@marianmeres/webrtc 1.3.0 → 1.4.0

package/AGENTS.md

@@ -41,10 +41,10 @@ development:
 src/
   mod.ts              # Entry point, re-exports all public APIs
   types.ts            # Type definitions (interfaces, enums)
-  webrtc-manager.ts   # Main WebRtcManager class
+  webrtc-manager.ts   # Main WebRTCManager class
 
 tests/
-  mocks.ts                    # Mock WebRtcFactory for testing
+  mocks.ts                    # Mock WebRTCFactory for testing
   webrtc-manager.test.ts      # Deno unit tests
   browser/
     p2p-tests.ts              # Browser integration tests
@@ -65,7 +65,7 @@ scripts/
 
 ## State Machine
 
-### States (WebRtcState)
+### States (WebRTCState)
 
 | State | Description | Valid Outgoing Transitions |
 |-------|-------------|---------------------------|
@@ -77,7 +77,7 @@ scripts/
 | DISCONNECTED | Connection closed, resources may be cleaned up | CONNECTING, RECONNECTING, IDLE |
 | ERROR | Error state, requires reset() to recover | IDLE |
 
-### Events (WebRtcFsmEvent)
+### Events (WebRTCFsmEvent)
 
 | Event | Value | Description |
 |-------|-------|-------------|
@@ -114,7 +114,7 @@ ERROR         --RESET-->       IDLE
 ### Constructor
 
 ```typescript
-new WebRtcManager<TContext = unknown>(factory: WebRtcFactory, config?: WebRtcManagerConfig)
+new WebRTCManager<TContext = unknown>(factory: WebRTCFactory, config?: WebRTCManagerConfig)
 ```
 
 **Type Parameter:** `TContext` - Optional type for the `context` property (default: `unknown`)
@@ -134,27 +134,26 @@ interface Logger {
 
 Each method returns a string representation of the first argument, enabling patterns like `throw new Error(logger.error("msg"))`.
 
-### WebRtcFactory Interface
+### WebRTCFactory Interface
 
 ```typescript
-interface WebRtcFactory {
+interface WebRTCFactory {
   createPeerConnection(config?: RTCConfiguration): RTCPeerConnection;
   getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>;
   enumerateDevices(): Promise<MediaDeviceInfo[]>;
 }
 ```
 
-### WebRtcManagerConfig Interface
+### WebRTCManagerConfig Interface
 
 ```typescript
-interface WebRtcManagerConfig {
+interface WebRTCManagerConfig {
   peerConfig?: RTCConfiguration;      // ICE servers, certificates
   enableMicrophone?: boolean;         // Default: false
   dataChannelLabel?: string;          // Auto-create data channel
   autoReconnect?: boolean;            // Default: false
   maxReconnectAttempts?: number;      // Default: 5
   reconnectDelay?: number;            // Default: 1000ms
-  debug?: boolean;                    // Default: false
   logger?: Logger;                    // Custom logger, falls back to console
 }
 ```
@@ -172,7 +171,7 @@ interface GatherIceCandidatesOptions {
 
 | Property | Type | Description |
 |----------|------|-------------|
-| state | WebRtcState | Current FSM state |
+| state | WebRTCState | Current FSM state |
 | localStream | MediaStream \| null | Local audio stream |
 | remoteStream | MediaStream \| null | Remote audio stream |
 | dataChannels | ReadonlyMap<string, RTCDataChannel> | Active data channels |
@@ -223,7 +222,7 @@ interface GatherIceCandidatesOptions {
 
 | Method | Signature | Description |
 |--------|-----------|-------------|
-| on | `(event: keyof WebRtcEvents, handler: (data: any) => void): () => void` | Subscribe to specific event |
+| on | `(event: keyof WebRTCEvents, handler: (data: any) => void): () => void` | Subscribe to specific event |
 | subscribe | `(handler: (state: OverallState) => void): () => void` | Subscribe to overall state (Svelte compatible) |
 
 ### Utility Methods
@@ -236,7 +235,7 @@ interface GatherIceCandidatesOptions {
 
 | Constant | Value | Payload Type |
 |----------|-------|--------------|
-| EVENT_STATE_CHANGE | "state_change" | WebRtcState |
+| EVENT_STATE_CHANGE | "state_change" | WebRTCState |
 | EVENT_LOCAL_STREAM | "local_stream" | MediaStream \| null |
 | EVENT_REMOTE_STREAM | "remote_stream" | MediaStream \| null |
 | EVENT_DATA_CHANNEL_OPEN | "data_channel_open" | RTCDataChannel |
@@ -337,7 +336,7 @@ deno task serve:example # Run signaling server
 ### Minimal P2P Setup
 
 ```typescript
-const manager = new WebRtcManager(factory, { enableMicrophone: true });
+const manager = new WebRTCManager(factory, { enableMicrophone: true });
 await manager.initialize();
 await manager.connect();
 const offer = await manager.createOffer();
@@ -348,7 +347,7 @@ await manager.setLocalDescription(offer);
 ### With Data Channel
 
 ```typescript
-const manager = new WebRtcManager(factory, { dataChannelLabel: "chat" });
+const manager = new WebRTCManager(factory, { dataChannelLabel: "chat" });
 manager.on("data_channel_message", ({ data }) => console.log(data));
 // After connection...
 manager.sendData("chat", "Hello!");
@@ -358,7 +357,7 @@ manager.sendData("chat", "Hello!");
 
 ```svelte
 <script>
-const manager = new WebRtcManager(factory, config);
+const manager = new WebRTCManager(factory, config);
 // $manager reactive access to state
 </script>
 {$manager.state}
@@ -367,7 +366,7 @@ const manager = new WebRtcManager(factory, config);
 ### Auto-reconnection Handling
 
 ```typescript
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   autoReconnect: true,
   maxReconnectAttempts: 5,
   reconnectDelay: 1000,

package/API.md

@@ -4,7 +4,7 @@ Complete API documentation for `@marianmeres/webrtc`.
 
 ## Table of Contents
 
-- [WebRtcManager](#webrtcmanager)
+- [WebRTCManager](#webrtcmanager)
   - [Constructor](#constructor)
   - [Properties](#properties)
   - [Lifecycle Methods](#lifecycle-methods)
@@ -14,24 +14,24 @@ Complete API documentation for `@marianmeres/webrtc`.
   - [Event Methods](#event-methods)
   - [Utility Methods](#utility-methods)
 - [Types](#types)
-  - [WebRtcFactory](#webrtcfactory)
-  - [WebRtcManagerConfig](#webrtcmanagerconfig)
-  - [WebRtcState](#webrtcstate)
-  - [WebRtcFsmEvent](#webrtcfsmevent)
-  - [WebRtcEvents](#webrtcevents)
+  - [WebRTCFactory](#webrtcfactory)
+  - [WebRTCManagerConfig](#webrtcmanagerconfig)
+  - [WebRTCState](#webrtcstate)
+  - [WebRTCFsmEvent](#webrtcfsmevent)
+  - [WebRTCEvents](#webrtcevents)
 - [Event Constants](#event-constants)
 - [State Machine](#state-machine)
 
 ---
 
-## WebRtcManager
+## WebRTCManager
 
 The main class for managing WebRTC connections.
 
 ### Constructor
 
 ```typescript
-new WebRtcManager<TContext = unknown>(factory: WebRtcFactory, config?: WebRtcManagerConfig)
+new WebRTCManager<TContext = unknown>(factory: WebRTCFactory, config?: WebRTCManagerConfig)
 ```
 
 **Type Parameters:**
@@ -44,13 +44,13 @@ new WebRtcManager<TContext = unknown>(factory: WebRtcFactory, config?: WebRtcMan
 
 | Name | Type | Required | Description |
 |------|------|----------|-------------|
-| factory | `WebRtcFactory` | Yes | Factory for creating WebRTC primitives |
-| config | `WebRtcManagerConfig` | No | Configuration options |
+| factory | `WebRTCFactory` | Yes | Factory for creating WebRTC primitives |
+| config | `WebRTCManagerConfig` | No | Configuration options |
 
 **Example:**
 
 ```typescript
-import { WebRtcManager } from '@marianmeres/webrtc';
+import { WebRTCManager } from '@marianmeres/webrtc';
 
 const factory = {
   createPeerConnection: (config) => new RTCPeerConnection(config),
@@ -58,7 +58,7 @@ const factory = {
   enumerateDevices: () => navigator.mediaDevices.enumerateDevices(),
 };
 
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   peerConfig: { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] },
   enableMicrophone: true,
   autoReconnect: true,
@@ -72,12 +72,12 @@ const manager = new WebRtcManager(factory, {
 #### state
 
 ```typescript
-get state(): WebRtcState
+get state(): WebRTCState
 ```
 
 Returns the current state of the WebRTC connection.
 
-**Returns:** `WebRtcState` - One of: `IDLE`, `INITIALIZING`, `CONNECTING`, `CONNECTED`, `RECONNECTING`, `DISCONNECTED`, `ERROR`
+**Returns:** `WebRTCState` - One of: `IDLE`, `INITIALIZING`, `CONNECTING`, `CONNECTED`, `RECONNECTING`, `DISCONNECTED`, `ERROR`
 
 ---
 
@@ -145,12 +145,12 @@ User-defined context object for storing arbitrary data associated with this mana
 
 ```typescript
 // With type parameter for full type safety:
-const manager = new WebRtcManager<{ audioStream: MediaStream; sessionId: string }>(factory);
+const manager = new WebRTCManager<{ audioStream: MediaStream; sessionId: string }>(factory);
 manager.context = { audioStream: myStream, sessionId: '123' };
 manager.context.audioStream; // typed as MediaStream
 
 // Without type parameter (backwards compatible):
-const manager = new WebRtcManager(factory);
+const manager = new WebRTCManager(factory);
 manager.context = { anything: 'goes' };
 ```
 
@@ -614,9 +614,9 @@ stats?.forEach(report => {
 #### on()
 
 ```typescript
-on<K extends keyof WebRtcEvents>(
+on<K extends keyof WebRTCEvents>(
   event: K,
-  handler: (data: WebRtcEvents[K]) => void
+  handler: (data: WebRTCEvents[K]) => void
 ): () => void
 ```
 
@@ -626,7 +626,7 @@ Subscribe to a specific WebRTC event.
 
 | Name | Type | Description |
 |------|------|-------------|
-| event | `keyof WebRtcEvents` | Event name |
+| event | `keyof WebRTCEvents` | Event name |
 | handler | `function` | Callback receiving event data |
 
 **Returns:** `() => void` - Unsubscribe function
@@ -661,7 +661,7 @@ Subscribe to the overall state of the manager. Svelte store compatible - immedia
 
 ```typescript
 {
-  state: WebRtcState;
+  state: WebRTCState;
   localStream: MediaStream | null;
   remoteStream: MediaStream | null;
   dataChannels: ReadonlyMap<string, RTCDataChannel>;
@@ -675,7 +675,7 @@ Subscribe to the overall state of the manager. Svelte store compatible - immedia
 
 ```svelte
 <script>
-  const manager = new WebRtcManager(factory, config);
+  const manager = new WebRTCManager(factory, config);
 </script>
 
 <p>State: {$manager.state}</p>
@@ -736,20 +736,19 @@ import { clog } from '@marianmeres/clog';
 
 const logger = clog('WebRTC');
 
-const manager = new WebRtcManager(factory, {
-  debug: true,
+const manager = new WebRTCManager(factory, {
   logger: logger,
 });
 ```
 
 ---
 
-### WebRtcFactory
+### WebRTCFactory
 
 Interface for dependency injection of WebRTC primitives.
 
 ```typescript
-interface WebRtcFactory {
+interface WebRTCFactory {
   createPeerConnection(config?: RTCConfiguration): RTCPeerConnection;
   getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>;
   enumerateDevices(): Promise<MediaDeviceInfo[]>;
@@ -759,7 +758,7 @@ interface WebRtcFactory {
 **Browser Implementation:**
 
 ```typescript
-const factory: WebRtcFactory = {
+const factory: WebRTCFactory = {
   createPeerConnection: (config) => new RTCPeerConnection(config),
   getUserMedia: (constraints) => navigator.mediaDevices.getUserMedia(constraints),
   enumerateDevices: () => navigator.mediaDevices.enumerateDevices(),
@@ -768,12 +767,12 @@ const factory: WebRtcFactory = {
 
 ---
 
-### WebRtcManagerConfig
+### WebRTCManagerConfig
 
-Configuration options for WebRtcManager.
+Configuration options for WebRTCManager.
 
 ```typescript
-interface WebRtcManagerConfig {
+interface WebRTCManagerConfig {
   /** RTCConfiguration for ICE servers, certificates, etc. */
   peerConfig?: RTCConfiguration;
 
@@ -802,9 +801,6 @@ interface WebRtcManagerConfig {
     strategy: "ice-restart" | "full";
   }) => boolean;
 
-  /** Enable debug logging. Default: false */
-  debug?: boolean;
-
   /** Custom logger instance. If not provided, falls back to console. */
   logger?: Logger;
 }
@@ -812,12 +808,12 @@ interface WebRtcManagerConfig {
 
 ---
 
-### WebRtcState
+### WebRTCState
 
 Enum of possible connection states.
 
 ```typescript
-enum WebRtcState {
+enum WebRTCState {
   IDLE = "IDLE",
   INITIALIZING = "INITIALIZING",
   CONNECTING = "CONNECTING",
@@ -840,12 +836,12 @@ enum WebRtcState {
 
 ---
 
-### WebRtcFsmEvent
+### WebRTCFsmEvent
 
 Internal FSM events (for reference).
 
 ```typescript
-enum WebRtcFsmEvent {
+enum WebRTCFsmEvent {
   INIT = "initialize",
   CONNECT = "connect",
   CONNECTED = "connected",
@@ -858,13 +854,13 @@ enum WebRtcFsmEvent {
 
 ---
 
-### WebRtcEvents
+### WebRTCEvents
 
 Type definition for all events and their payloads.
 
 ```typescript
-interface WebRtcEvents {
-  state_change: WebRtcState;
+interface WebRTCEvents {
+  state_change: WebRTCState;
   local_stream: MediaStream | null;
   remote_stream: MediaStream | null;
   data_channel_open: RTCDataChannel;
@@ -883,11 +879,11 @@ interface WebRtcEvents {
 
 ## Event Constants
 
-Static event name constants on `WebRtcManager`.
+Static event name constants on `WebRTCManager`.
 
 | Constant | Value | Payload |
 |----------|-------|---------|
-| `EVENT_STATE_CHANGE` | `"state_change"` | `WebRtcState` |
+| `EVENT_STATE_CHANGE` | `"state_change"` | `WebRTCState` |
 | `EVENT_LOCAL_STREAM` | `"local_stream"` | `MediaStream \| null` |
 | `EVENT_REMOTE_STREAM` | `"remote_stream"` | `MediaStream \| null` |
 | `EVENT_DATA_CHANNEL_OPEN` | `"data_channel_open"` | `RTCDataChannel` |
@@ -903,7 +899,7 @@ Static event name constants on `WebRtcManager`.
 **Usage:**
 
 ```typescript
-manager.on(WebRtcManager.EVENT_ICE_CANDIDATE, (candidate) => {
+manager.on(WebRTCManager.EVENT_ICE_CANDIDATE, (candidate) => {
   // Send to remote peer
 });
 ```
@@ -1003,7 +999,7 @@ Use the `shouldReconnect` callback to suppress reconnection when the peer discon
 ```typescript
 let peerLeftIntentionally = false;
 
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   autoReconnect: true,
   shouldReconnect: ({ attempt, maxAttempts, strategy }) => {
     // Return false to suppress reconnection

package/README.md

@@ -23,7 +23,7 @@ npm install @marianmeres/webrtc
 
 ## High-Level Overview
 
-The `WebRtcManager` class handles the complete WebRTC connection lifecycle:
+The `WebRTCManager` class handles the complete WebRTC connection lifecycle:
 
 1. **Initialization**: Sets up RTCPeerConnection, media streams, and data channels
 2. **Connection Management**: Handles state transitions, reconnection, and cleanup
@@ -38,12 +38,12 @@ The manager doesn't handle the signaling transport layer - you're responsible fo
 ### Constructor
 
 ```typescript
-const manager = new WebRtcManager<TContext>(factory, config);
+const manager = new WebRTCManager<TContext>(factory, config);
 ```
 
 - `TContext`: Optional type parameter for the `context` property (default: `unknown`)
 
-- `factory`: Object implementing `WebRtcFactory` interface (provides `createPeerConnection`, `getUserMedia`, `enumerateDevices`)
+- `factory`: Object implementing `WebRTCFactory` interface (provides `createPeerConnection`, `getUserMedia`, `enumerateDevices`)
 - `config`: Optional configuration object
 
 **Configuration Options:**
@@ -55,13 +55,12 @@ const manager = new WebRtcManager<TContext>(factory, config);
 - `reconnectDelay`: Initial reconnection delay in ms (default: 1000)
 - `fullReconnectTimeout`: Timeout in ms for full reconnection to succeed (default: 30000)
 - `shouldReconnect`: Callback to control whether reconnection should proceed (see below)
-- `debug`: Enable debug logging (default: false)
 - `logger`: Custom logger instance implementing `Logger` interface (default: console)
 
 ### State and Properties
 
 ```typescript
-manager.state                 // Current WebRtcState
+manager.state                 // Current WebRTCState
 manager.localStream           // MediaStream | null
 manager.remoteStream          // MediaStream | null
 manager.dataChannels          // ReadonlyMap<string, RTCDataChannel>
@@ -147,7 +146,7 @@ You might want to keep `enableMicrophone: false` even when your application uses
 To use your own audio stream, access the `peerConnection` directly and add tracks after initialization:
 
 ```typescript
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   peerConfig: { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] },
   enableMicrophone: false, // We'll handle the audio stream ourselves
 });
@@ -186,7 +185,7 @@ const offer = await manager.createOffer();
 
 ```typescript
 // Subscribe to specific event
-const unsub = manager.on(WebRtcManager.EVENT_STATE_CHANGE, (state) => {
+const unsub = manager.on(WebRTCManager.EVENT_STATE_CHANGE, (state) => {
   console.log('State changed:', state);
 });
 
@@ -218,7 +217,7 @@ When `autoReconnect` is enabled, you can use the `shouldReconnect` callback to c
 ```typescript
 let peerLeftIntentionally = false;
 
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   autoReconnect: true,
   shouldReconnect: ({ attempt, maxAttempts, strategy }) => {
     if (peerLeftIntentionally) {
@@ -229,7 +228,7 @@ const manager = new WebRtcManager(factory, {
 });
 
 // Listen for "goodbye" message from peer before they disconnect
-manager.on(WebRtcManager.EVENT_DATA_CHANNEL_MESSAGE, ({ data }) => {
+manager.on(WebRTCManager.EVENT_DATA_CHANNEL_MESSAGE, ({ data }) => {
   const msg = JSON.parse(data);
   if (msg.type === 'bye') {
     peerLeftIntentionally = true;
@@ -260,7 +259,7 @@ When a full reconnection is triggered, the manager will:
 **Important:** The manager cannot automatically complete the signaling handshake for full reconnections. You must listen for the `reconnecting` event and re-establish signaling when the strategy is `'full'`:
 
 ```typescript
-manager.on(WebRtcManager.EVENT_RECONNECTING, async ({ attempt, strategy }) => {
+manager.on(WebRTCManager.EVENT_RECONNECTING, async ({ attempt, strategy }) => {
   console.log(`Reconnecting (attempt ${attempt}, strategy: ${strategy})`);
 
   if (strategy === 'full') {
@@ -280,7 +279,7 @@ If the full reconnection doesn't reach `CONNECTED` state within `fullReconnectTi
 ### Basic Usage (Vanilla JavaScript)
 
 ```typescript
-import { WebRtcManager, WebRtcState } from '@marianmeres/webrtc';
+import { WebRTCManager, WebRTCState } from '@marianmeres/webrtc';
 
 // Create factory (browser implementation)
 const factory = {
@@ -290,7 +289,7 @@ const factory = {
 };
 
 // Create manager
-const manager = new WebRtcManager(factory, {
+const manager = new WebRTCManager(factory, {
   peerConfig: {
     iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
   },
@@ -299,12 +298,12 @@ const manager = new WebRtcManager(factory, {
 });
 
 // Subscribe to events
-manager.on(WebRtcManager.EVENT_ICE_CANDIDATE, (candidate) => {
+manager.on(WebRTCManager.EVENT_ICE_CANDIDATE, (candidate) => {
   // Send candidate to remote peer via your signaling channel
   signalingChannel.send({ type: 'candidate', candidate });
 });
 
-manager.on(WebRtcManager.EVENT_REMOTE_STREAM, (stream) => {
+manager.on(WebRTCManager.EVENT_REMOTE_STREAM, (stream) => {
   // Attach remote stream to audio element
   audioElement.srcObject = stream;
 });
@@ -332,7 +331,7 @@ signalingChannel.onmessage = async (msg) => {
 
 ```svelte
 <script>
-  import { WebRtcManager, WebRtcState } from '@marianmeres/webrtc';
+  import { WebRTCManager, WebRTCState } from '@marianmeres/webrtc';
   import { onMount } from 'svelte';
 
   const factory = {
@@ -341,7 +340,7 @@ signalingChannel.onmessage = async (msg) => {
     enumerateDevices: () => navigator.mediaDevices.enumerateDevices(),
   };
 
-  const manager = new WebRtcManager(factory, {
+  const manager = new WebRTCManager(factory, {
     peerConfig: {
       iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
     },
@@ -356,7 +355,7 @@ signalingChannel.onmessage = async (msg) => {
 
   onMount(() => {
     const unsubDevices = manager.on(
-      WebRtcManager.EVENT_DEVICE_CHANGED,
+      WebRTCManager.EVENT_DEVICE_CHANGED,
       (devs) => devices = devs
     );
 
@@ -398,14 +397,14 @@ signalingChannel.onmessage = async (msg) => {
 ### Complete Peer-to-Peer Example
 
 ```typescript
-import { WebRtcManager } from '@marianmores/webrtc';
+import { WebRTCManager } from '@marianmores/webrtc';
 
 class P2PConnection {
-  manager: WebRtcManager;
+  manager: WebRTCManager;
   signalingChannel: WebSocket;
 
   constructor(signalingUrl: string) {
-    this.manager = new WebRtcManager(
+    this.manager = new WebRTCManager(
       {
         createPeerConnection: (config) => new RTCPeerConnection(config),
         getUserMedia: (constraints) => navigator.mediaDevices.getUserMedia(constraints),
@@ -446,7 +445,7 @@ class P2PConnection {
 
   setupManagerEvents() {
     // Send ICE candidates to remote peer
-    this.manager.on(WebRtcManager.EVENT_ICE_CANDIDATE, (candidate) => {
+    this.manager.on(WebRTCManager.EVENT_ICE_CANDIDATE, (candidate) => {
       this.signalingChannel.send(JSON.stringify({
         type: 'candidate',
         candidate,
@@ -454,18 +453,18 @@ class P2PConnection {
     });
 
     // Handle remote audio stream
-    this.manager.on(WebRtcManager.EVENT_REMOTE_STREAM, (stream) => {
+    this.manager.on(WebRTCManager.EVENT_REMOTE_STREAM, (stream) => {
       const audio = document.getElementById('remote-audio') as HTMLAudioElement;
       audio.srcObject = stream;
     });
 
     // Handle data channel messages
-    this.manager.on(WebRtcManager.EVENT_DATA_CHANNEL_MESSAGE, ({ data }) => {
+    this.manager.on(WebRTCManager.EVENT_DATA_CHANNEL_MESSAGE, ({ data }) => {
       console.log('Received message:', data);
     });
 
     // Handle reconnection
-    this.manager.on(WebRtcManager.EVENT_RECONNECTING, ({ attempt, strategy }) => {
+    this.manager.on(WebRTCManager.EVENT_RECONNECTING, ({ attempt, strategy }) => {
       console.log(`Reconnecting (attempt ${attempt}, strategy: ${strategy})`);
       if (strategy === 'full') {
         // For full reconnection, we need to re-do the signaling handshake

package/dist/types.d.ts

@@ -10,10 +10,10 @@ export interface Logger {
     error: (...args: any[]) => any;
 }
 /**
- * Configuration options for WebRtcManager.
+ * Configuration options for WebRTCManager.
  * All options are optional with sensible defaults.
  */
-export interface WebRtcManagerConfig {
+export interface WebRTCManagerConfig {
     /** Initial peer configuration (ICE servers, etc.) */
     peerConfig?: RTCConfiguration;
     /** Whether to enable microphone initially. Defaults to false. */
@@ -42,8 +42,6 @@ export interface WebRtcManagerConfig {
         /** Strategy that will be used: "ice-restart" for first attempts, "full" for later */
         strategy: "ice-restart" | "full";
     }) => boolean;
-    /** Enable debug logging. Defaults to false. */
-    debug?: boolean;
     /** Custom logger instance. If not provided, falls back to console. */
     logger?: Logger;
 }
@@ -51,7 +49,7 @@ export interface WebRtcManagerConfig {
  * Factory interface for creating WebRTC primitives.
  * Allows dependency injection of browser APIs for testing and flexibility.
  */
-export interface WebRtcFactory {
+export interface WebRTCFactory {
     /**
      * Creates a new RTCPeerConnection instance.
      * @param config - Optional RTCConfiguration for ICE servers, certificates, etc.
@@ -74,7 +72,7 @@ export interface WebRtcFactory {
  * Possible states of the WebRTC connection lifecycle.
  * The manager transitions between these states based on connection events.
  */
-export declare enum WebRtcState {
+export declare enum WebRTCState {
     /** Initial state, no resources allocated. */
     IDLE = "IDLE",
     /** Creating peer connection and setting up media tracks. */
@@ -94,7 +92,7 @@ export declare enum WebRtcState {
  * Internal FSM events that trigger state transitions.
  * These events are dispatched internally by the manager methods.
  */
-export declare enum WebRtcFsmEvent {
+export declare enum WebRTCFsmEvent {
     /** Triggers transition from IDLE to INITIALIZING. */
     INIT = "initialize",
     /** Triggers transition to CONNECTING state. */
@@ -114,9 +112,9 @@ export declare enum WebRtcFsmEvent {
  * Type definitions for all WebRTC manager events and their payloads.
  * Use with the `on()` method to subscribe to specific events.
  */
-export interface WebRtcEvents {
-    /** Emitted when connection state changes. Payload: the new WebRtcState. */
-    state_change: WebRtcState;
+export interface WebRTCEvents {
+    /** Emitted when connection state changes. Payload: the new WebRTCState. */
+    state_change: WebRTCState;
     /** Emitted when local media stream changes. Payload: MediaStream or null if disabled. */
     local_stream: MediaStream | null;
     /** Emitted when remote media stream is received. Payload: MediaStream or null. */