@guava-ai/guava-sdk 0.37.0 → 0.39.0

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@0.39.0/dist/guava-webrtc.min.js"
+  integrity="sha384-1Y1pGUhd4tRQdEQaEUk/2nx9w2Mvl2EEXIZiTw4YFGfUSgp5o9hMJS8UibgJXrNp"
+  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@0.39.0/dist/guava-webrtc.esm.min.js": "sha384-jtuG1DKqz8m5JdMoUZp4jTbGTZTrqJqWqf3tVxsWHqG08UPs7lU9EC4AZkdIzzp0"
+    }
+  }
+</script>
+<script type="module">
+  import * as Guava from "https://cdn.jsdelivr.net/npm/@guava-ai/guava-sdk@0.39.0/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 f=Object.defineProperty;var S=(a,e,t)=>e in a?f(a,e,{enumerable:!0,configurable:!0,writable:!0,value:t}):a[e]=t;var i=(a,e,t)=>S(a,typeof e!="symbol"?e+"":e,t);function _(a,e){return a*(60/e)}var u=class{constructor(){i(this,"context",null);i(this,"intervalId",null);i(this,"volume",.15);i(this,"onSeconds",2);i(this,"periodSeconds",6);i(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 o of[440,480]){let d=e.createOscillator();d.frequency.value=o,d.connect(t),d.start()}let n=e.currentTime+this.startDelay,s=()=>{t.gain.setValueAtTime(this.volume,n),t.gain.setValueAtTime(0,n+this.onSeconds),n+=this.periodSeconds};s(),this.intervalId=setInterval(s,this.periodSeconds*1e3),this.context=e}stop(){this.context&&(this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null),this.context.close(),this.context=null)}},h=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",.3);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=_(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,s,o]=t,d={C:0,D:2,E:4,F:5,G:7,A:9,B:11},c=s==="#"?1:s==="b"?-1:0,l=(Number(o)+1)*12+d[n]+c;return 440*Math.pow(2,(l-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=s=>e+_(s,this.bpm),n=_(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 s=this.context.createOscillator(),o=this.context.createGain();s.type="sine",s.frequency.value=this.noteToFrequency(e),o.gain.setValueAtTime(0,t),o.gain.linearRampToValueAtTime(.7,t+.05),o.gain.exponentialRampToValueAtTime(.001,t+n),s.connect(o),o.connect(this.masterGain),s.start(t),s.stop(t+n+.01),s.onended=()=>{s.disconnect(),o.disconnect()}}},p=class{play(){let e=new AudioContext;e.state==="suspended"&&e.resume();let t=e.currentTime,n=this.playNote(e,{frequency:293.665,startTime:t,attack:.065,decay:.085,volume:.2}),s=this.playNote(e,{frequency:220,startTime:t+.139,attack:.018,decay:.075,volume:.2});Promise.all([n,s]).then(()=>e.close())}playNote(e,{frequency:t,startTime:n,attack:s,decay:o,volume:d}){let c=e.createOscillator(),l=e.createGain();c.type="sine",c.frequency.setValueAtTime(t,n),l.gain.setValueAtTime(0,n),l.gain.linearRampToValueAtTime(d,n+s);let m=n+s,v=m+o*5;return l.gain.setTargetAtTime(1e-5,m,o),c.connect(l),l.connect(e.destination),c.start(n),c.stop(v),new Promise(y=>{c.onended=()=>{c.disconnect(),l.disconnect(),y()}})}};var r=class extends Error{constructor(t,n,s){super(n,s);i(this,"code");this.name="CallError",this.code=t}},C=[{urls:["stun:stun.l.google.com:19302","stun:stun.sipgate.net:10000"]}],T={"http:":"ws:","https:":"wss:","ws:":"ws:","wss:":"wss:"},g=class{constructor({webrtcCode:e,baseUrl:t="https://app.goguava.ai/",connectingTone:n=new h,leavingTone:s=new p,connectionTimeoutMs:o=3e4,customPayload:d={}}){i(this,"_state","not-started");i(this,"_error");i(this,"_stateHandlers",new Set);i(this,"_remoteStreamHandlers",new Set);i(this,"_messageHandlers",new Set);i(this,"_webrtcCode");i(this,"_sockUrl");i(this,"_remoteAudio");i(this,"_connectingTone");i(this,"_leavingTone");i(this,"_connectionTimeoutMs");i(this,"_customPayload");i(this,"_connectionTimer");i(this,"_sock");i(this,"_chatSockUrl");i(this,"_chatSessionId",crypto.randomUUID());i(this,"_chatSock");i(this,"_chatReady",!1);i(this,"_pendingChatMessages",[]);i(this,"_localStream");i(this,"_peerConnection");i(this,"_remoteStream");i(this,"_iceServers");i(this,"_pendingStart");this._webrtcCode=e,this._customPayload=d,n!=="disabled"&&(this._connectingTone=n),s!=="disabled"&&(this._leavingTone=s),o!=="disabled"&&(this._connectionTimeoutMs=o),this._sockUrl=new URL("/webrtc/",t);let c=T[this._sockUrl.protocol];if(c===void 0)throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);this._sockUrl.protocol=c,this._chatSockUrl=new URL("/webrtc-chat/",this._sockUrl),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)}}onMessage(e){return this._messageHandlers.add(e),()=>{this._messageHandlers.delete(e)}}_setState(e){if(e!==this._state){e==="ended"&&this._leavingTone?.play(),this._state=e,e==="connecting"?(this._connectingTone?.start(),this._connectionTimeoutMs!==void 0&&(this._connectionTimer=setTimeout(()=>{this._send({type:"hangup"}),this._end(new r("CONNECTION_TIMEOUT","Timed out connecting the call."))},this._connectionTimeoutMs))):(this._connectingTone?.stop(),clearTimeout(this._connectionTimer),this._connectionTimer=void 0);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 s=new r("MICROPHONE_ACCESS_FAILED","Failed to access microphone.",{cause:n});throw this._end(s),s}finally{clearTimeout(e)}if(this.state==="ended"){this._teardown();return}this._setState("connecting");let t=new Promise((n,s)=>{this._pendingStart={resolve:n,reject:s}});this._sock=new WebSocket(this._sockUrl),this._sock.onopen=()=>{this._send({type:"auth",body:JSON.stringify({webrtc_code:this._webrtcCode,chat_session_id:this._chatSessionId})})},this._sock.onmessage=n=>{this._onMessage(JSON.parse(n.data))},this._sock.onclose=()=>{this._end(new r("CONNECTION_FAILED","Signaling connection closed."))},await t}sendMessage(e){if(this._state==="not-started"||this._state==="ended")throw new Error(`sendMessage() called while the call is ${this._state}.`);let t=JSON.stringify({type:"webrtc-message",payload:e});this._chatReady&&this._chatSock!==void 0?this._chatSock.send(t):this._pendingChatMessages.push(t)}_openChatSocket(){this._chatSock=new WebSocket(this._chatSockUrl),this._chatSock.onopen=()=>{this._chatSock?.send(JSON.stringify({webrtc_code:this._webrtcCode,chat_session_id:this._chatSessionId}))},this._chatSock.onmessage=e=>{let t=JSON.parse(e.data);if(t.type==="ready"&&!this._chatReady){this._chatReady=!0;for(let n of this._pendingChatMessages)this._chatSock?.send(n);this._pendingChatMessages=[]}else if(t.command_type==="send-webrtc-message")for(let n of this._messageHandlers)n(t.payload??{})}}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 r("AUTHENTICATION_FAILED",e.reason||"Authentication failed."));return}this._iceServers=e.ice_servers,this._openChatSocket(),this._send({type:"connect",body:{custom_payload:this._customPayload}}),this._connectPeer();break;case"desc":if(!this._peerConnection){this._end(new r("PROTOCOL_ERROR","Received desc before the peer connection was created."));return}this._peerConnection.setRemoteDescription(e.body).catch(t=>{this._end(new r("CONNECTION_FAILED","Failed to set remote description.",{cause:t}))});break;case"candidate":if(!this._peerConnection){this._end(new r("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 r("REMOTE_HANGUP","The remote hung up before the call connected."));break;case"error":this._end(new r("SERVER_ERROR",e.body||"Unknown server error."));break;case"transfer":this._end(new r("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 r("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 r("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._chatSock&&(this._chatSock.close(),this._chatSock=void 0),this._chatReady=!1,this._pendingChatMessages=[],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{g as Call,r as CallError,h as JingleTone,p as LeavingTone,u 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 = 0.3;\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\nexport interface Tone {\n  play(): void;\n}\n\nexport class LeavingTone implements Tone {\n  public play(): void {\n    const ctx = new AudioContext();\n\n    if (ctx.state === \"suspended\") {\n      void ctx.resume();\n    }\n\n    const now = ctx.currentTime;\n\n    // First note: D4\n    const first = this.playNote(ctx, {\n      frequency: 293.665,\n      startTime: now,\n      attack: 0.065,\n      decay: 0.085,\n      volume: 0.2,\n    });\n\n    // Second note: A3, starts 139ms later\n    const second = this.playNote(ctx, {\n      frequency: 220.0,\n      startTime: now + 0.139,\n      attack: 0.018,\n      decay: 0.075,\n      volume: 0.2,\n    });\n\n    void Promise.all([first, second]).then(() => ctx.close());\n  }\n\n  // Resolves once the note has finished playing.\n  private playNote(\n    ctx: AudioContext,\n    {\n      frequency,\n      startTime,\n      attack,\n      decay,\n      volume,\n    }: {\n      frequency: number;\n      startTime: number;\n      attack: number;\n      decay: number;\n      volume: number;\n    },\n  ): Promise<void> {\n    const oscillator = ctx.createOscillator();\n    const gain = ctx.createGain();\n\n    oscillator.type = \"sine\";\n    oscillator.frequency.setValueAtTime(frequency, startTime);\n\n    // Smooth onset\n    gain.gain.setValueAtTime(0, startTime);\n    gain.gain.linearRampToValueAtTime(volume, startTime + attack);\n\n    // Exponential decay\n    const peakTime = startTime + attack;\n    const endTime = peakTime + decay * 5;\n\n    gain.gain.setTargetAtTime(0.00001, peakTime, decay);\n\n    oscillator.connect(gain);\n    gain.connect(ctx.destination);\n\n    oscillator.start(startTime);\n    oscillator.stop(endTime);\n\n    return new Promise((resolve) => {\n      oscillator.onended = () => {\n        oscillator.disconnect();\n        gain.disconnect();\n        resolve();\n      };\n    });\n  }\n}\n", "import { type ConnectingTone, JingleTone, LeavingTone, type Tone } from \"./tones.ts\";\n\nexport { type ConnectingTone, JingleTone, RingbackTone, LeavingTone, type Tone } 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  | \"CONNECTION_TIMEOUT\"; // The call didn't connect within connectionTimeoutMs.\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 _messageHandlers = new Set<(payload: Record<string, unknown>) => void>();\n  private readonly _webrtcCode: string;\n  private readonly _sockUrl: URL;\n\n  private readonly _remoteAudio: HTMLAudioElement;\n  private readonly _connectingTone?: ConnectingTone;\n  private readonly _leavingTone?: Tone;\n  private readonly _connectionTimeoutMs?: number;\n  private readonly _customPayload: Record<string, unknown>;\n  private _connectionTimer?: ReturnType<typeof setTimeout>;\n  private _sock?: WebSocket;\n\n  private readonly _chatSockUrl: URL;\n  // Links the chat socket to this call's session on the server.\n  private readonly _chatSessionId = crypto.randomUUID();\n  private _chatSock?: WebSocket;\n  private _chatReady = false;\n  // Messages sent before the chat socket is ready.\n  private _pendingChatMessages: string[] = [];\n\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    leavingTone = new LeavingTone(),\n    connectionTimeoutMs = 30_000,\n    customPayload = {},\n  }: {\n    /** Code used to authenticate with the signaling server. */\n    webrtcCode: string;\n    /** Base URL of the Guava server. Defaults to `https://app.goguava.ai/`. */\n    baseUrl?: string;\n    /** Played while the call is in the \"connecting\" state. Defaults to {@link JingleTone}. */\n    connectingTone?: ConnectingTone | \"disabled\";\n    /** Played when the call ends. Defaults to {@link LeavingTone}. */\n    leavingTone?: Tone | \"disabled\";\n    /** Max time in the \"connecting\" state before the call fails with CONNECTION_TIMEOUT. Defaults to 30 seconds. */\n    connectionTimeoutMs?: number | \"disabled\";\n    /** A custom payload that is forwarded to the agent at the start of the call. */\n    customPayload?: Record<string, unknown>;\n  }) {\n    this._webrtcCode = webrtcCode;\n    this._customPayload = customPayload;\n    if (connectingTone !== \"disabled\") {\n      this._connectingTone = connectingTone;\n    }\n    if (leavingTone !== \"disabled\") {\n      this._leavingTone = leavingTone;\n    }\n    if (connectionTimeoutMs !== \"disabled\") {\n      this._connectionTimeoutMs = connectionTimeoutMs;\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    this._chatSockUrl = new URL(\"/webrtc-chat/\", this._sockUrl);\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  /**\n   * Registers a handler for messages the agent sends with `call.send_webrtc_message()`.\n   * @param handler - Called with each message's payload.\n   * @returns A function that unregisters the handler.\n   */\n  onMessage(handler: (payload: Record<string, unknown>) => void): () => void {\n    this._messageHandlers.add(handler);\n    return () => {\n      this._messageHandlers.delete(handler);\n    };\n  }\n\n  private _setState(state: CallState) {\n    if (state === this._state) {\n      return;\n    }\n\n    if (state === \"ended\") {\n      this._leavingTone?.play();\n    }\n\n    this._state = state;\n    if (state === \"connecting\") {\n      this._connectingTone?.start();\n      if (this._connectionTimeoutMs !== undefined) {\n        this._connectionTimer = setTimeout(() => {\n          this._send({ type: \"hangup\" });\n          this._end(new CallError(\"CONNECTION_TIMEOUT\", \"Timed out connecting the call.\"));\n        }, this._connectionTimeoutMs);\n      }\n    } else {\n      this._connectingTone?.stop();\n      clearTimeout(this._connectionTimer);\n      this._connectionTimer = undefined;\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({\n        type: \"auth\",\n        body: JSON.stringify({\n          webrtc_code: this._webrtcCode,\n          chat_session_id: this._chatSessionId,\n        }),\n      });\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  /**\n   * Sends a message to the agent, delivered to its `on_webrtc_message` handler.\n   * Messages sent before the message channel is ready are queued.\n   * @param payload - A JSON-serializable object.\n   */\n  sendMessage(payload: Record<string, unknown>) {\n    if (this._state === \"not-started\" || this._state === \"ended\") {\n      throw new Error(`sendMessage() called while the call is ${this._state}.`);\n    }\n    const message = JSON.stringify({ type: \"webrtc-message\", payload });\n    if (this._chatReady && this._chatSock !== undefined) {\n      this._chatSock.send(message);\n    } else {\n      this._pendingChatMessages.push(message);\n    }\n  }\n\n  private _openChatSocket() {\n    this._chatSock = new WebSocket(this._chatSockUrl);\n    this._chatSock.onopen = () => {\n      this._chatSock?.send(\n        JSON.stringify({ webrtc_code: this._webrtcCode, chat_session_id: this._chatSessionId }),\n      );\n    };\n    this._chatSock.onmessage = (event) => {\n      const message = JSON.parse(event.data) as {\n        type?: string;\n        command_type?: string;\n        payload?: Record<string, unknown>;\n      };\n      // The server sends \"ready\" once the call session is found.\n      if (message.type === \"ready\" && !this._chatReady) {\n        this._chatReady = true;\n        for (const pending of this._pendingChatMessages) {\n          this._chatSock?.send(pending);\n        }\n        this._pendingChatMessages = [];\n      } else if (message.command_type === \"send-webrtc-message\") {\n        for (const handler of this._messageHandlers) {\n          handler(message.payload ?? {});\n        }\n      }\n    };\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._openChatSocket();\n        this._send({ type: \"connect\", body: { custom_payload: this._customPayload } });\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    // Close the signaling socket.\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\n    // Close the chat socket.\n    if (this._chatSock) {\n      this._chatSock.close();\n      this._chatSock = undefined;\n    }\n    this._chatReady = false;\n    this._pendingChatMessages = [];\n\n    // Close the peer connection.\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\n    // Remove the audio element.\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", "LeavingTone", "ctx", "first", "second", "attack", "decay", "volume", "peakTime", "endTime", "resolve", "CallError", "code", "message", "options", "__publicField", "DEFAULT_ICE_SERVERS", "SOCKET_PROTOCOLS", "Call", "webrtcCode", "baseUrl", "connectingTone", "JingleTone", "leavingTone", "LeavingTone", "connectionTimeoutMs", "customPayload", "sockProtocol", "handler", "state", "permissionTimer", "error", "callError", "started", "resolve", "reject", "event", "payload", "pending", "track"]
+}

package/dist/guava-webrtc.min.js

@@ -0,0 +1,2 @@
+var Guava=(()=>{var _=Object.defineProperty;var S=Object.getOwnPropertyDescriptor;var C=Object.getOwnPropertyNames;var T=Object.prototype.hasOwnProperty;var w=(o,e,t)=>e in o?_(o,e,{enumerable:!0,configurable:!0,writable:!0,value:t}):o[e]=t;var b=(o,e)=>{for(var t in e)_(o,t,{get:e[t],enumerable:!0})},R=(o,e,t,n)=>{if(e&&typeof e=="object"||typeof e=="function")for(let s of C(e))!T.call(o,s)&&s!==t&&_(o,s,{get:()=>e[s],enumerable:!(n=S(e,s))||n.enumerable});return o};var k=o=>R(_({},"__esModule",{value:!0}),o);var i=(o,e,t)=>w(o,typeof e!="symbol"?e+"":e,t);var I={};b(I,{Call:()=>g,CallError:()=>a,JingleTone:()=>h,LeavingTone:()=>p,RingbackTone:()=>u});function m(o,e){return o*(60/e)}var u=class{constructor(){i(this,"context",null);i(this,"intervalId",null);i(this,"volume",.15);i(this,"onSeconds",2);i(this,"periodSeconds",6);i(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 r of[440,480]){let d=e.createOscillator();d.frequency.value=r,d.connect(t),d.start()}let n=e.currentTime+this.startDelay,s=()=>{t.gain.setValueAtTime(this.volume,n),t.gain.setValueAtTime(0,n+this.onSeconds),n+=this.periodSeconds};s(),this.intervalId=setInterval(s,this.periodSeconds*1e3),this.context=e}stop(){this.context&&(this.intervalId&&(clearInterval(this.intervalId),this.intervalId=null),this.context.close(),this.context=null)}},h=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",.3);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=m(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,s,r]=t,d={C:0,D:2,E:4,F:5,G:7,A:9,B:11},c=s==="#"?1:s==="b"?-1:0,l=(Number(r)+1)*12+d[n]+c;return 440*Math.pow(2,(l-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=s=>e+m(s,this.bpm),n=m(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 s=this.context.createOscillator(),r=this.context.createGain();s.type="sine",s.frequency.value=this.noteToFrequency(e),r.gain.setValueAtTime(0,t),r.gain.linearRampToValueAtTime(.7,t+.05),r.gain.exponentialRampToValueAtTime(.001,t+n),s.connect(r),r.connect(this.masterGain),s.start(t),s.stop(t+n+.01),s.onended=()=>{s.disconnect(),r.disconnect()}}},p=class{play(){let e=new AudioContext;e.state==="suspended"&&e.resume();let t=e.currentTime,n=this.playNote(e,{frequency:293.665,startTime:t,attack:.065,decay:.085,volume:.2}),s=this.playNote(e,{frequency:220,startTime:t+.139,attack:.018,decay:.075,volume:.2});Promise.all([n,s]).then(()=>e.close())}playNote(e,{frequency:t,startTime:n,attack:s,decay:r,volume:d}){let c=e.createOscillator(),l=e.createGain();c.type="sine",c.frequency.setValueAtTime(t,n),l.gain.setValueAtTime(0,n),l.gain.linearRampToValueAtTime(d,n+s);let v=n+s,y=v+r*5;return l.gain.setTargetAtTime(1e-5,v,r),c.connect(l),l.connect(e.destination),c.start(n),c.stop(y),new Promise(f=>{c.onended=()=>{c.disconnect(),l.disconnect(),f()}})}};var a=class extends Error{constructor(t,n,s){super(n,s);i(this,"code");this.name="CallError",this.code=t}},E=[{urls:["stun:stun.l.google.com:19302","stun:stun.sipgate.net:10000"]}],O={"http:":"ws:","https:":"wss:","ws:":"ws:","wss:":"wss:"},g=class{constructor({webrtcCode:e,baseUrl:t="https://app.goguava.ai/",connectingTone:n=new h,leavingTone:s=new p,connectionTimeoutMs:r=3e4,customPayload:d={}}){i(this,"_state","not-started");i(this,"_error");i(this,"_stateHandlers",new Set);i(this,"_remoteStreamHandlers",new Set);i(this,"_messageHandlers",new Set);i(this,"_webrtcCode");i(this,"_sockUrl");i(this,"_remoteAudio");i(this,"_connectingTone");i(this,"_leavingTone");i(this,"_connectionTimeoutMs");i(this,"_customPayload");i(this,"_connectionTimer");i(this,"_sock");i(this,"_chatSockUrl");i(this,"_chatSessionId",crypto.randomUUID());i(this,"_chatSock");i(this,"_chatReady",!1);i(this,"_pendingChatMessages",[]);i(this,"_localStream");i(this,"_peerConnection");i(this,"_remoteStream");i(this,"_iceServers");i(this,"_pendingStart");this._webrtcCode=e,this._customPayload=d,n!=="disabled"&&(this._connectingTone=n),s!=="disabled"&&(this._leavingTone=s),r!=="disabled"&&(this._connectionTimeoutMs=r),this._sockUrl=new URL("/webrtc/",t);let c=O[this._sockUrl.protocol];if(c===void 0)throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);this._sockUrl.protocol=c,this._chatSockUrl=new URL("/webrtc-chat/",this._sockUrl),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)}}onMessage(e){return this._messageHandlers.add(e),()=>{this._messageHandlers.delete(e)}}_setState(e){if(e!==this._state){e==="ended"&&this._leavingTone?.play(),this._state=e,e==="connecting"?(this._connectingTone?.start(),this._connectionTimeoutMs!==void 0&&(this._connectionTimer=setTimeout(()=>{this._send({type:"hangup"}),this._end(new a("CONNECTION_TIMEOUT","Timed out connecting the call."))},this._connectionTimeoutMs))):(this._connectingTone?.stop(),clearTimeout(this._connectionTimer),this._connectionTimer=void 0);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 s=new a("MICROPHONE_ACCESS_FAILED","Failed to access microphone.",{cause:n});throw this._end(s),s}finally{clearTimeout(e)}if(this.state==="ended"){this._teardown();return}this._setState("connecting");let t=new Promise((n,s)=>{this._pendingStart={resolve:n,reject:s}});this._sock=new WebSocket(this._sockUrl),this._sock.onopen=()=>{this._send({type:"auth",body:JSON.stringify({webrtc_code:this._webrtcCode,chat_session_id:this._chatSessionId})})},this._sock.onmessage=n=>{this._onMessage(JSON.parse(n.data))},this._sock.onclose=()=>{this._end(new a("CONNECTION_FAILED","Signaling connection closed."))},await t}sendMessage(e){if(this._state==="not-started"||this._state==="ended")throw new Error(`sendMessage() called while the call is ${this._state}.`);let t=JSON.stringify({type:"webrtc-message",payload:e});this._chatReady&&this._chatSock!==void 0?this._chatSock.send(t):this._pendingChatMessages.push(t)}_openChatSocket(){this._chatSock=new WebSocket(this._chatSockUrl),this._chatSock.onopen=()=>{this._chatSock?.send(JSON.stringify({webrtc_code:this._webrtcCode,chat_session_id:this._chatSessionId}))},this._chatSock.onmessage=e=>{let t=JSON.parse(e.data);if(t.type==="ready"&&!this._chatReady){this._chatReady=!0;for(let n of this._pendingChatMessages)this._chatSock?.send(n);this._pendingChatMessages=[]}else if(t.command_type==="send-webrtc-message")for(let n of this._messageHandlers)n(t.payload??{})}}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 a("AUTHENTICATION_FAILED",e.reason||"Authentication failed."));return}this._iceServers=e.ice_servers,this._openChatSocket(),this._send({type:"connect",body:{custom_payload:this._customPayload}}),this._connectPeer();break;case"desc":if(!this._peerConnection){this._end(new a("PROTOCOL_ERROR","Received desc before the peer connection was created."));return}this._peerConnection.setRemoteDescription(e.body).catch(t=>{this._end(new a("CONNECTION_FAILED","Failed to set remote description.",{cause:t}))});break;case"candidate":if(!this._peerConnection){this._end(new a("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 a("REMOTE_HANGUP","The remote hung up before the call connected."));break;case"error":this._end(new a("SERVER_ERROR",e.body||"Unknown server error."));break;case"transfer":this._end(new a("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??E}),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 a("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 a("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._chatSock&&(this._chatSock.close(),this._chatSock=void 0),this._chatReady=!1,this._pendingChatMessages=[],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 k(I);})();
+//# 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, LeavingTone, type Tone } from \"./tones.ts\";\n\nexport { type ConnectingTone, JingleTone, RingbackTone, LeavingTone, type Tone } 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  | \"CONNECTION_TIMEOUT\"; // The call didn't connect within connectionTimeoutMs.\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 _messageHandlers = new Set<(payload: Record<string, unknown>) => void>();\n  private readonly _webrtcCode: string;\n  private readonly _sockUrl: URL;\n\n  private readonly _remoteAudio: HTMLAudioElement;\n  private readonly _connectingTone?: ConnectingTone;\n  private readonly _leavingTone?: Tone;\n  private readonly _connectionTimeoutMs?: number;\n  private readonly _customPayload: Record<string, unknown>;\n  private _connectionTimer?: ReturnType<typeof setTimeout>;\n  private _sock?: WebSocket;\n\n  private readonly _chatSockUrl: URL;\n  // Links the chat socket to this call's session on the server.\n  private readonly _chatSessionId = crypto.randomUUID();\n  private _chatSock?: WebSocket;\n  private _chatReady = false;\n  // Messages sent before the chat socket is ready.\n  private _pendingChatMessages: string[] = [];\n\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    leavingTone = new LeavingTone(),\n    connectionTimeoutMs = 30_000,\n    customPayload = {},\n  }: {\n    /** Code used to authenticate with the signaling server. */\n    webrtcCode: string;\n    /** Base URL of the Guava server. Defaults to `https://app.goguava.ai/`. */\n    baseUrl?: string;\n    /** Played while the call is in the \"connecting\" state. Defaults to {@link JingleTone}. */\n    connectingTone?: ConnectingTone | \"disabled\";\n    /** Played when the call ends. Defaults to {@link LeavingTone}. */\n    leavingTone?: Tone | \"disabled\";\n    /** Max time in the \"connecting\" state before the call fails with CONNECTION_TIMEOUT. Defaults to 30 seconds. */\n    connectionTimeoutMs?: number | \"disabled\";\n    /** A custom payload that is forwarded to the agent at the start of the call. */\n    customPayload?: Record<string, unknown>;\n  }) {\n    this._webrtcCode = webrtcCode;\n    this._customPayload = customPayload;\n    if (connectingTone !== \"disabled\") {\n      this._connectingTone = connectingTone;\n    }\n    if (leavingTone !== \"disabled\") {\n      this._leavingTone = leavingTone;\n    }\n    if (connectionTimeoutMs !== \"disabled\") {\n      this._connectionTimeoutMs = connectionTimeoutMs;\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    this._chatSockUrl = new URL(\"/webrtc-chat/\", this._sockUrl);\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  /**\n   * Registers a handler for messages the agent sends with `call.send_webrtc_message()`.\n   * @param handler - Called with each message's payload.\n   * @returns A function that unregisters the handler.\n   */\n  onMessage(handler: (payload: Record<string, unknown>) => void): () => void {\n    this._messageHandlers.add(handler);\n    return () => {\n      this._messageHandlers.delete(handler);\n    };\n  }\n\n  private _setState(state: CallState) {\n    if (state === this._state) {\n      return;\n    }\n\n    if (state === \"ended\") {\n      this._leavingTone?.play();\n    }\n\n    this._state = state;\n    if (state === \"connecting\") {\n      this._connectingTone?.start();\n      if (this._connectionTimeoutMs !== undefined) {\n        this._connectionTimer = setTimeout(() => {\n          this._send({ type: \"hangup\" });\n          this._end(new CallError(\"CONNECTION_TIMEOUT\", \"Timed out connecting the call.\"));\n        }, this._connectionTimeoutMs);\n      }\n    } else {\n      this._connectingTone?.stop();\n      clearTimeout(this._connectionTimer);\n      this._connectionTimer = undefined;\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({\n        type: \"auth\",\n        body: JSON.stringify({\n          webrtc_code: this._webrtcCode,\n          chat_session_id: this._chatSessionId,\n        }),\n      });\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  /**\n   * Sends a message to the agent, delivered to its `on_webrtc_message` handler.\n   * Messages sent before the message channel is ready are queued.\n   * @param payload - A JSON-serializable object.\n   */\n  sendMessage(payload: Record<string, unknown>) {\n    if (this._state === \"not-started\" || this._state === \"ended\") {\n      throw new Error(`sendMessage() called while the call is ${this._state}.`);\n    }\n    const message = JSON.stringify({ type: \"webrtc-message\", payload });\n    if (this._chatReady && this._chatSock !== undefined) {\n      this._chatSock.send(message);\n    } else {\n      this._pendingChatMessages.push(message);\n    }\n  }\n\n  private _openChatSocket() {\n    this._chatSock = new WebSocket(this._chatSockUrl);\n    this._chatSock.onopen = () => {\n      this._chatSock?.send(\n        JSON.stringify({ webrtc_code: this._webrtcCode, chat_session_id: this._chatSessionId }),\n      );\n    };\n    this._chatSock.onmessage = (event) => {\n      const message = JSON.parse(event.data) as {\n        type?: string;\n        command_type?: string;\n        payload?: Record<string, unknown>;\n      };\n      // The server sends \"ready\" once the call session is found.\n      if (message.type === \"ready\" && !this._chatReady) {\n        this._chatReady = true;\n        for (const pending of this._pendingChatMessages) {\n          this._chatSock?.send(pending);\n        }\n        this._pendingChatMessages = [];\n      } else if (message.command_type === \"send-webrtc-message\") {\n        for (const handler of this._messageHandlers) {\n          handler(message.payload ?? {});\n        }\n      }\n    };\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._openChatSocket();\n        this._send({ type: \"connect\", body: { custom_payload: this._customPayload } });\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    // Close the signaling socket.\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\n    // Close the chat socket.\n    if (this._chatSock) {\n      this._chatSock.close();\n      this._chatSock = undefined;\n    }\n    this._chatReady = false;\n    this._pendingChatMessages = [];\n\n    // Close the peer connection.\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\n    // Remove the audio element.\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 = 0.3;\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\nexport interface Tone {\n  play(): void;\n}\n\nexport class LeavingTone implements Tone {\n  public play(): void {\n    const ctx = new AudioContext();\n\n    if (ctx.state === \"suspended\") {\n      void ctx.resume();\n    }\n\n    const now = ctx.currentTime;\n\n    // First note: D4\n    const first = this.playNote(ctx, {\n      frequency: 293.665,\n      startTime: now,\n      attack: 0.065,\n      decay: 0.085,\n      volume: 0.2,\n    });\n\n    // Second note: A3, starts 139ms later\n    const second = this.playNote(ctx, {\n      frequency: 220.0,\n      startTime: now + 0.139,\n      attack: 0.018,\n      decay: 0.075,\n      volume: 0.2,\n    });\n\n    void Promise.all([first, second]).then(() => ctx.close());\n  }\n\n  // Resolves once the note has finished playing.\n  private playNote(\n    ctx: AudioContext,\n    {\n      frequency,\n      startTime,\n      attack,\n      decay,\n      volume,\n    }: {\n      frequency: number;\n      startTime: number;\n      attack: number;\n      decay: number;\n      volume: number;\n    },\n  ): Promise<void> {\n    const oscillator = ctx.createOscillator();\n    const gain = ctx.createGain();\n\n    oscillator.type = \"sine\";\n    oscillator.frequency.setValueAtTime(frequency, startTime);\n\n    // Smooth onset\n    gain.gain.setValueAtTime(0, startTime);\n    gain.gain.linearRampToValueAtTime(volume, startTime + attack);\n\n    // Exponential decay\n    const peakTime = startTime + attack;\n    const endTime = peakTime + decay * 5;\n\n    gain.gain.setTargetAtTime(0.00001, peakTime, decay);\n\n    oscillator.connect(gain);\n    gain.connect(ctx.destination);\n\n    oscillator.start(startTime);\n    oscillator.stop(endTime);\n\n    return new Promise((resolve) => {\n      oscillator.onended = () => {\n        oscillator.disconnect();\n        gain.disconnect();\n        resolve();\n      };\n    });\n  }\n}\n"],
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+  "names": ["index_exports", "__export", "Call", "CallError", "JingleTone", "LeavingTone", "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", "LeavingTone", "ctx", "first", "second", "attack", "decay", "volume", "peakTime", "endTime", "resolve", "CallError", "code", "message", "options", "__publicField", "DEFAULT_ICE_SERVERS", "SOCKET_PROTOCOLS", "Call", "webrtcCode", "baseUrl", "connectingTone", "JingleTone", "leavingTone", "LeavingTone", "connectionTimeoutMs", "customPayload", "sockProtocol", "handler", "state", "permissionTimer", "error", "callError", "started", "resolve", "reject", "event", "payload", "pending", "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/socket/call-info.d.ts

@@ -9,6 +9,7 @@ export type PSTNCallInfo = z.infer<typeof PSTNCallInfo>;
 export declare const WebRTCCallInfo: z.ZodObject<{
     call_type: z.ZodLiteral<"webrtc">;
     webrtc_code: z.ZodString;
+    custom_payload: z.ZodDefault<z.ZodRecord<z.ZodString, z.ZodUnknown>>;
 }, z.core.$strip>;
 export type WebRTCCallInfo = z.infer<typeof WebRTCCallInfo>;
 export declare class SIPCallInfo {
@@ -28,6 +29,7 @@ export declare const CallInfo: z.ZodDiscriminatedUnion<[z.ZodObject<{
 }, z.core.$strip>, z.ZodObject<{
     call_type: z.ZodLiteral<"webrtc">;
     webrtc_code: z.ZodString;
+    custom_payload: z.ZodDefault<z.ZodRecord<z.ZodString, z.ZodUnknown>>;
 }, z.core.$strip>, z.ZodPipe<z.ZodObject<{
     call_type: z.ZodLiteral<"sip">;
     from_aor: z.ZodString;

package/dist/src/socket/call-info.js

@@ -44,6 +44,7 @@ exports.PSTNCallInfo = z.object({
 exports.WebRTCCallInfo = z.object({
     call_type: z.literal("webrtc"),
     webrtc_code: z.string(),
+    custom_payload: z.record(z.string(), z.unknown()).default({}),
 });
 const SIPCallInfoSchema = z
     .object({

package/dist/src/socket/call-info.js.map

@@ -1 +1 @@
-{"version":3,"file":"call-info.js","sourceRoot":"","sources":["../../../src/socket/call-info.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;AAAA,uCAAyB;AAEZ,QAAA,YAAY,GAAG,CAAC,CAAC,MAAM,CAAC;IACnC,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,MAAM,CAAC;IAC5B,WAAW,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;IAClC,SAAS,EAAE,CAAC,CAAC,MAAM,EAAE;IACrB,SAAS,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;CACjC,CAAC,CAAC;AAGU,QAAA,cAAc,GAAG,CAAC,CAAC,MAAM,CAAC;IACrC,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,QAAQ,CAAC;IAC9B,WAAW,EAAE,CAAC,CAAC,MAAM,EAAE;CACxB,CAAC,CAAC;AAGH,MAAM,iBAAiB,GAAG,CAAC;KACxB,MAAM,CAAC;IACN,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,KAAK,CAAC;IAC3B,QAAQ,EAAE,CAAC;SACR,MAAM,EAAE;SACR,QAAQ,CACP,2JAA2J,CAC5J;IACH,QAAQ,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;IAC/B,WAAW,EAAE,CAAC,CAAC,MAAM,CAAC,CAAC,CAAC,MAAM,EAAE,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,CAAC,OAAO,CAAC,EAAE,CAAC;CAC1D,CAAC;KACD,SAAS,CAAC,CAAC,IAAI,EAAE,EAAE,CAAC,IAAI,WAAW,CAAC,IAAI,CAAC,QAAQ,EAAE,IAAI,CAAC,QAAQ,EAAE,IAAI,CAAC,WAAW,CAAC,CAAC,CAAC;AAExF,MAAa,WAAW;IAIX;IACA;IACA;IALF,SAAS,GAAG,KAAc,CAAC;IAEpC,YACW,QAAgB,EAChB,QAAiB,EACjB,cAAsC,EAAE;QAFxC,aAAQ,GAAR,QAAQ,CAAQ;QAChB,aAAQ,GAAR,QAAQ,CAAS;QACjB,gBAAW,GAAX,WAAW,CAA6B;IAChD,CAAC;IAEJ,oBAAoB;IACpB,IAAI,OAAO;QACT,MAAM,KAAK,GAAG,IAAI,CAAC,QAAQ,CAAC,IAAI,EAAE,CAAC;QAEnC,4EAA4E;QAC5E,IAAI,CAAC,GAAG,CAAC,CAAC;QACV,IAAI,KAAK,CAAC,UAAU,CAAC,GAAG,CAAC,EAAE,CAAC;YAC1B,CAAC,GAAG,CAAC,CAAC;YACN,OAAO,CAAC,GAAG,KAAK,CAAC,MAAM,IAAI,KAAK,CAAC,CAAC,CAAC,KAAK,GAAG,EAAE,CAAC;gBAC5C,CAAC,IAAI,KAAK,CAAC,CAAC,CAAC,KAAK,IAAI,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;YACjC,CAAC;YACD,CAAC,IAAI,CAAC,CAAC;QACT,CAAC;QAED,6EAA6E;QAC7E,MAAM,KAAK,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC;QACpC,MAAM,IAAI,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC;QACnC,IAAI,KAAK,KAAK,CAAC,CAAC,IAAI,CAAC,IAAI,KAAK,CAAC,CAAC,IAAI,KAAK,GAAG,IAAI,CAAC,EAAE,CAAC;YAClD,MAAM,GAAG,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,KAAK,CAAC,CAAC;YACtC,IAAI,GAAG,KAAK,CAAC,CAAC,EAAE,CAAC;gBACf,OAAO,KAAK,CAAC,KAAK,CAAC,KAAK,GAAG,CAAC,EAAE,GAAG,CAAC,CAAC,IAAI,EAAE,CAAC;YAC5C,CAAC;QACH,CAAC;QAED,OAAO,KAAK,CAAC,KAAK,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,IAAI,EAAE,CAAC;IACvC,CAAC;CACF;AAnCD,kCAmCC;AAEY,QAAA,QAAQ,GAAG,CAAC,CAAC,kBAAkB,CAAC,WAAW,EAAE;IACxD,oBAAY;IACZ,sBAAc;IACd,iBAAiB;CAClB,CAAC,CAAC"}
+{"version":3,"file":"call-info.js","sourceRoot":"","sources":["../../../src/socket/call-info.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;AAAA,uCAAyB;AAEZ,QAAA,YAAY,GAAG,CAAC,CAAC,MAAM,CAAC;IACnC,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,MAAM,CAAC;IAC5B,WAAW,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;IAClC,SAAS,EAAE,CAAC,CAAC,MAAM,EAAE;IACrB,SAAS,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;CACjC,CAAC,CAAC;AAGU,QAAA,cAAc,GAAG,CAAC,CAAC,MAAM,CAAC;IACrC,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,QAAQ,CAAC;IAC9B,WAAW,EAAE,CAAC,CAAC,MAAM,EAAE;IACvB,cAAc,EAAE,CAAC,CAAC,MAAM,CAAC,CAAC,CAAC,MAAM,EAAE,EAAE,CAAC,CAAC,OAAO,EAAE,CAAC,CAAC,OAAO,CAAC,EAAE,CAAC;CAC9D,CAAC,CAAC;AAGH,MAAM,iBAAiB,GAAG,CAAC;KACxB,MAAM,CAAC;IACN,SAAS,EAAE,CAAC,CAAC,OAAO,CAAC,KAAK,CAAC;IAC3B,QAAQ,EAAE,CAAC;SACR,MAAM,EAAE;SACR,QAAQ,CACP,2JAA2J,CAC5J;IACH,QAAQ,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,QAAQ,EAAE;IAC/B,WAAW,EAAE,CAAC,CAAC,MAAM,CAAC,CAAC,CAAC,MAAM,EAAE,EAAE,CAAC,CAAC,MAAM,EAAE,CAAC,CAAC,OAAO,CAAC,EAAE,CAAC;CAC1D,CAAC;KACD,SAAS,CAAC,CAAC,IAAI,EAAE,EAAE,CAAC,IAAI,WAAW,CAAC,IAAI,CAAC,QAAQ,EAAE,IAAI,CAAC,QAAQ,EAAE,IAAI,CAAC,WAAW,CAAC,CAAC,CAAC;AAExF,MAAa,WAAW;IAIX;IACA;IACA;IALF,SAAS,GAAG,KAAc,CAAC;IAEpC,YACW,QAAgB,EAChB,QAAiB,EACjB,cAAsC,EAAE;QAFxC,aAAQ,GAAR,QAAQ,CAAQ;QAChB,aAAQ,GAAR,QAAQ,CAAS;QACjB,gBAAW,GAAX,WAAW,CAA6B;IAChD,CAAC;IAEJ,oBAAoB;IACpB,IAAI,OAAO;QACT,MAAM,KAAK,GAAG,IAAI,CAAC,QAAQ,CAAC,IAAI,EAAE,CAAC;QAEnC,4EAA4E;QAC5E,IAAI,CAAC,GAAG,CAAC,CAAC;QACV,IAAI,KAAK,CAAC,UAAU,CAAC,GAAG,CAAC,EAAE,CAAC;YAC1B,CAAC,GAAG,CAAC,CAAC;YACN,OAAO,CAAC,GAAG,KAAK,CAAC,MAAM,IAAI,KAAK,CAAC,CAAC,CAAC,KAAK,GAAG,EAAE,CAAC;gBAC5C,CAAC,IAAI,KAAK,CAAC,CAAC,CAAC,KAAK,IAAI,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;YACjC,CAAC;YACD,CAAC,IAAI,CAAC,CAAC;QACT,CAAC;QAED,6EAA6E;QAC7E,MAAM,KAAK,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC;QACpC,MAAM,IAAI,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC;QACnC,IAAI,KAAK,KAAK,CAAC,CAAC,IAAI,CAAC,IAAI,KAAK,CAAC,CAAC,IAAI,KAAK,GAAG,IAAI,CAAC,EAAE,CAAC;YAClD,MAAM,GAAG,GAAG,KAAK,CAAC,OAAO,CAAC,GAAG,EAAE,KAAK,CAAC,CAAC;YACtC,IAAI,GAAG,KAAK,CAAC,CAAC,EAAE,CAAC;gBACf,OAAO,KAAK,CAAC,KAAK,CAAC,KAAK,GAAG,CAAC,EAAE,GAAG,CAAC,CAAC,IAAI,EAAE,CAAC;YAC5C,CAAC;QACH,CAAC;QAED,OAAO,KAAK,CAAC,KAAK,CAAC,GAAG,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,IAAI,EAAE,CAAC;IACvC,CAAC;CACF;AAnCD,kCAmCC;AAEY,QAAA,QAAQ,GAAG,CAAC,CAAC,kBAAkB,CAAC,WAAW,EAAE;IACxD,oBAAY;IACZ,sBAAc;IACd,iBAAiB;CAClB,CAAC,CAAC"}

package/dist/src/socket/listen-inbound.d.ts

@@ -21,6 +21,7 @@ export declare const AssignCall: z.ZodObject<{
     }, z.core.$strip>, z.ZodObject<{
         call_type: z.ZodLiteral<"webrtc">;
         webrtc_code: z.ZodString;
+        custom_payload: z.ZodDefault<z.ZodRecord<z.ZodString, z.ZodUnknown>>;
     }, z.core.$strip>, z.ZodPipe<z.ZodObject<{
         call_type: z.ZodLiteral<"sip">;
         from_aor: z.ZodString;
@@ -51,6 +52,7 @@ export declare const ServerMessage: z.ZodDiscriminatedUnion<[z.ZodObject<{
     }, z.core.$strip>, z.ZodObject<{
         call_type: z.ZodLiteral<"webrtc">;
         webrtc_code: z.ZodString;
+        custom_payload: z.ZodDefault<z.ZodRecord<z.ZodString, z.ZodUnknown>>;
     }, z.core.$strip>, z.ZodPipe<z.ZodObject<{
         call_type: z.ZodLiteral<"sip">;
         from_aor: z.ZodString;

package/dist/src/version.d.ts

@@ -1 +1 @@
-export declare const SDK_VERSION = "0.37.0";
+export declare const SDK_VERSION = "0.39.0";

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 = "0.39.0";
 //# sourceMappingURL=version.js.map