    return this._state;\n  }\n\n  // Set if the call ended due to a failure.\n  get error(): CallError | undefined {\n    return this._error;\n  }\n\n  // Set once the remote audio track arrives, which happens before the call reaches \"connected\".\n  get remoteStream(): MediaStream | undefined {\n    return this._remoteStream;\n  }\n\n  // Returns a function that unregisters the handler.\n  onStateChange(handler: (state: CallState) => void): () => void {\n    this._stateHandlers.add(handler);\n    return () => {\n      this._stateHandlers.delete(handler);\n    };\n  }\n\n  // Fires whenever the remote stream is replaced, including the first time it arrives.\n  // Returns a function that unregisters the handler.\n  onRemoteStream(handler: (stream: MediaStream) => void): () => void {\n    this._remoteStreamHandlers.add(handler);\n    return () => {\n      this._remoteStreamHandlers.delete(handler);\n    };\n  }\n\n  private _setState(state: CallState) {\n    if (state === this._state) {\n      return;\n    }\n    this._state = state;\n    if (state === \"connecting\") {\n      this._connectingTone?.start();\n    } else {\n      this._connectingTone?.stop();\n    }\n    for (const handler of this._stateHandlers) {\n      handler(state);\n    }\n  }\n\n  async start() {\n    if (this._state !== \"not-started\") {\n      throw new Error(`start() was already called.`);\n    }\n    this._setState(\"starting\");\n\n    // Add the audio element.\n    document.body.appendChild(this._remoteAudio);\n\n    // We start by requesting device permissions first. The idea is that, after permission has been granted,\n    // it makes sence to play a ringback, but it doesn't make sense to do so during the permission prompt.\n\n    const permissionTimer = setTimeout(() => {\n      this._setState(\"waiting-for-permission\");\n    }, 500);\n\n    try {\n      this._localStream = await navigator.mediaDevices.getUserMedia({ audio: true, video: false });\n    } catch (error) {\n      const callError = new CallError(\"MICROPHONE_ACCESS_FAILED\", \"Failed to access microphone.\", {\n        cause: error,\n      });\n      this._end(callError);\n      throw callError;\n    } finally {\n      clearTimeout(permissionTimer);\n    }\n\n    // hangup() was called during the permission prompt.\n    if (this.state === \"ended\") {\n      this._teardown();\n      return;\n    }\n\n    // Now open and authenticate the signaling socket.\n    this._setState(\"connecting\"); // UI can start playing rinback tones here.\n    const started = new Promise<void>((resolve, reject) => {\n      this._pendingStart = { resolve, reject };\n    });\n\n    this._sock = new WebSocket(this._sockUrl);\n    this._sock.onopen = () => {\n      this._send({ type: \"auth\", body: this._webrtcCode });\n    };\n    this._sock.onmessage = (event) => {\n      this._onMessage(JSON.parse(event.data) as SignalingMessage);\n    };\n    this._sock.onclose = () => {\n      this._end(new CallError(\"CONNECTION_FAILED\", \"Signaling connection closed.\"));\n    };\n\n    await started;\n  }\n\n  async hangup() {\n    if (this._state === \"not-started\") {\n      throw new Error(`hangup() called before start().`);\n    }\n    if (this._state === \"ended\") {\n      return;\n    }\n    this._send({ type: \"hangup\" });\n    this._end();\n  }\n\n  private _send(message: { type: string; body?: unknown }) {\n    if (this._sock !== undefined && this._sock.readyState === WebSocket.OPEN) {\n      this._sock.send(JSON.stringify(message));\n    }\n  }\n\n  private _onMessage(message: SignalingMessage) {\n    switch (message.type) {\n      case \"auth\":\n        if (message.body !== \"OK\") {\n          this._end(\n            new CallError(\"AUTHENTICATION_FAILED\", message.reason || \"Authentication failed.\"),\n          );\n          return;\n        }\n        this._iceServers = message.ice_servers;\n        this._send({ type: \"connect\", body: {} });\n        void this._connectPeer();\n        break;\n      case \"desc\":\n        if (!this._peerConnection) {\n          this._end(\n            new CallError(\n              \"PROTOCOL_ERROR\",\n              \"Received desc before the peer connection was created.\",\n            ),\n          );\n          return;\n        }\n        this._peerConnection.setRemoteDescription(message.body).catch((error) => {\n          this._end(\n            new CallError(\"CONNECTION_FAILED\", \"Failed to set remote description.\", {\n              cause: error,\n            }),\n          );\n        });\n        break;\n      case \"candidate\":\n        if (!this._peerConnection) {\n          this._end(\n            new CallError(\n              \"PROTOCOL_ERROR\",\n              \"Received candidate before the peer connection was created.\",\n            ),\n          );\n          return;\n        }\n        this._peerConnection.addIceCandidate(message.body).catch((error) => {\n          console.warn(\"Failed to add ICE candidate:\", error);\n        });\n        break;\n      case \"hangup\":\n        if (this._state === \"connected\") {\n          this._end();\n        } else {\n          this._end(\n            new CallError(\"REMOTE_HANGUP\", \"The remote hung up before the call connected.\"),\n          );\n        }\n        break;\n      case \"error\":\n        this._end(new CallError(\"SERVER_ERROR\", message.body || \"Unknown server error.\"));\n        break;\n      case \"transfer\":\n        this._end(new CallError(\"PROTOCOL_ERROR\", \"Transfers are not supported.\"));\n        break;\n      default:\n        console.warn(\"Unknown signaling message:\", message);\n    }\n  }\n\n  private async _connectPeer() {\n    if (!this._localStream) {\n      return;\n    }\n    this._peerConnection = new RTCPeerConnection({\n      iceServers: this._iceServers ?? DEFAULT_ICE_SERVERS,\n    });\n    this._peerConnection.onicecandidate = (event) => {\n      if (event.candidate) {\n        this._send({ type: \"candidate\", body: event.candidate });\n      }\n    };\n    this._peerConnection.ontrack = (event) => {\n      this._remoteStream = new MediaStream([event.track]);\n      this._remoteAudio.srcObject = this._remoteStream;\n      for (const handler of this._remoteStreamHandlers) {\n        handler(this._remoteStream);\n      }\n    };\n    this._peerConnection.onconnectionstatechange = () => {\n      if (this._peerConnection === undefined) {\n        return;\n      }\n      // \"disconnected\" is often transient, so wait for \"failed\" before ending the call.\n      if (this._peerConnection.connectionState === \"connected\") {\n        this._setState(\"connected\");\n        if (this._pendingStart !== undefined) {\n          this._pendingStart.resolve();\n          this._pendingStart = undefined;\n        }\n      } else if (this._peerConnection.connectionState === \"failed\") {\n        this._end(new CallError(\"CONNECTION_FAILED\", \"Peer connection failed.\"));\n      }\n    };\n    for (const track of this._localStream.getAudioTracks()) {\n      this._peerConnection.addTrack(track);\n    }\n\n    try {\n      await this._peerConnection.setLocalDescription();\n    } catch (error) {\n      this._end(new CallError(\"CONNECTION_FAILED\", \"Failed to create offer.\", { cause: error }));\n      return;\n    }\n\n    // The call may have ended while creating the offer.\n    if (this.state === \"ended\") {\n      return;\n    }\n    this._send({ type: \"desc\", body: this._peerConnection.localDescription });\n  }\n\n  private _end(error?: CallError) {\n    if (this._state === \"ended\") {\n      return;\n    }\n    this._error = error;\n    this._teardown();\n    this._setState(\"ended\");\n\n    // If start() is still pending, it rejects on failure and resolves on hangup().\n    if (this._pendingStart !== undefined) {\n      if (error) {\n        this._pendingStart.reject(error);\n      } else {\n        this._pendingStart.resolve();\n      }\n      this._pendingStart = undefined;\n    }\n  }\n\n  private _teardown() {\n    if (this._sock) {\n      // Don't fire close handlers for a socket we closed ourselves.\n      this._sock.onclose = null;\n      this._sock.close();\n      this._sock = undefined;\n    }\n    if (this._peerConnection !== undefined) {\n      this._peerConnection.close();\n      this._peerConnection = undefined;\n    }\n    for (const track of this._localStream?.getTracks() ?? []) {\n      track.stop();\n    }\n    this._localStream = undefined;\n    this._remoteStream = undefined;\n    this._remoteAudio.srcObject = null;\n    this._remoteAudio.remove();\n  }\n}\n", "// Seconds from the start of bar 0 to the given 0-indexed beat.\nfunction beatTime(beat: number, bpm: number): number {\n  return beat * (60 / bpm);\n}\n\nexport interface ConnectingTone {\n  start(): void;\n  stop(): void;\n}\n\n// North American ringback: 440 + 480 Hz, 2s on, 4s off.\nexport class RingbackTone implements ConnectingTone {\n  private context: AudioContext | null = null;\n  private intervalId: ReturnType<typeof setInterval> | null = null;\n\n  private readonly volume = 0.15;\n  private readonly onSeconds = 2;\n  private readonly periodSeconds = 6;\n  // Delay before the first ring, so near-instant connections stay silent.\n  private readonly startDelay = 0.3;\n\n  start(): void {\n    if (this.context) return;\n\n    const context = new AudioContext();\n    const gain = context.createGain();\n    gain.gain.value = 0;\n    gain.connect(context.destination);\n\n    for (const frequency of [440, 480]) {\n      const oscillator = context.createOscillator();\n      oscillator.frequency.value = frequency;\n      oscillator.connect(gain);\n      oscillator.start();\n    }\n\n    let nextRing = context.currentTime + this.startDelay;\n    const ring = () => {\n      gain.gain.setValueAtTime(this.volume, nextRing);\n      gain.gain.setValueAtTime(0, nextRing + this.onSeconds);\n      nextRing += this.periodSeconds;\n    };\n    ring();\n    this.intervalId = setInterval(ring, this.periodSeconds * 1000);\n    this.context = context;\n  }\n\n  stop(): void {\n    if (!this.context) return;\n\n    if (this.intervalId) {\n      clearInterval(this.intervalId);\n      this.intervalId = null;\n    }\n    // Closing the context stops the oscillators.\n    void this.context.close();\n    this.context = null;\n  }\n}\n\nexport class JingleTone implements ConnectingTone {\n  private context: AudioContext | null = null;\n  private masterGain: GainNode | null = null;\n  private intervalId: ReturnType<typeof setInterval> | null = null;\n  private running = false;\n  private pulseCount = 0;\n  // Audio-clock time of the next bar to schedule.\n  private nextBarStart = 0;\n\n  // Configuration\n  private readonly volume = 0.25;\n  private readonly bpm = 120;\n  private readonly beatsPerBar = 4;\n  // Delay before the first note, so near-instant connections stay silent.\n  private readonly startDelay = 0.3;\n\n  start(): void {\n    if (this.running) return;\n\n    this.running = true;\n    this.pulseCount = 0;\n    this.context = new AudioContext();\n    this.masterGain = this.context.createGain();\n    this.masterGain.gain.value = this.volume;\n    this.masterGain.connect(this.context.destination);\n    this.nextBarStart = this.context.currentTime + this.startDelay;\n\n    // Stay one bar ahead so timer jitter doesn't shift the beat.\n    this.playPulse();\n    this.playPulse();\n\n    const barLength = beatTime(this.beatsPerBar, this.bpm);\n    this.intervalId = setInterval(() => {\n      this.playPulse();\n    }, barLength * 1000);\n  }\n\n  private noteToFrequency(note: string): number {\n    const match = note.match(/^([A-G])(#|b)?(-?\\d+)$/);\n\n    if (!match) {\n      throw new Error(`Invalid note: ${note}`);\n    }\n\n    const [, name, accidental, octave] = match;\n\n    const semitones: Record<string, number> = {\n      C: 0,\n      D: 2,\n      E: 4,\n      F: 5,\n      G: 7,\n      A: 9,\n      B: 11,\n    };\n\n    const offset = accidental === \"#\" ? 1 : accidental === \"b\" ? -1 : 0;\n    const midi = (Number(octave) + 1) * 12 + semitones[name] + offset;\n\n    return 440 * Math.pow(2, (midi - 69) / 12);\n  }\n\n  stop(): void {\n    if (!this.running) return;\n\n    this.running = false;\n\n    if (this.intervalId) {\n      clearInterval(this.intervalId);\n      this.intervalId = null;\n    }\n\n    const context = this.context;\n    const gain = this.masterGain;\n\n    if (!context || !gain) return;\n\n    // Smooth fade-out\n    const now = context.currentTime;\n    gain.gain.setValueAtTime(gain.gain.value, now);\n    gain.gain.linearRampToValueAtTime(0, now + 0.15);\n\n    setTimeout(() => {\n      void context.close();\n    }, 200);\n\n    this.context = null;\n    this.masterGain = null;\n  }\n\n  private playPulse(): void {\n    if (!this.context || !this.masterGain) return;\n\n    const bar = this.nextBarStart;\n    const beat = (n: number) => bar + beatTime(n, this.bpm);\n    const beatLength = beatTime(1, this.bpm);\n\n    // Notes on beats 1-3, silence on beat 4.\n    if (this.pulseCount % 4 === 0 || this.pulseCount % 4 === 2) {\n      this.playNote(\"C5\", beat(0), beatLength);\n      this.playNote(\"E5\", beat(0.5), beatLength);\n      this.playNote(\"G4\", beat(1), beatLength);\n    } else if (this.pulseCount % 4 === 3) {\n      this.playNote(\"E5\", beat(0), beatLength);\n      this.playNote(\"G5\", beat(0.5), beatLength);\n      this.playNote(\"C5\", beat(1), beatLength);\n    } else {\n      this.playNote(\"E5\", beat(0), beatLength);\n      this.playNote(\"G4\", beat(0.5), beatLength);\n      this.playNote(\"C5\", beat(1), beatLength);\n    }\n    this.pulseCount++;\n    this.nextBarStart = beat(this.beatsPerBar);\n  }\n\n  private playNote(note: string, startTime: number, duration: number): void {\n    if (!this.context || !this.masterGain) return;\n\n    const oscillator = this.context.createOscillator();\n    const envelope = this.context.createGain();\n\n    oscillator.type = \"sine\";\n    oscillator.frequency.value = this.noteToFrequency(note);\n\n    envelope.gain.setValueAtTime(0, startTime);\n    envelope.gain.linearRampToValueAtTime(0.7, startTime + 0.05);\n    envelope.gain.exponentialRampToValueAtTime(0.001, startTime + duration);\n\n    oscillator.connect(envelope);\n    envelope.connect(this.masterGain);\n\n    oscillator.start(startTime);\n    oscillator.stop(startTime + duration + 0.01);\n\n    oscillator.onended = () => {\n      oscillator.disconnect();\n      envelope.disconnect();\n    };\n  }\n}\n"],
+  "mappings": 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+  "names": ["index_exports", "__export", "Call", "CallError", "JingleTone", "RingbackTone", "beatTime", "beat", "bpm", "RingbackTone", "__publicField", "context", "gain", "frequency", "oscillator", "nextRing", "ring", "JingleTone", "barLength", "note", "match", "name", "accidental", "octave", "semitones", "offset", "midi", "now", "bar", "n", "beatLength", "startTime", "duration", "envelope", "CallError", "code", "message", "options", "__publicField", "DEFAULT_ICE_SERVERS", "SOCKET_PROTOCOLS", "Call", "webrtcCode", "baseUrl", "connectingTone", "JingleTone", "sockProtocol", "handler", "state", "permissionTimer", "error", "callError", "started", "resolve", "reject", "event", "track"]
+}

package/dist/src/call.d.ts

@@ -4,6 +4,7 @@ import { type DTMFDigit } from "./events.ts";
 import type * as z from "zod";
 import type { FieldItem, SayItem } from "./action-item.ts";
 import type { CallInfo } from "./socket/call-info.ts";
+import type { SipUri } from "./sip.ts";
 export type TaskObjective = {
     objective: string;
 } | {
@@ -74,21 +75,21 @@ export declare class Call {
      * This method does not interfere with turn-taking — use this when
      * you want the agent to naturally transfer the call.
      *
-     * @param destination - The phone number or SIP address to transfer the call to.
+     * @param destination - The phone number, SIP address, or SipUri to transfer the call to.
      * @param instructions - Optional guidance for how the agent should handle the transfer.
      */
-    transfer(destination: string, instructions?: string): Promise<void>;
+    transfer(destination: string | SipUri, instructions?: string): Promise<void>;
     /**
      * Transfer the call immediately.
      *
      * Unlike transfer(), this does not respect turn-taking. The agent immediately
      * initiates the transfer, saying the script (if provided).
      *
-     * @param destination - The phone number or SIP address to transfer the call to.
+     * @param destination - The phone number, SIP address, or SipUri to transfer the call to.
      * @param finalScript - Optional message for the agent to deliver immediately
      *   before transferring. Defaults to no message.
      */
-    immediateTransfer(destination: string, finalScript?: string): Promise<void>;
+    immediateTransfer(destination: string | SipUri, finalScript?: string): Promise<void>;
     addInfo(label: string, info: unknown): Promise<void>;
     retryTask(reason: string): Promise<void>;
     readScript(script: string): Promise<void>;

package/dist/src/call.js

@@ -213,13 +213,13 @@ let Call = (() => {
          * This method does not interfere with turn-taking — use this when
          * you want the agent to naturally transfer the call.
          *
-         * @param destination - The phone number or SIP address to transfer the call to.
+         * @param destination - The phone number, SIP address, or SipUri to transfer the call to.
          * @param instructions - Optional guidance for how the agent should handle the transfer.
          */
         async transfer(destination, instructions) {
             await this.sendCommand(commands_ts_1.TransferCommand, {
                 command_type: "transfer-call",
-                to_number: destination,
+                to_number: destination.toString(),
                 transfer_message: instructions ?? "Notify the caller that you will be transferring them, and then transfer.",
                 soft_transfer: true,
             });
@@ -230,14 +230,14 @@ let Call = (() => {
          * Unlike transfer(), this does not respect turn-taking. The agent immediately
          * initiates the transfer, saying the script (if provided).
          *
-         * @param destination - The phone number or SIP address to transfer the call to.
+         * @param destination - The phone number, SIP address, or SipUri to transfer the call to.
          * @param finalScript - Optional message for the agent to deliver immediately
          *   before transferring. Defaults to no message.
          */
         async immediateTransfer(destination, finalScript) {
             await this.sendCommand(commands_ts_1.TransferCommand, {
                 command_type: "transfer-call",
-                to_number: destination,
+                to_number: destination.toString(),
                 transfer_message: finalScript ?? "",
                 soft_transfer: false,
             });

package/dist/src/call.js.map

@@ -1 +1 @@
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package/dist/src/index.d.ts

@@ -7,6 +7,7 @@ export { Say, Field } from "./action-item.ts";
 export { Logger, getConsoleLogger, getDefaultLogger } from "./logging.ts";
 export { Agent, CallInfo } from "./agent.ts";
 export { SIPCallInfo } from "./socket/call-info.ts";
+export { SipUri } from "./sip.ts";
 export { Call, DEFAULT_REACH_PERSON_OUTCOMES } from "./call.ts";
 export type { ReachPersonOutcome } from "./call.ts";
 export type { SpeedPreset, Language } from "./commands.ts";

package/dist/src/index.js

@@ -1,6 +1,6 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.MockCall = exports.TestSession = exports.DEFAULT_REACH_PERSON_OUTCOMES = exports.Call = exports.SIPCallInfo = exports.Agent = exports.getDefaultLogger = exports.getConsoleLogger = exports.Field = exports.Say = exports.Campaign = exports.Runner = exports.Client = void 0;
+exports.MockCall = exports.TestSession = exports.DEFAULT_REACH_PERSON_OUTCOMES = exports.Call = exports.SipUri = exports.SIPCallInfo = exports.Agent = exports.getDefaultLogger = exports.getConsoleLogger = exports.Field = exports.Say = exports.Campaign = exports.Runner = exports.Client = void 0;
 var client_ts_1 = require("./client.js");
 Object.defineProperty(exports, "Client", { enumerable: true, get: function () { return client_ts_1.Client; } });
 var runner_ts_1 = require("./runner.js");
@@ -17,6 +17,8 @@ var agent_ts_1 = require("./agent.js");
 Object.defineProperty(exports, "Agent", { enumerable: true, get: function () { return agent_ts_1.Agent; } });
 var call_info_ts_1 = require("./socket/call-info.js");
 Object.defineProperty(exports, "SIPCallInfo", { enumerable: true, get: function () { return call_info_ts_1.SIPCallInfo; } });
+var sip_ts_1 = require("./sip.js");
+Object.defineProperty(exports, "SipUri", { enumerable: true, get: function () { return sip_ts_1.SipUri; } });
 var call_ts_1 = require("./call.js");
 Object.defineProperty(exports, "Call", { enumerable: true, get: function () { return call_ts_1.Call; } });
 Object.defineProperty(exports, "DEFAULT_REACH_PERSON_OUTCOMES", { enumerable: true, get: function () { return call_ts_1.DEFAULT_REACH_PERSON_OUTCOMES; } });

package/dist/src/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/index.ts"],"names":[],"mappings":";;;AAAA,yCAA6D;AAApD,mGAAA,MAAM,OAAA;AACf,yCAAqC;AAA5B,mGAAA,MAAM,OAAA;AAEf,+CAA0C;AAAjC,wGAAA,QAAQ,OAAA;AAQjB,mDAA8C;AAArC,qGAAA,GAAG,OAAA;AAAE,uGAAA,KAAK,OAAA;AACnB,2CAA0E;AAAzD,8GAAA,gBAAgB,OAAA;AAAE,8GAAA,gBAAgB,OAAA;AACnD,uCAA6C;AAApC,iGAAA,KAAK,OAAA;AACd,sDAAoD;AAA3C,2GAAA,WAAW,OAAA;AACpB,qCAAgE;AAAvD,+FAAA,IAAI,OAAA;AAAE,wHAAA,6BAA6B,OAAA;AAW5C,mDAAmD;AAA1C,yGAAA,WAAW,OAAA;AACpB,+CAA8C;AAArC,oGAAA,QAAQ,OAAA"}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/index.ts"],"names":[],"mappings":";;;AAAA,yCAA6D;AAApD,mGAAA,MAAM,OAAA;AACf,yCAAqC;AAA5B,mGAAA,MAAM,OAAA;AAEf,+CAA0C;AAAjC,wGAAA,QAAQ,OAAA;AAQjB,mDAA8C;AAArC,qGAAA,GAAG,OAAA;AAAE,uGAAA,KAAK,OAAA;AACnB,2CAA0E;AAAzD,8GAAA,gBAAgB,OAAA;AAAE,8GAAA,gBAAgB,OAAA;AACnD,uCAA6C;AAApC,iGAAA,KAAK,OAAA;AACd,sDAAoD;AAA3C,2GAAA,WAAW,OAAA;AACpB,mCAAkC;AAAzB,gGAAA,MAAM,OAAA;AACf,qCAAgE;AAAvD,+FAAA,IAAI,OAAA;AAAE,wHAAA,6BAA6B,OAAA;AAW5C,mDAAmD;AAA1C,yGAAA,WAAW,OAAA;AACpB,+CAA8C;AAArC,oGAAA,QAAQ,OAAA"}

package/dist/src/sip.d.ts

@@ -0,0 +1,15 @@
+export declare class SipUri {
+    host: string;
+    user?: string;
+    /** A `null` value renders as a valueless param, e.g. `;lr`. */
+    uriParams: Record<string, string | null>;
+    /** A `null` value renders as a valueless header, e.g. `?X-Foo=`. */
+    headers: Record<string, string | null>;
+    constructor(args: {
+        host: string;
+        user?: string;
+        uriParams?: Record<string, string | null>;
+        headers?: Record<string, string | null>;
+    });
+    toString(): string;
+}

package/dist/src/sip.js

@@ -0,0 +1,52 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.SipUri = void 0;
+// Characters allowed unescaped in each component beyond RFC 3261's `unreserved`,
+// which encodeURIComponent already leaves alone. See RFC 3261 section 25.1.
+const USER_SAFE = "&=+$,;?/";
+const PARAM_SAFE = "[]/:&+$";
+const HEADER_SAFE = "[]/?:+$";
+function sipEscape(value, safe) {
+    return encodeURIComponent(value).replace(/%[0-9A-F]{2}/g, (escaped) => {
+        // Not decodeURIComponent, which throws on single bytes of multibyte UTF-8 sequences.
+        const char = String.fromCharCode(Number.parseInt(escaped.slice(1), 16));
+        return safe.includes(char) ? char : escaped;
+    });
+}
+class SipUri {
+    host;
+    user;
+    /** A `null` value renders as a valueless param, e.g. `;lr`. */
+    uriParams;
+    /** A `null` value renders as a valueless header, e.g. `?X-Foo=`. */
+    headers;
+    constructor(args) {
+        this.host = args.host;
+        this.user = args.user;
+        this.uriParams = args.uriParams ?? {};
+        this.headers = args.headers ?? {};
+    }
+    toString() {
+        let url = "sip:";
+        if (this.user !== undefined) {
+            url += `${sipEscape(this.user, USER_SAFE)}@`;
+        }
+        url += this.host;
+        for (const [name, value] of Object.entries(this.uriParams)) {
+            url += `;${sipEscape(name, PARAM_SAFE)}`;
+            if (value !== null) {
+                url += `=${sipEscape(value, PARAM_SAFE)}`;
+            }
+        }
+        // Headers always need an "=", so a null value renders as "name=".
+        const headers = Object.entries(this.headers);
+        if (headers.length > 0) {
+            url += `?${headers
+                .map(([name, value]) => `${sipEscape(name, HEADER_SAFE)}=${sipEscape(value === null ? "" : String(value), HEADER_SAFE)}`)
+                .join("&")}`;
+        }
+        return url;
+    }
+}
+exports.SipUri = SipUri;
+//# sourceMappingURL=sip.js.map

package/dist/src/sip.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"sip.js","sourceRoot":"","sources":["../../src/sip.ts"],"names":[],"mappings":";;;AAAA,iFAAiF;AACjF,4EAA4E;AAC5E,MAAM,SAAS,GAAG,UAAU,CAAC;AAC7B,MAAM,UAAU,GAAG,SAAS,CAAC;AAC7B,MAAM,WAAW,GAAG,SAAS,CAAC;AAE9B,SAAS,SAAS,CAAC,KAAa,EAAE,IAAY;IAC5C,OAAO,kBAAkB,CAAC,KAAK,CAAC,CAAC,OAAO,CAAC,eAAe,EAAE,CAAC,OAAO,EAAE,EAAE;QACpE,qFAAqF;QACrF,MAAM,IAAI,GAAG,MAAM,CAAC,YAAY,CAAC,MAAM,CAAC,QAAQ,CAAC,OAAO,CAAC,KAAK,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC;QACxE,OAAO,IAAI,CAAC,QAAQ,CAAC,IAAI,CAAC,CAAC,CAAC,CAAC,IAAI,CAAC,CAAC,CAAC,OAAO,CAAC;IAC9C,CAAC,CAAC,CAAC;AACL,CAAC;AAED,MAAa,MAAM;IACjB,IAAI,CAAS;IACb,IAAI,CAAU;IACd,+DAA+D;IAC/D,SAAS,CAAgC;IACzC,oEAAoE;IACpE,OAAO,CAAgC;IAEvC,YAAY,IAKX;QACC,IAAI,CAAC,IAAI,GAAG,IAAI,CAAC,IAAI,CAAC;QACtB,IAAI,CAAC,IAAI,GAAG,IAAI,CAAC,IAAI,CAAC;QACtB,IAAI,CAAC,SAAS,GAAG,IAAI,CAAC,SAAS,IAAI,EAAE,CAAC;QACtC,IAAI,CAAC,OAAO,GAAG,IAAI,CAAC,OAAO,IAAI,EAAE,CAAC;IACpC,CAAC;IAED,QAAQ;QACN,IAAI,GAAG,GAAG,MAAM,CAAC;QACjB,IAAI,IAAI,CAAC,IAAI,KAAK,SAAS,EAAE,CAAC;YAC5B,GAAG,IAAI,GAAG,SAAS,CAAC,IAAI,CAAC,IAAI,EAAE,SAAS,CAAC,GAAG,CAAC;QAC/C,CAAC;QACD,GAAG,IAAI,IAAI,CAAC,IAAI,CAAC;QAEjB,KAAK,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,IAAI,MAAM,CAAC,OAAO,CAAC,IAAI,CAAC,SAAS,CAAC,EAAE,CAAC;YAC3D,GAAG,IAAI,IAAI,SAAS,CAAC,IAAI,EAAE,UAAU,CAAC,EAAE,CAAC;YACzC,IAAI,KAAK,KAAK,IAAI,EAAE,CAAC;gBACnB,GAAG,IAAI,IAAI,SAAS,CAAC,KAAK,EAAE,UAAU,CAAC,EAAE,CAAC;YAC5C,CAAC;QACH,CAAC;QAED,kEAAkE;QAClE,MAAM,OAAO,GAAG,MAAM,CAAC,OAAO,CAAC,IAAI,CAAC,OAAO,CAAC,CAAC;QAC7C,IAAI,OAAO,CAAC,MAAM,GAAG,CAAC,EAAE,CAAC;YACvB,GAAG,IAAI,IAAI,OAAO;iBACf,GAAG,CACF,CAAC,CAAC,IAAI,EAAE,KAAK,CAAC,EAAE,EAAE,CAChB,GAAG,SAAS,CAAC,IAAI,EAAE,WAAW,CAAC,IAAI,SAAS,CAAC,KAAK,KAAK,IAAI,CAAC,CAAC,CAAC,EAAE,CAAC,CAAC,CAAC,MAAM,CAAC,KAAK,CAAC,EAAE,WAAW,CAAC,EAAE,CACnG;iBACA,IAAI,CAAC,GAAG,CAAC,EAAE,CAAC;QACjB,CAAC;QAED,OAAO,GAAG,CAAC;IACb,CAAC;CACF;AA/CD,wBA+CC"}

package/dist/src/version.d.ts

@@ -1 +1 @@
-export declare const SDK_VERSION = "0.37.0";
+export declare const SDK_VERSION = "0.38.0";

package/dist/src/version.js

@@ -1,5 +1,5 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.SDK_VERSION = void 0;
-exports.SDK_VERSION = "0.37.0";
+exports.SDK_VERSION = "0.38.0";
 //# sourceMappingURL=version.js.map

package/dist/src/webrtc/index.d.ts

@@ -0,0 +1,42 @@
+import { type ConnectingTone } from "./tones.ts";
+export { type ConnectingTone, JingleTone, RingbackTone } from "./tones.ts";
+export type CallState = "not-started" | "starting" | "waiting-for-permission" | "connecting" | "connected" | "ended";
+export type CallErrorCode = "MICROPHONE_ACCESS_FAILED" | "CONNECTION_FAILED" | "AUTHENTICATION_FAILED" | "PROTOCOL_ERROR" | "SERVER_ERROR" | "REMOTE_HANGUP";
+export declare class CallError extends Error {
+    readonly code: CallErrorCode;
+    constructor(code: CallErrorCode, message: string, options?: ErrorOptions);
+}
+export declare class Call {
+    private _state;
+    private _error?;
+    private readonly _stateHandlers;
+    private readonly _remoteStreamHandlers;
+    private readonly _webrtcCode;
+    private readonly _sockUrl;
+    private readonly _remoteAudio;
+    private readonly _connectingTone?;
+    private _sock?;
+    private _localStream?;
+    private _peerConnection?;
+    private _remoteStream?;
+    private _iceServers?;
+    private _pendingStart?;
+    constructor({ webrtcCode, baseUrl, connectingTone, }: {
+        webrtcCode: string;
+        baseUrl?: string;
+        connectingTone?: ConnectingTone | "disabled";
+    });
+    get state(): CallState;
+    get error(): CallError | undefined;
+    get remoteStream(): MediaStream | undefined;
+    onStateChange(handler: (state: CallState) => void): () => void;
+    onRemoteStream(handler: (stream: MediaStream) => void): () => void;
+    private _setState;
+    start(): Promise<void>;
+    hangup(): Promise<void>;
+    private _send;
+    private _onMessage;
+    private _connectPeer;
+    private _end;
+    private _teardown;
+}