npm - @hyperttp/transport-webrtc - Versions diffs - 0.2.1 → 0.3.0 - Mend

@hyperttp/transport-webrtc 0.2.1 → 0.3.0

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Files changed (12) hide show

package/README.md CHANGED Viewed

@@ -168,3 +168,45 @@ the signal aborts; a late authorization result cannot create a new session.
 Core forwards these options unchanged. The high-level `server.webrtc.signal()`
 facade additionally injects typed application services into both callbacks.
 The default remains `/signal` without authorization for backward compatibility.
+## Server sessions and signaling ownership
+A server signaling authorization result may carry a trusted owner identity:
+```ts
+const transport = new WebRtcTransport({
+  localStream: async (sessionId, context) => {
+    // context.ownerId and context.authorization come from authorize(), not the wire.
+    return mediaForOwner(context.ownerId, context.signal);
+  },
+  disposeLocalStream: (sessionId, stream) => releaseMedia(sessionId, stream),
+});
+await transport.listen({
+  signaling: {
+    path: "/signal",
+    authorize: async (request) => {
+      const identity = await authenticate(request);
+      return identity
+        ? { ownerId: identity.id, context: identity }
+        : false;
+    },
+  },
+});
+```
+`ownerId` is stored when the session is created. A later POST or DELETE must be
+authorized as the same owner. Session identifiers, headers and signaling bodies
+are never used as owner identity. An authorization callback that returns `true`
+creates an anonymous session (`ownerId` is `undefined`); use the structured result
+when ownership isolation is required.
+Use `closeServerSession(sessionId)` for idempotent application cleanup. The
+`onServerSession` listener receives `created` and `closed` events, including the
+trusted authorization context and an abort signal. `WebRtcTransport.close()`
+closes all listener-owned sessions and emits the corresponding close events.
+The server currently supports **full-ICE-only signaling**. It gathers candidates
+before returning the SDP answer. Trickle ICE (`candidate` messages after the
+initial offer/answer) is not implemented; an ICE gathering timeout fails the
+signaling request rather than returning a partial answer.

package/dist/index.d.ts CHANGED Viewed

@@ -1,4 +1,4 @@
 export { WebRtcTransport } from "./webrtc.js";
 export type { WebRtcTransportOptions, WebRtcTransportServer, WebRtcServerSessionView, } from "./webrtc.js";
-export type { WebRtcSessionView, WebRtcTrackEvent, WebRtcMediaController, WebRtcTransportRequest, } from "./types/types.js";
+export type { WebRtcSessionView, WebRtcServerSessionContext, WebRtcServerSessionEvent, WebRtcTrackEvent, WebRtcMediaController, WebRtcTransportRequest, } from "./types/types.js";
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map CHANGED Viewed

	@@ -1 +1 @@
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package/dist/index.js.map CHANGED Viewed

@@ -1,7 +1,7 @@
 {
   "version": 3,
   "sources": ["../src/index.ts"],
-  "sourcesContent": ["export { WebRtcTransport } from \"./webrtc.js\";\nexport type {\n  WebRtcTransportOptions,\n  WebRtcTransportServer,\n  WebRtcServerSessionView,\n} from \"./webrtc.js\";\nexport type {\n  WebRtcSessionView,\n  WebRtcTrackEvent,\n  WebRtcMediaController,\n  WebRtcTransportRequest,\n} from \"./types/types.js\";\n"],
+  "sourcesContent": ["export { WebRtcTransport } from \"./webrtc.js\";\nexport type {\n  WebRtcTransportOptions,\n  WebRtcTransportServer,\n  WebRtcServerSessionView,\n} from \"./webrtc.js\";\nexport type {\n  WebRtcSessionView,\n  WebRtcServerSessionContext,\n  WebRtcServerSessionEvent,\n  WebRtcTrackEvent,\n  WebRtcMediaController,\n  WebRtcTransportRequest,\n} from \"./types/types.js\";\n"],
   "mappings": "AAAA,OAAS,mBAAAA,MAAuB",
   "names": ["WebRtcTransport"]
 }

package/dist/types/types.d.ts CHANGED Viewed

@@ -53,6 +53,16 @@ export interface WebRtcSessionView {
     readonly localStream?: WeriftMediaStream;
     readonly media: WebRtcMediaController;
 }
+export interface WebRtcServerSessionContext {
+    readonly sessionId: string;
+    /** Undefined only for explicitly anonymous/legacy boolean authorization. */
+    readonly ownerId: string | undefined;
+    readonly authorization: unknown;
+    readonly signal: AbortSignal;
+}
+export interface WebRtcServerSessionEvent extends WebRtcServerSessionContext {
+    readonly type: "created" | "closed";
+}
 export interface WebRtcTransportOptions {
     readonly iceServers?: RTCIceServer[];
     readonly iceGatheringTimeoutMs?: number;
@@ -64,7 +74,7 @@ export interface WebRtcTransportOptions {
     readonly createPeerConnection?: (configuration: RTCConfiguration) => WeriftPeerConnection;
     readonly createMediaStream?: () => WeriftMediaStream;
     readonly disposeLocalStream?: (sessionId: string, stream: WeriftMediaStream) => void;
-    readonly localStream?: WeriftMediaStream | ((sessionId: string) => WeriftMediaStream | undefined);
+    readonly localStream?: WeriftMediaStream | ((sessionId: string, context: WebRtcServerSessionContext) => WeriftMediaStream | undefined | Promise<WeriftMediaStream | undefined>);
 }
 export interface PendingResponse {
     resolve: (response: TransportResponse) => void;
@@ -89,6 +99,10 @@ export interface ClientSession {
 }
 export interface ServerSession {
     negotiating?: boolean;
+    context: WebRtcServerSessionContext;
+    controller: AbortController;
+    closing?: Promise<void>;
+    release: () => void;
     id: string;
     peerConnection: WeriftPeerConnection;
     remoteStream: WeriftMediaStream;

package/dist/types/types.d.ts.map CHANGED Viewed

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package/dist/types/types.js.map CHANGED Viewed

@@ -1,7 +1,7 @@
 {
   "version": 3,
   "sources": ["../../src/types/types.ts"],
-  "sourcesContent": ["import type { LogLevel, TransportRequest, TransportResponse } from \"@hyperttp/types\";\nimport type {\n  MediaStream as WeriftMediaStream,\n  RTCPeerConnection as WeriftPeerConnection,\n  RTCDataChannel as WeriftDataChannel,\n  MediaStreamTrack as WeriftMediaStreamTrack,\n} from \"werift\";\n\nexport const WEBRTC_BYTES_MARKER = \"__hyperttp_webrtc_bytes_v1__\";\nexport const MAX_SIGNALING_BODY_BYTES = 1_000_000;\n\nexport type EncodedWebRtcBytes = {\n  [WEBRTC_BYTES_MARKER]: true;\n  base64: string;\n};\n\nexport type WebRtcAction = \"connect\" | \"stream\" | \"datachannel\";\n\nexport type WebRtcTransportRequest = TransportRequest & {\n  peerUrl?: string;\n  action?: WebRtcAction;\n  iceServers?: RTCIceServer[];\n  localStream?: WeriftMediaStream;\n  receiveAudio?: boolean;\n  receiveVideo?: boolean;\n  channelLabel?: string;\n  timeout?: number;\n};\n\nexport interface WebRtcTrackEvent {\n  readonly side: \"client\" | \"server\";\n  readonly sessionId: string;\n  readonly peerUrl?: string;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly track: WeriftMediaStreamTrack;\n  readonly stream: WeriftMediaStream;\n  readonly streams: readonly WeriftMediaStream[];\n}\n\nexport interface WebRtcMediaController {\n  readonly localStream: WeriftMediaStream | undefined;\n  readonly remoteStream: WeriftMediaStream;\n\n  addTrack(track: WeriftMediaStreamTrack, stream?: WeriftMediaStream): Promise<void>;\n  replaceTrack(kind: \"audio\" | \"video\", track: WeriftMediaStreamTrack | null): Promise<void>;\n  removeTrack(track: WeriftMediaStreamTrack): Promise<void>;\n  close(): Promise<void>;\n\n  on(\n    event: \"track\",\n    listener: (e: { track: WeriftMediaStreamTrack; stream: WeriftMediaStream }) => void,\n  ): () => void;\n  on(event: \"trackEnded\", listener: (e: { track: WeriftMediaStreamTrack }) => void): () => void;\n  on(event: \"stream\", listener: (e: { stream: WeriftMediaStream }) => void): () => void;\n}\n\nexport interface WebRtcSessionView {\n  readonly id: string;\n  readonly peerUrl: string;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly dataChannel: WeriftDataChannel;\n  readonly remoteStream: WeriftMediaStream;\n  readonly localStream?: WeriftMediaStream;\n  readonly media: WebRtcMediaController;\n}\n\nexport interface WebRtcTransportOptions {\n  readonly iceServers?: RTCIceServer[];\n  readonly iceGatheringTimeoutMs?: number;\n  readonly signalingTimeoutMs?: number;\n  readonly connectionTimeoutMs?: number;\n  readonly requestTimeoutMs?: number;\n  readonly logger?: (level: LogLevel, message: string, meta?: unknown) => void;\n  readonly verbose?: boolean;\n  readonly createPeerConnection?: (configuration: RTCConfiguration) => WeriftPeerConnection;\n  readonly createMediaStream?: () => WeriftMediaStream;\n  readonly disposeLocalStream?: (sessionId: string, stream: WeriftMediaStream) => void;\n  readonly localStream?: WeriftMediaStream | ((sessionId: string) => WeriftMediaStream | undefined);\n}\n\nexport interface PendingResponse {\n  resolve: (response: TransportResponse) => void;\n  reject: (error: Error) => void;\n  timer: ReturnType<typeof setTimeout>;\n  signal?: AbortSignal;\n  abortListener?: () => void;\n}\n\nexport interface ClientSession {\n  id: string;\n  peerUrl: string;\n  signalingUrl: string;\n  peerConnection: WeriftPeerConnection;\n  dataChannel: WeriftDataChannel;\n  remoteStream: WeriftMediaStream;\n  localStream?: WeriftMediaStream;\n  localTrackIds: Set<string>;\n  pending: Map<string, PendingResponse>;\n  negotiation: Promise<void>;\n  closed: boolean;\n  mediaController: WebRtcMediaController;\n}\n\nexport interface ServerSession {\n  negotiating?: boolean;\n  id: string;\n  peerConnection: WeriftPeerConnection;\n  remoteStream: WeriftMediaStream;\n  localTrackIds: Set<string>;\n  localStream?: WeriftMediaStream;\n  localStreamResolved: boolean;\n  ownsLocalStream: boolean;\n  closed: boolean;\n}\n\nexport interface RTCIceServer {\n  credential?: string;\n  urls: string | string[];\n  username?: string;\n}\n"],
+  "sourcesContent": ["import type { LogLevel, TransportRequest, TransportResponse } from \"@hyperttp/types\";\nimport type {\n  MediaStream as WeriftMediaStream,\n  RTCPeerConnection as WeriftPeerConnection,\n  RTCDataChannel as WeriftDataChannel,\n  MediaStreamTrack as WeriftMediaStreamTrack,\n} from \"werift\";\n\nexport const WEBRTC_BYTES_MARKER = \"__hyperttp_webrtc_bytes_v1__\";\nexport const MAX_SIGNALING_BODY_BYTES = 1_000_000;\n\nexport type EncodedWebRtcBytes = {\n  [WEBRTC_BYTES_MARKER]: true;\n  base64: string;\n};\n\nexport type WebRtcAction = \"connect\" | \"stream\" | \"datachannel\";\n\nexport type WebRtcTransportRequest = TransportRequest & {\n  peerUrl?: string;\n  action?: WebRtcAction;\n  iceServers?: RTCIceServer[];\n  localStream?: WeriftMediaStream;\n  receiveAudio?: boolean;\n  receiveVideo?: boolean;\n  channelLabel?: string;\n  timeout?: number;\n};\n\nexport interface WebRtcTrackEvent {\n  readonly side: \"client\" | \"server\";\n  readonly sessionId: string;\n  readonly peerUrl?: string;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly track: WeriftMediaStreamTrack;\n  readonly stream: WeriftMediaStream;\n  readonly streams: readonly WeriftMediaStream[];\n}\n\nexport interface WebRtcMediaController {\n  readonly localStream: WeriftMediaStream | undefined;\n  readonly remoteStream: WeriftMediaStream;\n\n  addTrack(track: WeriftMediaStreamTrack, stream?: WeriftMediaStream): Promise<void>;\n  replaceTrack(kind: \"audio\" | \"video\", track: WeriftMediaStreamTrack | null): Promise<void>;\n  removeTrack(track: WeriftMediaStreamTrack): Promise<void>;\n  close(): Promise<void>;\n\n  on(\n    event: \"track\",\n    listener: (e: { track: WeriftMediaStreamTrack; stream: WeriftMediaStream }) => void,\n  ): () => void;\n  on(event: \"trackEnded\", listener: (e: { track: WeriftMediaStreamTrack }) => void): () => void;\n  on(event: \"stream\", listener: (e: { stream: WeriftMediaStream }) => void): () => void;\n}\n\nexport interface WebRtcSessionView {\n  readonly id: string;\n  readonly peerUrl: string;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly dataChannel: WeriftDataChannel;\n  readonly remoteStream: WeriftMediaStream;\n  readonly localStream?: WeriftMediaStream;\n  readonly media: WebRtcMediaController;\n}\n\nexport interface WebRtcServerSessionContext {\n  readonly sessionId: string;\n  /** Undefined only for explicitly anonymous/legacy boolean authorization. */\n  readonly ownerId: string | undefined;\n  readonly authorization: unknown;\n  readonly signal: AbortSignal;\n}\nexport interface WebRtcServerSessionEvent extends WebRtcServerSessionContext {\n  readonly type: \"created\" | \"closed\";\n}\n\nexport interface WebRtcTransportOptions {\n  readonly iceServers?: RTCIceServer[];\n  readonly iceGatheringTimeoutMs?: number;\n  readonly signalingTimeoutMs?: number;\n  readonly connectionTimeoutMs?: number;\n  readonly requestTimeoutMs?: number;\n  readonly logger?: (level: LogLevel, message: string, meta?: unknown) => void;\n  readonly verbose?: boolean;\n  readonly createPeerConnection?: (configuration: RTCConfiguration) => WeriftPeerConnection;\n  readonly createMediaStream?: () => WeriftMediaStream;\n  readonly disposeLocalStream?: (sessionId: string, stream: WeriftMediaStream) => void;\n  readonly localStream?:\n    | WeriftMediaStream\n    | ((\n        sessionId: string,\n        context: WebRtcServerSessionContext,\n      ) => WeriftMediaStream | undefined | Promise<WeriftMediaStream | undefined>);\n}\n\nexport interface PendingResponse {\n  resolve: (response: TransportResponse) => void;\n  reject: (error: Error) => void;\n  timer: ReturnType<typeof setTimeout>;\n  signal?: AbortSignal;\n  abortListener?: () => void;\n}\n\nexport interface ClientSession {\n  id: string;\n  peerUrl: string;\n  signalingUrl: string;\n  peerConnection: WeriftPeerConnection;\n  dataChannel: WeriftDataChannel;\n  remoteStream: WeriftMediaStream;\n  localStream?: WeriftMediaStream;\n  localTrackIds: Set<string>;\n  pending: Map<string, PendingResponse>;\n  negotiation: Promise<void>;\n  closed: boolean;\n  mediaController: WebRtcMediaController;\n}\n\nexport interface ServerSession {\n  negotiating?: boolean;\n  context: WebRtcServerSessionContext;\n  controller: AbortController;\n  closing?: Promise<void>;\n  release: () => void;\n  id: string;\n  peerConnection: WeriftPeerConnection;\n  remoteStream: WeriftMediaStream;\n  localTrackIds: Set<string>;\n  localStream?: WeriftMediaStream;\n  localStreamResolved: boolean;\n  ownsLocalStream: boolean;\n  closed: boolean;\n}\n\nexport interface RTCIceServer {\n  credential?: string;\n  urls: string | string[];\n  username?: string;\n}\n"],
   "mappings": "AAQO,MAAMA,EAAsB,+BACtBC,EAA2B",
   "names": ["WEBRTC_BYTES_MARKER", "MAX_SIGNALING_BODY_BYTES"]
 }

package/dist/webrtc.d.ts CHANGED Viewed

@@ -1,30 +1,28 @@
-import type { HyperTransport, LogLevel, TransportListenOptions, TransportRequest, TransportResponse, TransportServer } from "@hyperttp/types";
-import { type WebRtcSessionView, type WebRtcTrackEvent, type WebRtcTransportOptions as InternalWebRtcTransportOptions } from "./types/types.js";
+import type { HyperTransport, TransportListenOptions, TransportRequest, TransportResponse, TransportServer } from "@hyperttp/types";
+import { type WebRtcServerSessionContext, type WebRtcServerSessionEvent, type WebRtcSessionView, type WebRtcTrackEvent, type WebRtcTransportOptions as InternalWebRtcTransportOptions } from "./types/types.js";
 import { MediaStream as WeriftMediaStream, RTCPeerConnection as WeriftPeerConnection } from "werift";
 declare module "@hyperttp/types" {
     interface TransportServer {
-        readonly webrtc?: Pick<WebRtcTransport, "getSession" | "getServerSession" | "onTrack">;
+        readonly webrtc?: Pick<WebRtcTransport, "getSession" | "getServerSession" | "onTrack" | "closeServerSession" | "onServerSession">;
     }
 }
-export type WebRtcTransportOptions = Omit<InternalWebRtcTransportOptions, "localStream" | "disposeLocalStream" | "createMediaStream"> & {
-    readonly localStream?: WeriftMediaStream | ((sessionId: string) => WeriftMediaStream | undefined);
-    readonly disposeLocalStream?: (sessionId: string, stream: WeriftMediaStream) => void;
-    readonly createMediaStream?: () => WeriftMediaStream;
-    readonly logger?: (level: LogLevel, message: string, meta?: unknown) => void;
-};
+export type WebRtcTransportOptions = InternalWebRtcTransportOptions;
 export interface WebRtcServerSessionView {
     readonly id: string;
+    readonly context: WebRtcServerSessionContext;
     readonly peerConnection: WeriftPeerConnection;
     readonly remoteStream: WeriftMediaStream;
     readonly localStream?: WeriftMediaStream;
 }
 export interface WebRtcTransportServer extends TransportServer {
-    readonly webrtc: Pick<WebRtcTransport, "getSession" | "getServerSession" | "onTrack">;
+    readonly webrtc: Pick<WebRtcTransport, "getSession" | "getServerSession" | "onTrack" | "closeServerSession" | "onServerSession">;
 }
 export declare class WebRtcTransport implements HyperTransport {
     private readonly sessions;
     private readonly connecting;
     private readonly serverSessions;
+    private readonly sessionListeners;
+    private readonly closingServerSessions;
     private readonly trackListeners;
     private readonly serverClosers;
     private requestSequence;
@@ -41,6 +39,8 @@ export declare class WebRtcTransport implements HyperTransport {
     constructor(options?: WebRtcTransportOptions);
     private log;
     onTrack(listener: (event: WebRtcTrackEvent) => void): () => void;
+    onServerSession(listener: (event: WebRtcServerSessionEvent) => void): () => void;
+    private emitServerSession;
     getServerSession(sessionId: string): WebRtcServerSessionView | undefined;
     getSession(peerUrl: string): WebRtcSessionView | undefined;
     execute(request: TransportRequest): Promise<TransportResponse>;
@@ -66,7 +66,8 @@ export declare class WebRtcTransport implements HyperTransport {
     private attachServerPeerEvents;
     private attachConfiguredServerStream;
     private disposeServerLocalStream;
-    private closeServerSession;
+    /** Idempotent: concurrent closes share peer/resource cleanup. */
+    closeServerSession(sessionId: string): Promise<void>;
     private emitTrack;
     private toSignalingUrl;
     private createId;

package/dist/webrtc.d.ts.map CHANGED Viewed

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package/dist/webrtc.js CHANGED Viewed

@@ -1,2 +1,2 @@
-var M=Object.defineProperty;var g=(T,e)=>M(T,"name",{value:e,configurable:!0});import{MAX_SIGNALING_BODY_BYTES as D}from"./types/types.js";import{abortError as v,decodeWireBody as k,encodeWireBody as R,errorFromUnknown as C}from"./utils/helper.js";import{createServer as A}from"node:http";import{MediaStream as O,RTCPeerConnection as x}from"werift";function $(T){return(T??[]).flatMap(t=>{const r=Array.isArray(t.urls)?t.urls:[t.urls];if(t.credential!==void 0&&typeof t.credential!="string")throw new TypeError("Werift currently requires ICE server credentials to be strings.");return r.map(o=>({urls:o,...t.username!==void 0?{username:t.username}:{},...t.credential!==void 0?{credential:t.credential}:{}}))})}g($,"normalizeIceServers");function N(T){const e={iceServers:$(T.iceServers),...T.iceTransportPolicy!==void 0?{iceTransportPolicy:T.iceTransportPolicy}:{}};return new x(e)}g(N,"createWeriftPeerConnection");function G(){return new O}g(G,"createWeriftMediaStream");class z{static{g(this,"WebRtcTransport")}sessions=new Map;connecting=new Map;serverSessions=new Map;trackListeners=new Set;serverClosers=new Set;requestSequence=0;closing=!1;pendingPeers=new Set;iceServers;iceGatheringTimeoutMs;signalingTimeoutMs;connectionTimeoutMs;requestTimeoutMs;createMediaStream;createPeerConnection;options;constructor(e={}){this.options=e,this.createPeerConnection=e.createPeerConnection??N,this.createMediaStream=e.createMediaStream??G,this.iceServers=e.iceServers??[{urls:"stun:stun.l.google.com:19302"}],this.iceGatheringTimeoutMs=e.iceGatheringTimeoutMs??1e4,this.signalingTimeoutMs=e.signalingTimeoutMs??1e4,this.connectionTimeoutMs=e.connectionTimeoutMs??15e3,this.requestTimeoutMs=e.requestTimeoutMs??15e3}log(e,t,r){if(this.options.logger){this.options.logger(e,t,r);return}if(!this.options.verbose&&e!=="error")return;const o=e==="debug"?console.debug:console[e];r===void 0?o(t):o(t,r)}onTrack(e){return this.trackListeners.add(e),()=>this.trackListeners.delete(e)}getServerSession(e){const t=this.serverSessions.get(e);if(!(!t||t.closed))return{id:t.id,peerConnection:t.peerConnection,remoteStream:t.remoteStream,...t.localStream?{localStream:t.localStream}:{}}}getSession(e){const t=this.sessions.get(e);if(!(!t||t.closed))return{id:t.id,peerUrl:t.peerUrl,peerConnection:t.peerConnection,dataChannel:t.dataChannel,remoteStream:t.remoteStream,media:t.mediaController,...t.localStream?{localStream:t.localStream}:{}}}async execute(e){if(this.closing)throw new Error("WebRTC transport is closed");const t=e,r=t.peerUrl??e.url;if(e.signal?.aborted)throw v(e.signal);const o=await this.getOrCreateSession(r,t);if(await this.ensureMedia(o,t),e.signal?.aborted)throw v(e.signal);return this.sendRequest(o,e,t.timeout)}async listen(e){if(this.closing)throw new Error("WebRTC transport is closed");const t=new Set,r=e.port??3e3,o=e.host??"127.0.0.1",n=e.signaling?.path??"/signal";if(!n.startsWith("/")||n.startsWith("//")||/[?#\\\s]/u.test(n)||Array.from(n).some(a=>{const l=a.codePointAt(0);return l<=31||l>=127&&l<=159})||new URL(n,"http://localhost").pathname!==n)throw new TypeError("WebRTC signaling path must be an absolute literal pathname without query or fragment");if(e.signal?.aborted)throw new Error("WebRtcTransport: server startup aborted");let s=!1,i;const c=new Set,d=A((a,l)=>{(async()=>{const u=g((h,p)=>{l.headersSent||l.destroyed||(l.writeHead(h,{"content-type":"application/json"}),l.end(JSON.stringify(p)))},"sendJson");if(s||this.closing){u(503,{error:"WebRTC server is closing"});return}let y;try{y=new URL(a.url??"/","http://localhost")}catch{u(400,{error:"Invalid request URL"});return}if(y.pathname!==n){u(404,{error:"Signaling endpoint not found"});return}if((a.method==="POST"||a.method==="DELETE")&&e.signaling?.authorize){const h=new AbortController,p=g(()=>h.abort(),"onDisconnect"),L=g(()=>{u(503,{error:"WebRTC signaling request aborted"})},"onAuthorizationAbort");a.once("aborted",p),l.once("close",p),h.signal.addEventListener("abort",L,{once:!0}),c.add(h);try{const b={};for(const[W,P]of Object.entries(a.headers))P!==void 0&&(b[W]=P);const I=await e.signaling.authorize({protocol:"webrtc",method:a.method,url:a.url??"/",headers:b,signal:h.signal});if(h.signal.aborted||a.aborted||l.destroyed)return;if(s||this.closing||e.signal?.aborted){u(503,{error:"WebRTC server is closing"});return}if(!I){u(403,{error:"Signaling forbidden"});return}}catch{u(403,{error:"Signaling forbidden"});return}finally{c.delete(h),a.off("aborted",p),l.off("close",p),h.signal.removeEventListener("abort",L)}}if(a.method==="DELETE"){const h=y.searchParams.get("sessionId");h&&t.has(h)&&(await this.closeServerSession(h),t.delete(h)),l.writeHead(204).end();return}if(a.method!=="POST"){u(405,{error:"Method not allowed"});return}let E="";try{for await(const h of a)if(E+=Buffer.isBuffer(h)?h.toString("utf8"):String(h),Buffer.byteLength(E,"utf8")>D){u(413,{error:"Signaling payload too large"});return}}catch(h){u(400,{error:C(h,"Failed to read body").message});return}let w,S;try{const h=JSON.parse(E);if(typeof h.sessionId!="string"||h.sessionId.length<1||h.sessionId.length>200||h.sdp?.type!=="offer"||typeof h.sdp.sdp!="string"){u(400,{error:"Expected sessionId and a valid SDP offer"});return}if(s||this.closing||e.signal?.aborted||a.aborted||l.destroyed)throw new Error("WebRTC signaling request aborted");const p=h.sessionId;if(S=this.serverSessions.get(p),S&&!t.has(p)){u(409,{error:"Session belongs to another listener"});return}if(S?.negotiating){u(409,{error:"Session negotiation already in progress"});return}!S||S.closed?(w=this.createPeerConnection({iceServers:this.iceServers}),S={id:p,peerConnection:w,remoteStream:this.createMediaStream(),localTrackIds:new Set,localStreamResolved:!1,ownsLocalStream:!1,closed:!1},t.add(p),this.serverSessions.set(p,S),this.attachServerPeerEvents(S,e.onRequest)):w=S.peerConnection,S.negotiating=!0,await this.setRemoteDescription(w,h.sdp),await this.attachConfiguredServerStream(S);const L=await w.createAnswer();await this.withDeadline(w.setLocalDescription(L),this.iceGatheringTimeoutMs),await this.waitForIceGatheringComplete(w,this.iceGatheringTimeoutMs);const b=w.localDescription;if(!b?.sdp||b.type!=="answer")throw new Error("Local SDP answer was not generated");if(s||S.closed)throw new Error("WebRTC session closed");S.negotiating=!1,u(200,b)}catch(h){if(this.log("error","[WebRtcTransport Server] \u041E\u0448\u0438\u0431\u043A\u0430 \u0441\u0438\u0433\u043D\u0430\u043B\u0438\u0437\u0430\u0446\u0438\u0438:",h),S)await this.closeServerSession(S.id);else if(w)try{await w.close()}catch{}u(400,{error:C(h,"Signaling failed").message})}})()});await new Promise((a,l)=>{const u=g(E=>{d.off("listening",y),l(E)},"onError"),y=g(()=>{d.off("error",u),a()},"onListening");d.once("error",u),d.once("listening",y),d.listen(r,o)}),this.log("info",`[WebRtcTransport] Listening on http://${o}:${r}${n}`);const m=g(()=>{f().catch(a=>{this.log("error","[WebRtcTransport] Server shutdown failed:",a)})},"onAbort"),f=g(()=>{if(i)return i;s=!0;for(const a of c)a.abort();return e.signal?.removeEventListener("abort",m),i=(async()=>{const a=d.listening?new Promise((l,u)=>{d.close(y=>y?u(y):l())}):Promise.resolve();await Promise.all([...t].map(l=>this.closeServerSession(l))),t.clear(),await a,this.serverClosers.delete(f)})(),i},"closeServer");return e.signal?.addEventListener("abort",m,{once:!0}),this.serverClosers.add(f),e.signal?.aborted&&!s&&await f(),{close:f,webrtc:{getSession:this.getSession.bind(this),getServerSession:this.getServerSession.bind(this),onTrack:this.onTrack.bind(this)}}}async close(){this.closing=!0,await Promise.all([...this.pendingPeers].map(e=>e.close().catch(()=>{}))),await Promise.allSettled(this.connecting.values()),await Promise.all([...this.sessions.values()].map(e=>this.disposeClientSession(e,!0))),await Promise.all([...this.serverClosers].map(e=>e()))}async destroy(){await this.close()}async getOrCreateSession(e,t){const r=this.sessions.get(e);if(r&&!r.closed&&r.dataChannel.readyState==="open"&&r.peerConnection.connectionState!=="failed"&&r.peerConnection.connectionState!=="closed")return r;r&&await this.disposeClientSession(r,!0);const o=this.connecting.get(e);if(o)return o;const n=this.createClientSession(e,t);this.connecting.set(e,n);try{const s=await n;if(this.closing||s.closed)throw await this.disposeClientSession(s,!0),new Error("WebRTC transport is closed");return this.sessions.set(e,s),s}finally{this.connecting.get(e)===n&&this.connecting.delete(e)}}async createClientSession(e,t){const r=this.toSignalingUrl(e),o=this.createId("session"),n=this.createPeerConnection({iceServers:t.iceServers??this.iceServers});this.pendingPeers.add(n);const s=n.createDataChannel(t.channelLabel??"hyperttp-webrtc-transport",{ordered:!0}),i={id:o,peerUrl:e,signalingUrl:r,peerConnection:n,dataChannel:s,remoteStream:this.createMediaStream(),...t.localStream?{localStream:t.localStream}:{},localTrackIds:new Set,pending:new Map,negotiation:Promise.resolve(),closed:!1,mediaController:{}},c=new Set,d=new Set,m=new Set;n.addEventListener("track",f=>{const{track:a}=f;i.remoteStream.getTracks().includes(a)||i.remoteStream.addTrack(a);for(const l of c)l({track:a,stream:i.remoteStream});for(const l of m)l({stream:i.remoteStream});a.addEventListener("ended",()=>{for(const l of d)l({track:a})})}),i.mediaController={localStream:i.localStream,remoteStream:i.remoteStream,addTrack:g((f,a)=>this.queueNegotiation(i,async()=>{n.addTrack(f,a??i.localStream??this.createMediaStream()),f.id&&i.localTrackIds.add(f.id),await this.negotiate(i)}),"addTrack"),replaceTrack:g(async(f,a)=>{const l=n.getSenders().find(u=>u.track&&u.track.kind===f);l&&await l.replaceTrack(a)},"replaceTrack"),removeTrack:g(f=>this.queueNegotiation(i,async()=>{const a=n.getSenders().find(l=>l.track===f);a&&(n.removeTrack(a),f.id&&i.localTrackIds.delete(f.id),await this.negotiate(i))}),"removeTrack"),close:g(async()=>{await this.disposeClientSession(i,!0)},"close"),on(f,a){return f==="track"&&c.add(a),f==="trackEnded"&&d.add(a),f==="stream"&&m.add(a),()=>{c.delete(a),d.delete(a),m.delete(a)}}};try{return this.attachClientPeerEvents(i),this.addLocalTracks(i,t.localStream),this.addReceiveTransceivers(i,t.receiveAudio,t.receiveVideo),await this.negotiate(i,t.signal),await this.waitForChannelOpen(s,this.connectionTimeoutMs,t.signal),i}catch(f){throw await this.disposeClientSession(i,!0),f}finally{this.pendingPeers.delete(n)}}attachClientPeerEvents(e){const{peerConnection:t,dataChannel:r}=e;t.addEventListener("track",o=>{const n=o.track;e.remoteStream.getTracks().some(s=>s.id===n.id)||e.remoteStream.addTrack(n),this.emitTrack({side:"client",sessionId:e.id,peerUrl:e.peerUrl,peerConnection:t,track:n,stream:e.remoteStream,streams:o.streams})}),t.addEventListener("connectionstatechange",()=>{(t.connectionState==="failed"||t.connectionState==="closed")&&this.disposeClientSession(e,!0)}),r.addEventListener("message",o=>{let n;try{n=JSON.parse(String(o.data))}catch{return}if(!n||typeof n!="object"||typeof n.requestId!="string")return;const s=e.pending.get(n.requestId);if(s)try{const i=k(n.body);this.finishPending(e,n.requestId),s.resolve({status:typeof n.status=="number"?n.status:200,statusText:typeof n.statusText=="string"?n.statusText:"OK",headers:n.headers&&typeof n.headers=="object"?n.headers:{},body:i,url:typeof n.url=="string"?n.url:e.peerUrl})}catch(i){this.finishPending(e,n.requestId),s.reject(C(i,"Invalid WebRTC response"))}}),r.addEventListener("close",()=>{this.rejectAllPending(e,new Error("WebRTC DataChannel closed")),this.disposeClientSession(e,!0)}),r.addEventListener("error",()=>{this.rejectAllPending(e,new Error("WebRTC DataChannel error"))})}queueNegotiation(e,t){const r=e.negotiation.catch(()=>{}).then(async()=>{if(e.closed)throw new Error("WebRTC session closed");try{await t()}catch(o){throw await this.disposeClientSession(e,!0),o}});return e.negotiation=r,r}async ensureMedia(e,t){!(t.localStream||t.receiveAudio||t.receiveVideo)||e.closed||await this.queueNegotiation(e,async()=>{let o=this.addLocalTracks(e,t.localStream);o=this.addReceiveTransceivers(e,t.receiveAudio,t.receiveVideo)||o,t.localStream&&(e.localStream=t.localStream,e.mediaController.localStream=t.localStream),o&&await this.negotiate(e,t.signal)})}addLocalTracks(e,t){if(!t)return!1;let r=!1;for(const o of t.getTracks())o.readyState!=="live"||e.localTrackIds.has(o.id)||(e.peerConnection.addTrack(o,t),e.localTrackIds.add(o.id),r=!0);return r}addReceiveTransceivers(e,t,r){const o=e.peerConnection;let n=!1;const s=g((i,c)=>{if(!c)return;o.getTransceivers().find(m=>m.receiver.track.kind===i&&(m.direction==="recvonly"||m.direction==="sendrecv"))||(o.addTransceiver(i,{direction:"recvonly"}),n=!0)},"ensureReceive");return s("audio",t),s("video",r),n}async negotiate(e,t){if(t?.aborted)throw v(t);const r=e.peerConnection;if(r.signalingState!=="stable")throw new Error(`Cannot negotiate WebRTC session in state ${r.signalingState}`);const o=await r.createOffer();await this.withDeadline(r.setLocalDescription(o),this.iceGatheringTimeoutMs,t),await this.waitForIceGatheringComplete(r,this.iceGatheringTimeoutMs);const n=r.localDescription;if(!n?.sdp||n.type!=="offer")throw new Error("Local SDP offer was not generated");const s=new AbortController,i=setTimeout(()=>s.abort(),this.signalingTimeoutMs),c=g(()=>s.abort(t?.reason),"abortListener");t?.addEventListener("abort",c,{once:!0}),t?.aborted&&c();try{const d=await fetch(e.signalingUrl,{method:"POST",headers:{"content-type":"application/json"},body:JSON.stringify({sessionId:e.id,sdp:n}),signal:s.signal});if(!d.ok){const f=await d.text();throw new Error(`Signaling HTTP error: ${d.status} ${d.statusText}${f?` \u2014 ${f}`:""}`)}const m=await d.json();if(m.type!=="answer"||typeof m.sdp!="string")throw new Error("Signaling server returned an invalid SDP answer");await this.setRemoteDescription(r,m)}catch(d){throw t?.aborted?v(t):s.signal.aborted?new Error(`Signaling timed out after ${this.signalingTimeoutMs} ms`):C(d,"WebRTC signaling failed")}finally{clearTimeout(i),t?.removeEventListener("abort",c)}}async setRemoteDescription(e,t){const r=new Set(e.dtlsTransports);try{await e.setRemoteDescription(t)}finally{const o=new Set(e.dtlsTransports);await Promise.all([...r].filter(n=>!o.has(n)).map(n=>n.stop()))}}async withDeadline(e,t,r){let o,n;try{return await Promise.race([e,new Promise((s,i)=>{o=setTimeout(()=>i(new Error(`WebRTC operation timed out after ${t} ms`)),t),n=g(()=>i(v(r)),"onAbort"),r?.addEventListener("abort",n,{once:!0}),r?.aborted&&n()})])}finally{clearTimeout(o),n&&r?.removeEventListener("abort",n)}}async waitForIceGatheringComplete(e,t){if(e.iceGatheringState==="complete")return!0;if(!await new Promise(o=>{let n=!1,s;const i=g(()=>{clearTimeout(s),e.removeEventListener("icegatheringstatechange",d)},"cleanup"),c=g(m=>{n||(n=!0,i(),o(m))},"finish"),d=g(()=>{e.iceGatheringState==="complete"&&c(!0)},"onStateChange");s=setTimeout(()=>c(!1),t),e.addEventListener("icegatheringstatechange",d),d()}))throw new Error(`ICE gathering timed out after ${t} ms`);return!0}async waitForChannelOpen(e,t,r){if(e.readyState!=="open"){if(r?.aborted)throw v(r);await new Promise((o,n)=>{let s=!1;const i=g(()=>{clearTimeout(l),e.removeEventListener("open",d),e.removeEventListener("error",m),e.removeEventListener("close",f),r?.removeEventListener("abort",a)},"cleanup"),c=g(u=>{s||(s=!0,i(),u?n(u):o())},"finish"),d=g(()=>c(),"onOpen"),m=g(()=>c(new Error("WebRTC DataChannel failed to open")),"onError"),f=g(()=>c(new Error("WebRTC DataChannel closed before opening")),"onClose"),a=g(()=>c(v(r)),"onAbort"),l=setTimeout(()=>c(new Error(`DataChannel open timeout after ${t} ms`)),t);e.addEventListener("open",d),e.addEventListener("error",m),e.addEventListener("close",f),r?.addEventListener("abort",a,{once:!0}),e.readyState==="open"?c():e.readyState==="closed"&&c(new Error("WebRTC DataChannel is closed"))})}}sendRequest(e,t,r){if(t.signal?.aborted)return Promise.reject(v(t.signal));if(e.dataChannel.readyState!=="open")return Promise.reject(new Error("WebRTC DataChannel is not open"));const o=this.createId(`request${++this.requestSequence}`),n=r&&r>0?r:this.requestTimeoutMs;return new Promise((s,i)=>{const c={resolve:s,reject:i,timer:setTimeout(()=>{this.finishPending(e,o),i(new Error(`WebRTC request timed out after ${n} ms`))},n),...t.signal?{signal:t.signal}:{}};t.signal&&(c.abortListener=()=>{this.finishPending(e,o),i(v(t.signal))},t.signal.addEventListener("abort",c.abortListener,{once:!0})),e.pending.set(o,c);try{e.dataChannel.send(JSON.stringify({requestId:o,method:t.method,url:t.url,headers:t.headers,body:R(t.body),protocol:t.protocol}))}catch(d){this.finishPending(e,o),i(C(d,"Failed to send WebRTC request"))}})}finishPending(e,t){const r=e.pending.get(t);r&&(clearTimeout(r.timer),r.signal&&r.abortListener&&r.signal.removeEventListener("abort",r.abortListener),e.pending.delete(t))}rejectAllPending(e,t){for(const[r,o]of e.pending)this.finishPending(e,r),o.reject(t)}async disposeClientSession(e,t){if(!e.closed){e.closed=!0,this.sessions.get(e.peerUrl)===e&&this.sessions.delete(e.peerUrl),this.rejectAllPending(e,new Error("WebRTC session closed"));try{e.dataChannel.close()}catch{}try{await e.peerConnection.close()}catch{}if(t)try{const r=new URL(e.signalingUrl);r.searchParams.set("sessionId",e.id);const o=new AbortController,n=setTimeout(()=>o.abort(),2e3);try{await fetch(r,{method:"DELETE",signal:o.signal})}finally{clearTimeout(n)}}catch{}}}attachServerPeerEvents(e,t){const r=e.peerConnection;r.addEventListener("datachannel",o=>{const n=o.channel;n.addEventListener("message",s=>{(async()=>{let i;try{const c=JSON.parse(String(s.data));if(i=typeof c.requestId=="string"?c.requestId:void 0,!i||typeof c.method!="string"||typeof c.url!="string")throw new Error("Invalid WebRTC transport request");const d={method:c.method,url:c.url,headers:c.headers&&typeof c.headers=="object"?{...c.headers,"x-webrtc-session-id":e.id}:{"x-webrtc-session-id":e.id},body:k(c.body),protocol:typeof c.protocol=="string"?c.protocol:"webrtc"};let m;if(t?m=await t(d):m={status:501,statusText:"Not Implemented",headers:{"content-type":"application/json"},body:{error:"No server request handler configured"},url:d.url},n.readyState!=="open")return;n.send(JSON.stringify({requestId:i,status:m.status,statusText:m.statusText??"",headers:m.headers,body:R(m.body),url:m.url??d.url}))}catch(c){if(this.log("error","[WebRtcTransport Server] Request failed:",c),i&&n.readyState==="open")try{n.send(JSON.stringify({requestId:i,status:500,statusText:"Internal Server Error",headers:{"content-type":"application/json"},body:{error:C(c,"Request handler failed").message}}))}catch{}}})()})}),r.addEventListener("track",o=>{const n=o.track;e.remoteStream.getTracks().some(s=>s.id===n.id)||e.remoteStream.addTrack(n),this.emitTrack({side:"server",sessionId:e.id,peerConnection:r,track:n,stream:e.remoteStream,streams:o.streams})}),r.addEventListener("connectionstatechange",()=>{(r.connectionState==="failed"||r.connectionState==="closed")&&this.closeServerSession(e.id)})}async attachConfiguredServerStream(e){if(e.closed)return;if(!e.localStreamResolved){const o=this.options.localStream,n=typeof o=="function",s=n?o(e.id):o;e.localStreamResolved=!0,s&&(e.localStream=s,e.ownsLocalStream=n)}const t=e.localStream;if(!t)return;const r=e.peerConnection;for(const o of t.getTracks())e.closed||o.readyState!=="live"||e.localTrackIds.has(o.id)||(r.addTrack(o,t),e.localTrackIds.add(o.id))}disposeServerLocalStream(e){const t=e.localStream,r=e.ownsLocalStream;if(delete e.localStream,e.ownsLocalStream=!1,!(!t||!r))try{this.options.disposeLocalStream?.(e.id,t)}catch{}finally{for(const o of t.getTracks())o.readyState!=="ended"&&o.stop()}}async closeServerSession(e){const t=this.serverSessions.get(e);if(!(!t||t.closed)){t.closed=!0,this.serverSessions.delete(e);try{await t.peerConnection.close()}catch{}finally{this.disposeServerLocalStream(t)}}}emitTrack(e){for(const t of this.trackListeners)try{t(e)}catch(r){this.log("error","[WebRtcTransport] onTrack listener failed:",r)}}toSignalingUrl(e){if(e.startsWith("webrtcs://"))return e.replace(/^webrtcs:\/\//,"https://");if(e.startsWith("webrtc://"))return e.replace(/^webrtc:\/\//,"http://");throw new Error(`Unsupported WebRTC signaling URL: ${e}`)}createId(e){return`${e}-${crypto.randomUUID()}`}}export{z as WebRtcTransport};
+var A=Object.defineProperty;var f=(T,e)=>A(T,"name",{value:e,configurable:!0});import{MAX_SIGNALING_BODY_BYTES as O}from"./types/types.js";import{abortError as y,decodeWireBody as D,encodeWireBody as x,errorFromUnknown as P}from"./utils/helper.js";import{createServer as $}from"node:http";import{MediaStream as N,RTCPeerConnection as j}from"werift";function G(T){return(T??[]).flatMap(t=>{const r=Array.isArray(t.urls)?t.urls:[t.urls];if(t.credential!==void 0&&typeof t.credential!="string")throw new TypeError("Werift currently requires ICE server credentials to be strings.");return r.map(o=>({urls:o,...t.username!==void 0?{username:t.username}:{},...t.credential!==void 0?{credential:t.credential}:{}}))})}f(G,"normalizeIceServers");function U(T){const e={iceServers:G(T.iceServers),...T.iceTransportPolicy!==void 0?{iceTransportPolicy:T.iceTransportPolicy}:{}};return new j(e)}f(U,"createWeriftPeerConnection");function F(){return new N}f(F,"createWeriftMediaStream");class Y{static{f(this,"WebRtcTransport")}sessions=new Map;connecting=new Map;serverSessions=new Map;sessionListeners=new Set;closingServerSessions=new Map;trackListeners=new Set;serverClosers=new Set;requestSequence=0;closing=!1;pendingPeers=new Set;iceServers;iceGatheringTimeoutMs;signalingTimeoutMs;connectionTimeoutMs;requestTimeoutMs;createMediaStream;createPeerConnection;options;constructor(e={}){this.options=e,this.createPeerConnection=e.createPeerConnection??U,this.createMediaStream=e.createMediaStream??F,this.iceServers=e.iceServers??[{urls:"stun:stun.l.google.com:19302"}],this.iceGatheringTimeoutMs=e.iceGatheringTimeoutMs??1e4,this.signalingTimeoutMs=e.signalingTimeoutMs??1e4,this.connectionTimeoutMs=e.connectionTimeoutMs??15e3,this.requestTimeoutMs=e.requestTimeoutMs??15e3}log(e,t,r){if(this.options.logger){this.options.logger(e,t,r);return}if(!this.options.verbose&&e!=="error")return;const o=e==="debug"?console.debug:console[e];r===void 0?o(t):o(t,r)}onTrack(e){return this.trackListeners.add(e),()=>this.trackListeners.delete(e)}onServerSession(e){return this.sessionListeners.add(e),()=>this.sessionListeners.delete(e)}emitServerSession(e,t){for(const r of this.sessionListeners)try{r({...e.context,type:t})}catch(o){this.log("error","WebRTC session listener failed",o)}}getServerSession(e){const t=this.serverSessions.get(e);if(!(!t||t.closed))return{id:t.id,context:t.context,peerConnection:t.peerConnection,remoteStream:t.remoteStream,...t.localStream?{localStream:t.localStream}:{}}}getSession(e){const t=this.sessions.get(e);if(!(!t||t.closed))return{id:t.id,peerUrl:t.peerUrl,peerConnection:t.peerConnection,dataChannel:t.dataChannel,remoteStream:t.remoteStream,media:t.mediaController,...t.localStream?{localStream:t.localStream}:{}}}async execute(e){if(this.closing)throw new Error("WebRTC transport is closed");const t=e,r=t.peerUrl??e.url;if(e.signal?.aborted)throw y(e.signal);const o=await this.getOrCreateSession(r,t);if(await this.ensureMedia(o,t),e.signal?.aborted)throw y(e.signal);return this.sendRequest(o,e,t.timeout)}async listen(e){if(this.closing)throw new Error("WebRTC transport is closed");const t=new Set,r=e.port??3e3,o=e.host??"127.0.0.1",n=e.signaling?.path??"/signal";if(!n.startsWith("/")||n.startsWith("//")||/[?#\\\s]/u.test(n)||Array.from(n).some(a=>{const l=a.codePointAt(0);return l<=31||l>=127&&l<=159})||new URL(n,"http://localhost").pathname!==n)throw new TypeError("WebRTC signaling path must be an absolute literal pathname without query or fragment");if(e.signal?.aborted)throw new Error("WebRtcTransport: server startup aborted");let s=!1,i;const c=new Set,h=$((a,l)=>{(async()=>{const g=f((d,p)=>{l.headersSent||l.destroyed||(l.writeHead(d,{"content-type":"application/json"}),l.end(JSON.stringify(p)))},"sendJson");if(s||this.closing){g(503,{error:"WebRTC server is closing"});return}let b;try{b=new URL(a.url??"/","http://localhost")}catch{g(400,{error:"Invalid request URL"});return}if(b.pathname!==n){g(404,{error:"Signaling endpoint not found"});return}let C,W;if((a.method==="POST"||a.method==="DELETE")&&e.signaling?.authorize){const d=new AbortController,p=f(()=>d.abort(),"onDisconnect"),E=f(()=>{g(503,{error:"WebRTC signaling request aborted"})},"onAuthorizationAbort");a.once("aborted",p),l.once("close",p),d.signal.addEventListener("abort",E,{once:!0}),c.add(d);try{const k={};for(const[I,L]of Object.entries(a.headers))L!==void 0&&(k[I]=L);const v=await e.signaling.authorize({protocol:"webrtc",method:a.method,url:a.url??"/",headers:k,signal:d.signal});if(d.signal.aborted||a.aborted||l.destroyed)return;if(s||this.closing||e.signal?.aborted){g(503,{error:"WebRTC server is closing"});return}if(typeof v=="object"&&v!==null){if(typeof v.ownerId!="string"||!v.ownerId.trim()){g(403,{error:"Invalid signaling identity"});return}C=v.ownerId,W=v.context}else if(v!==!0){g(403,{error:"Signaling forbidden"});return}}catch{g(403,{error:"Signaling forbidden"});return}finally{c.delete(d),a.off("aborted",p),l.off("close",p),d.signal.removeEventListener("abort",E)}}if(a.method==="DELETE"){const d=b.searchParams.get("sessionId"),p=d?this.serverSessions.get(d):void 0;if(p&&(!t.has(d)||p.context.ownerId!==C)){g(403,{error:"Signaling forbidden"});return}d&&t.has(d)&&(await this.closeServerSession(d),t.delete(d)),l.writeHead(204).end();return}if(a.method!=="POST"){g(405,{error:"Method not allowed"});return}let R="";try{for await(const d of a)if(R+=Buffer.isBuffer(d)?d.toString("utf8"):String(d),Buffer.byteLength(R,"utf8")>O){g(413,{error:"Signaling payload too large"});return}}catch(d){g(400,{error:P(d,"Failed to read body").message});return}let w,S;try{const d=JSON.parse(R);if(typeof d.sessionId!="string"||d.sessionId.length<1||d.sessionId.length>200||d.sdp?.type!=="offer"||typeof d.sdp.sdp!="string"){g(400,{error:"Expected sessionId and a valid SDP offer"});return}if(s||this.closing||e.signal?.aborted||a.aborted||l.destroyed)throw new Error("WebRTC signaling request aborted");const p=d.sessionId;if(S=this.serverSessions.get(p),S&&!t.has(p)){g(409,{error:"Session belongs to another listener"});return}if(S&&S.context.ownerId!==C){g(403,{error:"Signaling forbidden"});return}if(this.closingServerSessions.has(p)){g(409,{error:"Session is closing"});return}if(S?.negotiating){g(409,{error:"Session negotiation already in progress"});return}if(!S||S.closed){w=this.createPeerConnection({iceServers:this.iceServers});const M=new AbortController;S={controller:M,context:Object.freeze({sessionId:p,ownerId:C,authorization:W,signal:M.signal}),release:f(()=>t.delete(p),"release"),id:p,peerConnection:w,remoteStream:this.createMediaStream(),localTrackIds:new Set,localStreamResolved:!1,ownsLocalStream:!1,closed:!1},t.add(p),this.serverSessions.set(p,S),this.attachServerPeerEvents(S,e.onRequest),this.emitServerSession(S,"created")}else w=S.peerConnection;S.negotiating=!0;const E=S,k=f(()=>{l.writableEnded||this.closeServerSession(E.id)},"disconnect");l.once("close",k);const v=f(()=>g(503,{error:"WebRTC session closed"}),"onSessionAbort");if(E.controller.signal.addEventListener("abort",v,{once:!0}),l.once("finish",()=>{E.controller.signal.removeEventListener("abort",v),l.off("close",k)}),await this.setRemoteDescription(w,d.sdp),await this.attachConfiguredServerStream(S),S.closed)throw new Error("WebRTC session closed");const I=await w.createAnswer();if(await this.withDeadline(w.setLocalDescription(I),this.iceGatheringTimeoutMs),!await this.waitForIceGatheringComplete(w,this.iceGatheringTimeoutMs))throw new Error("Full ICE gathering timed out; trickle ICE is not supported");const L=w.localDescription;if(!L?.sdp||L.type!=="answer")throw new Error("Local SDP answer was not generated");if(s||S.closed)throw new Error("WebRTC session closed");S.negotiating=!1,g(200,L)}catch(d){if(this.log("error","[WebRtcTransport Server] \u041E\u0448\u0438\u0431\u043A\u0430 \u0441\u0438\u0433\u043D\u0430\u043B\u0438\u0437\u0430\u0446\u0438\u0438:",d),S)await this.closeServerSession(S.id);else if(w)try{await w.close()}catch{}g(400,{error:P(d,"Signaling failed").message})}})()});await new Promise((a,l)=>{const g=f(C=>{h.off("listening",b),l(C)},"onError"),b=f(()=>{h.off("error",g),a()},"onListening");h.once("error",g),h.once("listening",b),h.listen(r,o)}),this.log("info",`[WebRtcTransport] Listening on http://${o}:${r}${n}`);const u=f(()=>{m().catch(a=>{this.log("error","[WebRtcTransport] Server shutdown failed:",a)})},"onAbort"),m=f(()=>{if(i)return i;s=!0;for(const a of c)a.abort();return e.signal?.removeEventListener("abort",u),i=(async()=>{const a=h.listening?new Promise((l,g)=>{h.close(b=>b?g(b):l())}):Promise.resolve();await Promise.all([...t].map(l=>this.closeServerSession(l))),t.clear(),await a,this.serverClosers.delete(m)})(),i},"closeServer");return e.signal?.addEventListener("abort",u,{once:!0}),this.serverClosers.add(m),e.signal?.aborted&&!s&&await m(),{close:m,webrtc:{getSession:this.getSession.bind(this),getServerSession:this.getServerSession.bind(this),onTrack:this.onTrack.bind(this),closeServerSession:f(a=>t.has(a)?this.closeServerSession(a):Promise.resolve(),"closeServerSession"),onServerSession:this.onServerSession.bind(this)}}}async close(){this.closing=!0,await Promise.all([...this.pendingPeers].map(e=>e.close().catch(()=>{}))),await Promise.allSettled(this.connecting.values()),await Promise.all([...this.sessions.values()].map(e=>this.disposeClientSession(e,!0))),await Promise.all([...this.serverClosers].map(e=>e()))}async destroy(){await this.close()}async getOrCreateSession(e,t){const r=this.sessions.get(e);if(r&&!r.closed&&r.dataChannel.readyState==="open"&&r.peerConnection.connectionState!=="failed"&&r.peerConnection.connectionState!=="closed")return r;r&&await this.disposeClientSession(r,!0);const o=this.connecting.get(e);if(o)return o;const n=this.createClientSession(e,t);this.connecting.set(e,n);try{const s=await n;if(this.closing||s.closed)throw await this.disposeClientSession(s,!0),new Error("WebRTC transport is closed");return this.sessions.set(e,s),s}finally{this.connecting.get(e)===n&&this.connecting.delete(e)}}async createClientSession(e,t){const r=this.toSignalingUrl(e),o=this.createId("session"),n=this.createPeerConnection({iceServers:t.iceServers??this.iceServers});this.pendingPeers.add(n);const s=n.createDataChannel(t.channelLabel??"hyperttp-webrtc-transport",{ordered:!0}),i={id:o,peerUrl:e,signalingUrl:r,peerConnection:n,dataChannel:s,remoteStream:this.createMediaStream(),...t.localStream?{localStream:t.localStream}:{},localTrackIds:new Set,pending:new Map,negotiation:Promise.resolve(),closed:!1,mediaController:{}},c=new Set,h=new Set,u=new Set;n.addEventListener("track",m=>{const{track:a}=m;i.remoteStream.getTracks().includes(a)||i.remoteStream.addTrack(a);for(const l of c)l({track:a,stream:i.remoteStream});for(const l of u)l({stream:i.remoteStream});a.addEventListener("ended",()=>{for(const l of h)l({track:a})})}),i.mediaController={localStream:i.localStream,remoteStream:i.remoteStream,addTrack:f((m,a)=>this.queueNegotiation(i,async()=>{n.addTrack(m,a??i.localStream??this.createMediaStream()),m.id&&i.localTrackIds.add(m.id),await this.negotiate(i)}),"addTrack"),replaceTrack:f(async(m,a)=>{const l=n.getSenders().find(g=>g.track&&g.track.kind===m);l&&await l.replaceTrack(a)},"replaceTrack"),removeTrack:f(m=>this.queueNegotiation(i,async()=>{const a=n.getSenders().find(l=>l.track===m);a&&(n.removeTrack(a),m.id&&i.localTrackIds.delete(m.id),await this.negotiate(i))}),"removeTrack"),close:f(async()=>{await this.disposeClientSession(i,!0)},"close"),on(m,a){return m==="track"&&c.add(a),m==="trackEnded"&&h.add(a),m==="stream"&&u.add(a),()=>{c.delete(a),h.delete(a),u.delete(a)}}};try{return this.attachClientPeerEvents(i),this.addLocalTracks(i,t.localStream),this.addReceiveTransceivers(i,t.receiveAudio,t.receiveVideo),await this.negotiate(i,t.signal),await this.waitForChannelOpen(s,this.connectionTimeoutMs,t.signal),i}catch(m){throw await this.disposeClientSession(i,!0),m}finally{this.pendingPeers.delete(n)}}attachClientPeerEvents(e){const{peerConnection:t,dataChannel:r}=e;t.addEventListener("track",o=>{const n=o.track;e.remoteStream.getTracks().some(s=>s.id===n.id)||e.remoteStream.addTrack(n),this.emitTrack({side:"client",sessionId:e.id,peerUrl:e.peerUrl,peerConnection:t,track:n,stream:e.remoteStream,streams:o.streams})}),t.addEventListener("connectionstatechange",()=>{(t.connectionState==="failed"||t.connectionState==="closed")&&this.disposeClientSession(e,!0)}),r.addEventListener("message",o=>{let n;try{n=JSON.parse(String(o.data))}catch{return}if(!n||typeof n!="object"||typeof n.requestId!="string")return;const s=e.pending.get(n.requestId);if(s)try{const i=D(n.body);this.finishPending(e,n.requestId),s.resolve({status:typeof n.status=="number"?n.status:200,statusText:typeof n.statusText=="string"?n.statusText:"OK",headers:n.headers&&typeof n.headers=="object"?n.headers:{},body:i,url:typeof n.url=="string"?n.url:e.peerUrl})}catch(i){this.finishPending(e,n.requestId),s.reject(P(i,"Invalid WebRTC response"))}}),r.addEventListener("close",()=>{this.rejectAllPending(e,new Error("WebRTC DataChannel closed")),this.disposeClientSession(e,!0)}),r.addEventListener("error",()=>{this.rejectAllPending(e,new Error("WebRTC DataChannel error"))})}queueNegotiation(e,t){const r=e.negotiation.catch(()=>{}).then(async()=>{if(e.closed)throw new Error("WebRTC session closed");try{await t()}catch(o){throw await this.disposeClientSession(e,!0),o}});return e.negotiation=r,r}async ensureMedia(e,t){!(t.localStream||t.receiveAudio||t.receiveVideo)||e.closed||await this.queueNegotiation(e,async()=>{let o=this.addLocalTracks(e,t.localStream);o=this.addReceiveTransceivers(e,t.receiveAudio,t.receiveVideo)||o,t.localStream&&(e.localStream=t.localStream,e.mediaController.localStream=t.localStream),o&&await this.negotiate(e,t.signal)})}addLocalTracks(e,t){if(!t)return!1;let r=!1;for(const o of t.getTracks())o.readyState!=="live"||e.localTrackIds.has(o.id)||(e.peerConnection.addTrack(o,t),e.localTrackIds.add(o.id),r=!0);return r}addReceiveTransceivers(e,t,r){const o=e.peerConnection;let n=!1;const s=f((i,c)=>{if(!c)return;o.getTransceivers().find(u=>u.receiver.track.kind===i&&(u.direction==="recvonly"||u.direction==="sendrecv"))||(o.addTransceiver(i,{direction:"recvonly"}),n=!0)},"ensureReceive");return s("audio",t),s("video",r),n}async negotiate(e,t){if(t?.aborted)throw y(t);const r=e.peerConnection;if(r.signalingState!=="stable")throw new Error(`Cannot negotiate WebRTC session in state ${r.signalingState}`);const o=await r.createOffer();await this.withDeadline(r.setLocalDescription(o),this.iceGatheringTimeoutMs,t),await this.waitForIceGatheringComplete(r,this.iceGatheringTimeoutMs);const n=r.localDescription;if(!n?.sdp||n.type!=="offer")throw new Error("Local SDP offer was not generated");const s=new AbortController,i=setTimeout(()=>s.abort(),this.signalingTimeoutMs),c=f(()=>s.abort(t?.reason),"abortListener");t?.addEventListener("abort",c,{once:!0}),t?.aborted&&c();try{const h=await fetch(e.signalingUrl,{method:"POST",headers:{"content-type":"application/json"},body:JSON.stringify({sessionId:e.id,sdp:n}),signal:s.signal});if(!h.ok){const m=await h.text();throw new Error(`Signaling HTTP error: ${h.status} ${h.statusText}${m?` \u2014 ${m}`:""}`)}const u=await h.json();if(u.type!=="answer"||typeof u.sdp!="string")throw new Error("Signaling server returned an invalid SDP answer");await this.setRemoteDescription(r,u)}catch(h){throw t?.aborted?y(t):s.signal.aborted?new Error(`Signaling timed out after ${this.signalingTimeoutMs} ms`):P(h,"WebRTC signaling failed")}finally{clearTimeout(i),t?.removeEventListener("abort",c)}}async setRemoteDescription(e,t){const r=new Set(e.dtlsTransports);try{await e.setRemoteDescription(t)}finally{const o=new Set(e.dtlsTransports);await Promise.all([...r].filter(n=>!o.has(n)).map(n=>n.stop()))}}async withDeadline(e,t,r){let o,n;try{return await Promise.race([e,new Promise((s,i)=>{o=setTimeout(()=>i(new Error(`WebRTC operation timed out after ${t} ms`)),t),n=f(()=>i(y(r)),"onAbort"),r?.addEventListener("abort",n,{once:!0}),r?.aborted&&n()})])}finally{clearTimeout(o),n&&r?.removeEventListener("abort",n)}}async waitForIceGatheringComplete(e,t){if(e.iceGatheringState==="complete")return!0;if(!await new Promise(o=>{let n=!1,s;const i=f(()=>{clearTimeout(s),e.removeEventListener("icegatheringstatechange",h)},"cleanup"),c=f(u=>{n||(n=!0,i(),o(u))},"finish"),h=f(()=>{e.iceGatheringState==="complete"&&c(!0)},"onStateChange");s=setTimeout(()=>c(!1),t),e.addEventListener("icegatheringstatechange",h),h()}))throw new Error(`ICE gathering timed out after ${t} ms`);return!0}async waitForChannelOpen(e,t,r){if(e.readyState!=="open"){if(r?.aborted)throw y(r);await new Promise((o,n)=>{let s=!1;const i=f(()=>{clearTimeout(l),e.removeEventListener("open",h),e.removeEventListener("error",u),e.removeEventListener("close",m),r?.removeEventListener("abort",a)},"cleanup"),c=f(g=>{s||(s=!0,i(),g?n(g):o())},"finish"),h=f(()=>c(),"onOpen"),u=f(()=>c(new Error("WebRTC DataChannel failed to open")),"onError"),m=f(()=>c(new Error("WebRTC DataChannel closed before opening")),"onClose"),a=f(()=>c(y(r)),"onAbort"),l=setTimeout(()=>c(new Error(`DataChannel open timeout after ${t} ms`)),t);e.addEventListener("open",h),e.addEventListener("error",u),e.addEventListener("close",m),r?.addEventListener("abort",a,{once:!0}),e.readyState==="open"?c():e.readyState==="closed"&&c(new Error("WebRTC DataChannel is closed"))})}}sendRequest(e,t,r){if(t.signal?.aborted)return Promise.reject(y(t.signal));if(e.dataChannel.readyState!=="open")return Promise.reject(new Error("WebRTC DataChannel is not open"));const o=this.createId(`request${++this.requestSequence}`),n=r&&r>0?r:this.requestTimeoutMs;return new Promise((s,i)=>{const c={resolve:s,reject:i,timer:setTimeout(()=>{this.finishPending(e,o),i(new Error(`WebRTC request timed out after ${n} ms`))},n),...t.signal?{signal:t.signal}:{}};t.signal&&(c.abortListener=()=>{this.finishPending(e,o),i(y(t.signal))},t.signal.addEventListener("abort",c.abortListener,{once:!0})),e.pending.set(o,c);try{e.dataChannel.send(JSON.stringify({requestId:o,method:t.method,url:t.url,headers:t.headers,body:x(t.body),protocol:t.protocol}))}catch(h){this.finishPending(e,o),i(P(h,"Failed to send WebRTC request"))}})}finishPending(e,t){const r=e.pending.get(t);r&&(clearTimeout(r.timer),r.signal&&r.abortListener&&r.signal.removeEventListener("abort",r.abortListener),e.pending.delete(t))}rejectAllPending(e,t){for(const[r,o]of e.pending)this.finishPending(e,r),o.reject(t)}async disposeClientSession(e,t){if(!e.closed){e.closed=!0,this.sessions.get(e.peerUrl)===e&&this.sessions.delete(e.peerUrl),this.rejectAllPending(e,new Error("WebRTC session closed"));try{e.dataChannel.close()}catch{}try{await e.peerConnection.close()}catch{}if(t)try{const r=new URL(e.signalingUrl);r.searchParams.set("sessionId",e.id);const o=new AbortController,n=setTimeout(()=>o.abort(),2e3);try{await fetch(r,{method:"DELETE",signal:o.signal})}finally{clearTimeout(n)}}catch{}}}attachServerPeerEvents(e,t){const r=e.peerConnection;r.addEventListener("datachannel",o=>{const n=o.channel;n.addEventListener("message",s=>{(async()=>{let i;try{const c=JSON.parse(String(s.data));if(i=typeof c.requestId=="string"?c.requestId:void 0,!i||typeof c.method!="string"||typeof c.url!="string")throw new Error("Invalid WebRTC transport request");const h={method:c.method,url:c.url,headers:c.headers&&typeof c.headers=="object"?{...c.headers,"x-webrtc-session-id":e.id}:{"x-webrtc-session-id":e.id},body:D(c.body),protocol:typeof c.protocol=="string"?c.protocol:"webrtc"};let u;if(t?u=await t(h):u={status:501,statusText:"Not Implemented",headers:{"content-type":"application/json"},body:{error:"No server request handler configured"},url:h.url},n.readyState!=="open")return;n.send(JSON.stringify({requestId:i,status:u.status,statusText:u.statusText??"",headers:u.headers,body:x(u.body),url:u.url??h.url}))}catch(c){if(this.log("error","[WebRtcTransport Server] Request failed:",c),i&&n.readyState==="open")try{n.send(JSON.stringify({requestId:i,status:500,statusText:"Internal Server Error",headers:{"content-type":"application/json"},body:{error:P(c,"Request handler failed").message}}))}catch{}}})()})}),r.addEventListener("track",o=>{const n=o.track;e.remoteStream.getTracks().some(s=>s.id===n.id)||e.remoteStream.addTrack(n),this.emitTrack({side:"server",sessionId:e.id,peerConnection:r,track:n,stream:e.remoteStream,streams:o.streams})}),r.addEventListener("connectionstatechange",()=>{(r.connectionState==="failed"||r.connectionState==="closed")&&this.closeServerSession(e.id)})}async attachConfiguredServerStream(e){if(e.closed)return;if(!e.localStreamResolved){const o=this.options.localStream,n=typeof o=="function",s=n?await o(e.id,e.context):o;if(e.closed){s&&n&&(e.localStream=s,e.ownsLocalStream=!0,this.disposeServerLocalStream(e));return}e.localStreamResolved=!0,s&&(e.localStream=s,e.ownsLocalStream=n)}const t=e.localStream;if(!t)return;const r=e.peerConnection;for(const o of t.getTracks())e.closed||o.readyState!=="live"||e.localTrackIds.has(o.id)||(r.addTrack(o,t),e.localTrackIds.add(o.id))}disposeServerLocalStream(e){const t=e.localStream,r=e.ownsLocalStream;if(delete e.localStream,e.ownsLocalStream=!1,!(!t||!r))try{this.options.disposeLocalStream?.(e.id,t)}catch{}finally{for(const o of t.getTracks())o.readyState!=="ended"&&o.stop()}}closeServerSession(e){const t=this.closingServerSessions.get(e);if(t)return t;const r=this.serverSessions.get(e);if(!r||r.closed)return Promise.resolve();r.closed=!0,r.controller.abort(),r.release(),this.serverSessions.delete(e);const o=Promise.resolve().then(async()=>{try{await r.peerConnection.close()}catch(n){this.log("warn","WebRTC peer close failed",n)}finally{this.disposeServerLocalStream(r),this.emitServerSession(r,"closed")}}).finally(()=>this.closingServerSessions.delete(e));return this.closingServerSessions.set(e,o),o}emitTrack(e){for(const t of this.trackListeners)try{t(e)}catch(r){this.log("error","[WebRtcTransport] onTrack listener failed:",r)}}toSignalingUrl(e){if(e.startsWith("webrtcs://"))return e.replace(/^webrtcs:\/\//,"https://");if(e.startsWith("webrtc://"))return e.replace(/^webrtc:\/\//,"http://");throw new Error(`Unsupported WebRTC signaling URL: ${e}`)}createId(e){return`${e}-${crypto.randomUUID()}`}}export{Y as WebRtcTransport};
 //# sourceMappingURL=webrtc.js.map

package/dist/webrtc.js.map CHANGED Viewed

@@ -1,7 +1,7 @@
 {
   "version": 3,
   "sources": ["../src/webrtc.ts"],
-  "sourcesContent": ["import type {\n  HyperTransport,\n  LogLevel,\n  TransportListenOptions,\n  TransportRequest,\n  TransportResponse,\n  TransportServer,\n} from \"@hyperttp/types\";\nimport {\n  MAX_SIGNALING_BODY_BYTES,\n  type ClientSession,\n  type PendingResponse,\n  type ServerSession,\n  type WebRtcSessionView,\n  type WebRtcTrackEvent,\n  type WebRtcTransportOptions as InternalWebRtcTransportOptions,\n  type WebRtcTransportRequest,\n} from \"./types/types.js\";\nimport { abortError, decodeWireBody, encodeWireBody, errorFromUnknown } from \"./utils/helper.js\";\nimport { createServer, type Server as HttpServer } from \"node:http\";\nimport {\n  MediaStream as WeriftMediaStream,\n  RTCPeerConnection as WeriftPeerConnection,\n  RTCDataChannel as WeriftDataChannel,\n  type MediaStreamTrack as WeriftMediaStreamTrack,\n} from \"werift\";\n\ndeclare module \"@hyperttp/types\" {\n  interface TransportServer {\n    readonly webrtc?: Pick<WebRtcTransport, \"getSession\" | \"getServerSession\" | \"onTrack\">;\n  }\n}\n\ntype WeriftPeerConfig = NonNullable<ConstructorParameters<typeof WeriftPeerConnection>[0]>;\ntype WeriftIceServers = WeriftPeerConfig[\"iceServers\"];\n\nfunction normalizeIceServers(servers: RTCIceServer[] | undefined): WeriftIceServers {\n  const normalized = (servers ?? []).flatMap((server) => {\n    const urls = Array.isArray(server.urls) ? server.urls : [server.urls];\n    if (server.credential !== undefined && typeof server.credential !== \"string\") {\n      throw new TypeError(\"Werift currently requires ICE server credentials to be strings.\");\n    }\n    return urls.map((url) => ({\n      urls: url,\n      ...(server.username !== undefined ? { username: server.username } : {}),\n      ...(server.credential !== undefined ? { credential: server.credential } : {}),\n    }));\n  });\n  return normalized as unknown as WeriftIceServers;\n}\n\nfunction createWeriftPeerConnection(configuration: RTCConfiguration): WeriftPeerConnection {\n  const config = {\n    iceServers: normalizeIceServers(configuration.iceServers),\n    ...(configuration.iceTransportPolicy !== undefined\n      ? { iceTransportPolicy: configuration.iceTransportPolicy }\n      : {}),\n  } as unknown as WeriftPeerConfig;\n  return new WeriftPeerConnection(config);\n}\n\nfunction createWeriftMediaStream(): WeriftMediaStream {\n  return new WeriftMediaStream();\n}\n\nexport type WebRtcTransportOptions = Omit<\n  InternalWebRtcTransportOptions,\n  \"localStream\" | \"disposeLocalStream\" | \"createMediaStream\"\n> & {\n  readonly localStream?: WeriftMediaStream | ((sessionId: string) => WeriftMediaStream | undefined);\n  readonly disposeLocalStream?: (sessionId: string, stream: WeriftMediaStream) => void;\n  readonly createMediaStream?: () => WeriftMediaStream;\n  readonly logger?: (level: LogLevel, message: string, meta?: unknown) => void;\n};\n\nexport interface WebRtcServerSessionView {\n  readonly id: string;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly remoteStream: WeriftMediaStream;\n  readonly localStream?: WeriftMediaStream;\n}\n\nexport interface WebRtcTransportServer extends TransportServer {\n  readonly webrtc: Pick<WebRtcTransport, \"getSession\" | \"getServerSession\" | \"onTrack\">;\n}\n\nexport class WebRtcTransport implements HyperTransport {\n  private readonly sessions = new Map<string, ClientSession>();\n  private readonly connecting = new Map<string, Promise<ClientSession>>();\n  private readonly serverSessions = new Map<string, ServerSession>();\n  private readonly trackListeners = new Set<(event: WebRtcTrackEvent) => void>();\n  private readonly serverClosers = new Set<() => Promise<void>>();\n  private requestSequence = 0;\n  private closing = false;\n  private readonly pendingPeers = new Set<WeriftPeerConnection>();\n  private readonly iceServers: RTCIceServer[];\n  private readonly iceGatheringTimeoutMs: number;\n  private readonly signalingTimeoutMs: number;\n  private readonly connectionTimeoutMs: number;\n  private readonly requestTimeoutMs: number;\n  readonly createMediaStream: () => WeriftMediaStream;\n  private readonly createPeerConnection: (configuration: RTCConfiguration) => WeriftPeerConnection;\n  private readonly options: InternalWebRtcTransportOptions;\n\n  constructor(options: WebRtcTransportOptions = {}) {\n    this.options = options as unknown as InternalWebRtcTransportOptions;\n    this.createPeerConnection = options.createPeerConnection ?? createWeriftPeerConnection;\n    this.createMediaStream =\n      (options.createMediaStream as unknown as (() => WeriftMediaStream) | undefined) ??\n      createWeriftMediaStream;\n    this.iceServers = options.iceServers ?? [{ urls: \"stun:stun.l.google.com:19302\" }];\n    this.iceGatheringTimeoutMs = options.iceGatheringTimeoutMs ?? 10_000;\n    this.signalingTimeoutMs = options.signalingTimeoutMs ?? 10_000;\n    this.connectionTimeoutMs = options.connectionTimeoutMs ?? 15_000;\n    this.requestTimeoutMs = options.requestTimeoutMs ?? 15_000;\n  }\n\n  private log(level: \"debug\" | \"info\" | \"warn\" | \"error\", message: string, meta?: unknown): void {\n    if (this.options.logger) {\n      this.options.logger(level, message, meta);\n      return;\n    }\n    if (!this.options.verbose && level !== \"error\") return;\n    const output = level === \"debug\" ? console.debug : console[level];\n    if (meta === undefined) output(message);\n    else output(message, meta);\n  }\n\n  onTrack(listener: (event: WebRtcTrackEvent) => void): () => void {\n    this.trackListeners.add(listener);\n    return () => this.trackListeners.delete(listener);\n  }\n\n  getServerSession(sessionId: string): WebRtcServerSessionView | undefined {\n    const session = this.serverSessions.get(sessionId);\n    if (!session || session.closed) return undefined;\n    return {\n      id: session.id,\n      peerConnection: session.peerConnection,\n      remoteStream: session.remoteStream,\n      ...(session.localStream ? { localStream: session.localStream } : {}),\n    };\n  }\n\n  getSession(peerUrl: string): WebRtcSessionView | undefined {\n    const session = this.sessions.get(peerUrl);\n    if (!session || session.closed) return undefined;\n\n    return {\n      id: session.id,\n      peerUrl: session.peerUrl,\n      peerConnection: session.peerConnection,\n      dataChannel: session.dataChannel,\n      remoteStream: session.remoteStream,\n      media: session.mediaController,\n      ...(session.localStream ? { localStream: session.localStream } : {}),\n    };\n  }\n\n  async execute(request: TransportRequest): Promise<TransportResponse> {\n    if (this.closing) throw new Error(\"WebRTC transport is closed\");\n    const rtcRequest = request as WebRtcTransportRequest;\n    const peerUrl = rtcRequest.peerUrl ?? request.url;\n\n    if (request.signal?.aborted) throw abortError(request.signal);\n\n    const session = await this.getOrCreateSession(peerUrl, rtcRequest);\n    await this.ensureMedia(session, rtcRequest);\n\n    if (request.signal?.aborted) throw abortError(request.signal);\n\n    return this.sendRequest(session, request, rtcRequest.timeout);\n  }\n\n  public async listen(options: TransportListenOptions): Promise<WebRtcTransportServer> {\n    if (this.closing) throw new Error(\"WebRTC transport is closed\");\n    const ownedSessions = new Set<string>();\n    const port = options.port ?? 3000;\n    const host = options.host ?? \"127.0.0.1\";\n    const signalingPath = options.signaling?.path ?? \"/signal\";\n    if (\n      !signalingPath.startsWith(\"/\") ||\n      signalingPath.startsWith(\"//\") ||\n      /[?#\\\\\\s]/u.test(signalingPath) ||\n      Array.from(signalingPath).some((char) => {\n        const code = char.codePointAt(0)!;\n        return code <= 0x1f || (code >= 0x7f && code <= 0x9f);\n      }) ||\n      new URL(signalingPath, \"http://localhost\").pathname !== signalingPath\n    ) {\n      throw new TypeError(\n        \"WebRTC signaling path must be an absolute literal pathname without query or fragment\",\n      );\n    }\n\n    if (options.signal?.aborted) {\n      throw new Error(\"WebRtcTransport: server startup aborted\");\n    }\n\n    let closed = false;\n    let closePromise: Promise<void> | undefined;\n    const pendingAuthorizations = new Set<AbortController>();\n\n    const httpServer: HttpServer = createServer((req, res) => {\n      void (async () => {\n        const sendJson = (status: number, body: unknown) => {\n          if (res.headersSent || res.destroyed) return;\n          res.writeHead(status, { \"content-type\": \"application/json\" });\n          res.end(JSON.stringify(body));\n        };\n\n        if (closed || this.closing) {\n          sendJson(503, { error: \"WebRTC server is closing\" });\n          return;\n        }\n\n        let parsedUrl: URL;\n        try {\n          parsedUrl = new URL(req.url ?? \"/\", \"http://localhost\");\n        } catch {\n          sendJson(400, { error: \"Invalid request URL\" });\n          return;\n        }\n\n        if (parsedUrl.pathname !== signalingPath) {\n          sendJson(404, { error: \"Signaling endpoint not found\" });\n          return;\n        }\n\n        if ((req.method === \"POST\" || req.method === \"DELETE\") && options.signaling?.authorize) {\n          const controller = new AbortController();\n          const onDisconnect = () => controller.abort();\n          const onAuthorizationAbort = () => {\n            sendJson(503, { error: \"WebRTC signaling request aborted\" });\n          };\n          req.once(\"aborted\", onDisconnect);\n          res.once(\"close\", onDisconnect);\n          controller.signal.addEventListener(\"abort\", onAuthorizationAbort, { once: true });\n          pendingAuthorizations.add(controller);\n          try {\n            const headers: Record<string, string | string[]> = {};\n            for (const [name, value] of Object.entries(req.headers)) {\n              if (value !== undefined) headers[name] = value;\n            }\n            const allowed = await options.signaling.authorize({\n              protocol: \"webrtc\",\n              method: req.method,\n              url: req.url ?? \"/\",\n              headers,\n              signal: controller.signal,\n            });\n            if (controller.signal.aborted || req.aborted || res.destroyed) return;\n            if (closed || this.closing || options.signal?.aborted) {\n              sendJson(503, { error: \"WebRTC server is closing\" });\n              return;\n            }\n            if (!allowed) {\n              sendJson(403, { error: \"Signaling forbidden\" });\n              return;\n            }\n          } catch {\n            sendJson(403, { error: \"Signaling forbidden\" });\n            return;\n          } finally {\n            pendingAuthorizations.delete(controller);\n            req.off(\"aborted\", onDisconnect);\n            res.off(\"close\", onDisconnect);\n            controller.signal.removeEventListener(\"abort\", onAuthorizationAbort);\n          }\n        }\n\n        if (req.method === \"DELETE\") {\n          const sessionId = parsedUrl.searchParams.get(\"sessionId\");\n          if (sessionId && ownedSessions.has(sessionId)) {\n            await this.closeServerSession(sessionId);\n            ownedSessions.delete(sessionId);\n          }\n          res.writeHead(204).end();\n          return;\n        }\n\n        if (req.method !== \"POST\") {\n          sendJson(405, { error: \"Method not allowed\" });\n          return;\n        }\n\n        let raw = \"\";\n        try {\n          for await (const chunk of req) {\n            raw += Buffer.isBuffer(chunk) ? chunk.toString(\"utf8\") : String(chunk);\n            if (Buffer.byteLength(raw, \"utf8\") > MAX_SIGNALING_BODY_BYTES) {\n              sendJson(413, { error: \"Signaling payload too large\" });\n              return;\n            }\n          }\n        } catch (error) {\n          sendJson(400, { error: errorFromUnknown(error, \"Failed to read body\").message });\n          return;\n        }\n\n        let pc: WeriftPeerConnection | undefined;\n        let session: ServerSession | undefined;\n\n        try {\n          const payload = JSON.parse(raw) as {\n            sessionId?: unknown;\n            sdp?: RTCSessionDescriptionInit;\n          };\n\n          if (\n            typeof payload.sessionId !== \"string\" ||\n            payload.sessionId.length < 1 ||\n            payload.sessionId.length > 200 ||\n            payload.sdp?.type !== \"offer\" ||\n            typeof payload.sdp.sdp !== \"string\"\n          ) {\n            sendJson(400, { error: \"Expected sessionId and a valid SDP offer\" });\n            return;\n          }\n\n          if (closed || this.closing || options.signal?.aborted || req.aborted || res.destroyed) {\n            throw new Error(\"WebRTC signaling request aborted\");\n          }\n          const sessionId = payload.sessionId;\n          session = this.serverSessions.get(sessionId);\n          if (session && !ownedSessions.has(sessionId)) {\n            sendJson(409, { error: \"Session belongs to another listener\" });\n            return;\n          }\n          if (session?.negotiating) {\n            sendJson(409, { error: \"Session negotiation already in progress\" });\n            return;\n          }\n\n          if (!session || session.closed) {\n            pc = this.createPeerConnection({ iceServers: this.iceServers });\n            session = {\n              id: sessionId,\n              peerConnection: pc,\n              remoteStream: this.createMediaStream(),\n              localTrackIds: new Set<string>(),\n              localStreamResolved: false,\n              ownsLocalStream: false,\n              closed: false,\n            };\n            ownedSessions.add(sessionId);\n            this.serverSessions.set(sessionId, session);\n            this.attachServerPeerEvents(session, options.onRequest);\n          } else {\n            pc = session.peerConnection;\n          }\n\n          session.negotiating = true;\n          await this.setRemoteDescription(pc, payload.sdp);\n          await this.attachConfiguredServerStream(session);\n\n          const answer = await pc.createAnswer();\n          await this.withDeadline(pc.setLocalDescription(answer), this.iceGatheringTimeoutMs);\n          await this.waitForIceGatheringComplete(pc, this.iceGatheringTimeoutMs);\n\n          const localDescription = pc.localDescription;\n          if (!localDescription?.sdp || localDescription.type !== \"answer\") {\n            throw new Error(\"Local SDP answer was not generated\");\n          }\n\n          if (closed || session.closed) throw new Error(\"WebRTC session closed\");\n          session.negotiating = false;\n          sendJson(200, localDescription);\n        } catch (error) {\n          this.log(\"error\", \"[WebRtcTransport Server] \u041E\u0448\u0438\u0431\u043A\u0430 \u0441\u0438\u0433\u043D\u0430\u043B\u0438\u0437\u0430\u0446\u0438\u0438:\", error);\n          if (session) {\n            await this.closeServerSession(session.id);\n          } else if (pc) {\n            try {\n              await pc.close();\n            } catch {}\n          }\n          sendJson(400, { error: errorFromUnknown(error, \"Signaling failed\").message });\n        }\n      })();\n    });\n\n    await new Promise<void>((resolve, reject) => {\n      const onError = (error: Error) => {\n        httpServer.off(\"listening\", onListening);\n        reject(error);\n      };\n      const onListening = () => {\n        httpServer.off(\"error\", onError);\n        resolve();\n      };\n      httpServer.once(\"error\", onError);\n      httpServer.once(\"listening\", onListening);\n      httpServer.listen(port, host);\n    });\n\n    this.log(\"info\", `[WebRtcTransport] Listening on http://${host}:${port}${signalingPath}`);\n\n    const onAbort = () => {\n      void closeServer().catch((error) => {\n        this.log(\"error\", \"[WebRtcTransport] Server shutdown failed:\", error);\n      });\n    };\n\n    const closeServer = (): Promise<void> => {\n      if (closePromise) return closePromise;\n      closed = true;\n      for (const controller of pendingAuthorizations) controller.abort();\n      options.signal?.removeEventListener(\"abort\", onAbort);\n      closePromise = (async () => {\n        const stopped = httpServer.listening\n          ? new Promise<void>((resolve, reject) => {\n              httpServer.close((error) => (error ? reject(error) : resolve()));\n            })\n          : Promise.resolve();\n        await Promise.all([...ownedSessions].map((id) => this.closeServerSession(id)));\n        ownedSessions.clear();\n        await stopped;\n        this.serverClosers.delete(closeServer);\n      })();\n      return closePromise;\n    };\n\n    options.signal?.addEventListener(\"abort\", onAbort, { once: true });\n    this.serverClosers.add(closeServer);\n    if (options.signal?.aborted && !closed) await closeServer();\n\n    return {\n      close: closeServer,\n      webrtc: {\n        getSession: this.getSession.bind(this),\n        getServerSession: this.getServerSession.bind(this),\n        onTrack: this.onTrack.bind(this),\n      },\n    };\n  }\n\n  async close(): Promise<void> {\n    this.closing = true;\n    await Promise.all([...this.pendingPeers].map((pc) => pc.close().catch(() => {})));\n    await Promise.allSettled(this.connecting.values());\n    await Promise.all(\n      [...this.sessions.values()].map((session) => this.disposeClientSession(session, true)),\n    );\n    await Promise.all([...this.serverClosers].map((closeServer) => closeServer()));\n  }\n\n  async destroy(): Promise<void> {\n    await this.close();\n  }\n\n  private async getOrCreateSession(\n    peerUrl: string,\n    request: WebRtcTransportRequest,\n  ): Promise<ClientSession> {\n    const existing = this.sessions.get(peerUrl);\n    if (\n      existing &&\n      !existing.closed &&\n      existing.dataChannel.readyState === \"open\" &&\n      existing.peerConnection.connectionState !== \"failed\" &&\n      existing.peerConnection.connectionState !== \"closed\"\n    ) {\n      return existing;\n    }\n\n    if (existing) {\n      await this.disposeClientSession(existing, true);\n    }\n\n    const inFlight = this.connecting.get(peerUrl);\n    if (inFlight) return inFlight;\n\n    const connectionPromise = this.createClientSession(peerUrl, request);\n    this.connecting.set(peerUrl, connectionPromise);\n\n    try {\n      const session = await connectionPromise;\n      if (this.closing || session.closed) {\n        await this.disposeClientSession(session, true);\n        throw new Error(\"WebRTC transport is closed\");\n      }\n      this.sessions.set(peerUrl, session);\n      return session;\n    } finally {\n      if (this.connecting.get(peerUrl) === connectionPromise) {\n        this.connecting.delete(peerUrl);\n      }\n    }\n  }\n\n  private async createClientSession(\n    peerUrl: string,\n    request: WebRtcTransportRequest,\n  ): Promise<ClientSession> {\n    const signalingUrl = this.toSignalingUrl(peerUrl);\n    const id = this.createId(\"session\");\n    const pc = this.createPeerConnection({\n      iceServers: request.iceServers ?? this.iceServers,\n    });\n    this.pendingPeers.add(pc);\n    const channel = pc.createDataChannel(request.channelLabel ?? \"hyperttp-webrtc-transport\", {\n      ordered: true,\n    });\n\n    const session: ClientSession = {\n      id,\n      peerUrl,\n      signalingUrl,\n      peerConnection: pc,\n      dataChannel: channel,\n      remoteStream: this.createMediaStream(),\n      ...(request.localStream ? { localStream: request.localStream } : {}),\n      localTrackIds: new Set<string>(),\n      pending: new Map<string, PendingResponse>(),\n      negotiation: Promise.resolve(),\n      closed: false,\n      mediaController: {} as any,\n    };\n\n    const trackListeners = new Set<\n      (e: { track: WeriftMediaStreamTrack; stream: WeriftMediaStream }) => void\n    >();\n    const trackEndedListeners = new Set<(e: { track: WeriftMediaStreamTrack }) => void>();\n    const streamListeners = new Set<(e: { stream: WeriftMediaStream }) => void>();\n\n    pc.addEventListener(\"track\", (event: WebRtcTrackEvent) => {\n      const { track } = event;\n      if (!session.remoteStream.getTracks().includes(track)) session.remoteStream.addTrack(track);\n      for (const listener of trackListeners) listener({ track, stream: session.remoteStream });\n      for (const listener of streamListeners) listener({ stream: session.remoteStream });\n      track.addEventListener(\"ended\", () => {\n        for (const listener of trackEndedListeners) listener({ track });\n      });\n    });\n\n    session.mediaController = {\n      localStream: session.localStream,\n      remoteStream: session.remoteStream,\n      addTrack: (track, stream) =>\n        this.queueNegotiation(session, async () => {\n          pc.addTrack(track, stream ?? session.localStream ?? this.createMediaStream());\n          if (track.id) session.localTrackIds.add(track.id);\n          await this.negotiate(session);\n        }),\n      replaceTrack: async (kind: \"audio\" | \"video\", track: any | null) => {\n        const sender = pc.getSenders().find((s: any) => s.track && s.track.kind === kind);\n        if (sender) {\n          await sender.replaceTrack(track);\n        }\n      },\n      removeTrack: (track) =>\n        this.queueNegotiation(session, async () => {\n          const sender = pc.getSenders().find((item) => item.track === track);\n          if (sender) {\n            pc.removeTrack(sender);\n            if (track.id) session.localTrackIds.delete(track.id);\n            await this.negotiate(session);\n          }\n        }),\n      close: async () => {\n        await this.disposeClientSession(session, true);\n      },\n      on(event: string, listener: any) {\n        if (event === \"track\") trackListeners.add(listener);\n        if (event === \"trackEnded\") trackEndedListeners.add(listener);\n        if (event === \"stream\") streamListeners.add(listener);\n        return () => {\n          trackListeners.delete(listener);\n          trackEndedListeners.delete(listener);\n          streamListeners.delete(listener);\n        };\n      },\n    };\n\n    try {\n      this.attachClientPeerEvents(session);\n      this.addLocalTracks(session, request.localStream);\n      this.addReceiveTransceivers(session, request.receiveAudio, request.receiveVideo);\n      await this.negotiate(session, request.signal);\n      await this.waitForChannelOpen(channel, this.connectionTimeoutMs, request.signal);\n      return session;\n    } catch (error) {\n      await this.disposeClientSession(session, true);\n      throw error;\n    } finally {\n      this.pendingPeers.delete(pc);\n    }\n  }\n\n  private attachClientPeerEvents(session: ClientSession): void {\n    const { peerConnection: pc, dataChannel: channel } = session;\n\n    pc.addEventListener(\"track\", (event: any) => {\n      const track = event.track;\n      if (!session.remoteStream.getTracks().some((item: any) => item.id === track.id)) {\n        session.remoteStream.addTrack(track);\n      }\n\n      this.emitTrack({\n        side: \"client\",\n        sessionId: session.id,\n        peerUrl: session.peerUrl,\n        peerConnection: pc,\n        track,\n        stream: session.remoteStream,\n        streams: event.streams,\n      });\n    });\n\n    pc.addEventListener(\"connectionstatechange\", () => {\n      if (pc.connectionState === \"failed\" || pc.connectionState === \"closed\") {\n        void this.disposeClientSession(session, true);\n      }\n    });\n\n    channel.addEventListener(\"message\", (event: MessageEvent) => {\n      let message: Record<string, unknown>;\n      try {\n        message = JSON.parse(String(event.data)) as Record<string, unknown>;\n      } catch {\n        return;\n      }\n\n      if (!message || typeof message !== \"object\" || typeof message.requestId !== \"string\") return;\n      const pending = session.pending.get(message.requestId);\n      if (!pending) return;\n\n      try {\n        const body = decodeWireBody(message.body);\n        this.finishPending(session, message.requestId);\n        pending.resolve({\n          status: typeof message.status === \"number\" ? message.status : 200,\n          statusText: typeof message.statusText === \"string\" ? message.statusText : \"OK\",\n          headers:\n            message.headers && typeof message.headers === \"object\"\n              ? (message.headers as TransportResponse[\"headers\"])\n              : {},\n          body,\n          url: typeof message.url === \"string\" ? message.url : session.peerUrl,\n        });\n      } catch (error) {\n        this.finishPending(session, message.requestId);\n        pending.reject(errorFromUnknown(error, \"Invalid WebRTC response\"));\n      }\n    });\n\n    channel.addEventListener(\"close\", () => {\n      this.rejectAllPending(session, new Error(\"WebRTC DataChannel closed\"));\n      void this.disposeClientSession(session, true);\n    });\n\n    channel.addEventListener(\"error\", () => {\n      this.rejectAllPending(session, new Error(\"WebRTC DataChannel error\"));\n    });\n  }\n\n  private queueNegotiation(session: ClientSession, operation: () => Promise<void>): Promise<void> {\n    const pending = session.negotiation\n      .catch(() => undefined)\n      .then(async () => {\n        if (session.closed) throw new Error(\"WebRTC session closed\");\n        try {\n          await operation();\n        } catch (error) {\n          await this.disposeClientSession(session, true);\n          throw error;\n        }\n      });\n    session.negotiation = pending;\n    return pending;\n  }\n\n  private async ensureMedia(\n    session: ClientSession,\n    request: WebRtcTransportRequest,\n  ): Promise<void> {\n    const needsPotentialChanges =\n      Boolean(request.localStream) ||\n      Boolean(request.receiveAudio) ||\n      Boolean(request.receiveVideo);\n    if (!needsPotentialChanges || session.closed) return;\n\n    await this.queueNegotiation(session, async () => {\n      let changed = this.addLocalTracks(session, request.localStream);\n      changed =\n        this.addReceiveTransceivers(session, request.receiveAudio, request.receiveVideo) || changed;\n\n      if (request.localStream) {\n        session.localStream = request.localStream;\n        (session.mediaController as { localStream?: WeriftMediaStream }).localStream =\n          request.localStream;\n      }\n      if (changed) await this.negotiate(session, request.signal);\n    });\n  }\n\n  private addLocalTracks(session: ClientSession, stream?: WeriftMediaStream): boolean {\n    if (!stream) return false;\n    let changed = false;\n\n    for (const track of stream.getTracks()) {\n      if (track.readyState !== \"live\" || session.localTrackIds.has(track.id!)) continue;\n      session.peerConnection.addTrack(track, stream);\n      session.localTrackIds.add(track.id!);\n      changed = true;\n    }\n\n    return changed;\n  }\n\n  private addReceiveTransceivers(\n    session: ClientSession,\n    receiveAudio?: boolean,\n    receiveVideo?: boolean,\n  ): boolean {\n    const pc = session.peerConnection;\n    let changed = false;\n\n    const ensureReceive = (kind: \"audio\" | \"video\", enabled: boolean | undefined) => {\n      if (!enabled) return;\n      const transceivers = pc\n        .getTransceivers()\n        .find(\n          (item: any) =>\n            item.receiver.track.kind === kind &&\n            (item.direction === \"recvonly\" || item.direction === \"sendrecv\"),\n        );\n      if (!transceivers) {\n        pc.addTransceiver(kind, { direction: \"recvonly\" });\n        changed = true;\n      }\n    };\n\n    ensureReceive(\"audio\", receiveAudio);\n    ensureReceive(\"video\", receiveVideo);\n\n    return changed;\n  }\n\n  private async negotiate(session: ClientSession, signal?: AbortSignal): Promise<void> {\n    if (signal?.aborted) throw abortError(signal);\n\n    const pc = session.peerConnection;\n    if (pc.signalingState !== \"stable\") {\n      throw new Error(`Cannot negotiate WebRTC session in state ${pc.signalingState}`);\n    }\n\n    const offer = await pc.createOffer();\n    await this.withDeadline(pc.setLocalDescription(offer), this.iceGatheringTimeoutMs, signal);\n    await this.waitForIceGatheringComplete(pc, this.iceGatheringTimeoutMs);\n\n    const localDescription = pc.localDescription;\n    if (!localDescription?.sdp || localDescription.type !== \"offer\") {\n      throw new Error(\"Local SDP offer was not generated\");\n    }\n\n    const controller = new AbortController();\n    const timer = setTimeout(() => controller.abort(), this.signalingTimeoutMs);\n    const abortListener = () => controller.abort(signal?.reason);\n    signal?.addEventListener(\"abort\", abortListener, { once: true });\n    if (signal?.aborted) abortListener();\n\n    try {\n      const response = await fetch(session.signalingUrl, {\n        method: \"POST\",\n        headers: { \"content-type\": \"application/json\" },\n        body: JSON.stringify({ sessionId: session.id, sdp: localDescription }),\n        signal: controller.signal,\n      });\n\n      if (!response.ok) {\n        const text = await response.text();\n        throw new Error(\n          `Signaling HTTP error: ${response.status} ${response.statusText}${text ? ` \u2014 ${text}` : \"\"}`,\n        );\n      }\n\n      const answer = (await response.json()) as RTCSessionDescriptionInit;\n      if (answer.type !== \"answer\" || typeof answer.sdp !== \"string\") {\n        throw new Error(\"Signaling server returned an invalid SDP answer\");\n      }\n\n      await this.setRemoteDescription(pc, answer);\n    } catch (error) {\n      if (signal?.aborted) throw abortError(signal);\n      if (controller.signal.aborted) {\n        throw new Error(`Signaling timed out after ${this.signalingTimeoutMs} ms`);\n      }\n      throw errorFromUnknown(error, \"WebRTC signaling failed\");\n    } finally {\n      clearTimeout(timer);\n      signal?.removeEventListener(\"abort\", abortListener);\n    }\n  }\n\n  private async setRemoteDescription(\n    pc: WeriftPeerConnection,\n    description: RTCSessionDescriptionInit,\n  ): Promise<void> {\n    const previous = new Set(pc.dtlsTransports);\n    try {\n      await pc.setRemoteDescription(description);\n    } finally {\n      // Werift can replace transports while applying BUNDLE without stopping the\n      // displaced ICE sockets. They are no longer reachable by pc.close().\n      const active = new Set(pc.dtlsTransports);\n      await Promise.all(\n        [...previous]\n          .filter((transport) => !active.has(transport))\n          .map((transport) => transport.stop()),\n      );\n    }\n  }\n\n  private async withDeadline<T>(\n    operation: Promise<T>,\n    timeoutMs: number,\n    signal?: AbortSignal,\n  ): Promise<T> {\n    let timer: ReturnType<typeof setTimeout> | undefined;\n    let onAbort: (() => void) | undefined;\n    try {\n      return await Promise.race([\n        operation,\n        new Promise<never>((_resolve, reject) => {\n          timer = setTimeout(\n            () => reject(new Error(`WebRTC operation timed out after ${timeoutMs} ms`)),\n            timeoutMs,\n          );\n          onAbort = () => reject(abortError(signal));\n          signal?.addEventListener(\"abort\", onAbort, { once: true });\n          if (signal?.aborted) onAbort();\n        }),\n      ]);\n    } finally {\n      clearTimeout(timer);\n      if (onAbort) signal?.removeEventListener(\"abort\", onAbort);\n    }\n  }\n\n  private async waitForIceGatheringComplete(\n    pc: WeriftPeerConnection,\n    timeoutMs: number,\n  ): Promise<boolean> {\n    if (pc.iceGatheringState === \"complete\") return true;\n\n    const completed = await new Promise<boolean>((resolve) => {\n      let settled = false;\n      let timer: ReturnType<typeof setTimeout>;\n\n      const cleanup = () => {\n        clearTimeout(timer);\n        pc.removeEventListener(\"icegatheringstatechange\", onStateChange);\n      };\n\n      const finish = (value: boolean) => {\n        if (settled) return;\n        settled = true;\n        cleanup();\n        resolve(value);\n      };\n\n      const onStateChange = () => {\n        if (pc.iceGatheringState === \"complete\") finish(true);\n      };\n\n      timer = setTimeout(() => finish(false), timeoutMs);\n      pc.addEventListener(\"icegatheringstatechange\", onStateChange);\n      onStateChange();\n    });\n\n    if (!completed) throw new Error(`ICE gathering timed out after ${timeoutMs} ms`);\n    return true;\n  }\n\n  private async waitForChannelOpen(\n    channel: WeriftDataChannel,\n    timeoutMs: number,\n    signal?: AbortSignal,\n  ): Promise<void> {\n    if (channel.readyState === \"open\") return;\n    if (signal?.aborted) throw abortError(signal);\n\n    await new Promise<void>((resolve, reject) => {\n      let settled = false;\n      const cleanup = () => {\n        clearTimeout(timer);\n        channel.removeEventListener(\"open\", onOpen);\n        channel.removeEventListener(\"error\", onError);\n        channel.removeEventListener(\"close\", onClose);\n        signal?.removeEventListener(\"abort\", onAbort);\n      };\n      const finish = (error?: Error) => {\n        if (settled) return;\n        settled = true;\n        cleanup();\n        if (error) reject(error);\n        else resolve();\n      };\n      const onOpen = () => finish();\n      const onError = () => finish(new Error(\"WebRTC DataChannel failed to open\"));\n      const onClose = () => finish(new Error(\"WebRTC DataChannel closed before opening\"));\n      const onAbort = () => finish(abortError(signal));\n      const timer = setTimeout(\n        () => finish(new Error(`DataChannel open timeout after ${timeoutMs} ms`)),\n        timeoutMs,\n      );\n\n      channel.addEventListener(\"open\", onOpen);\n      channel.addEventListener(\"error\", onError);\n      channel.addEventListener(\"close\", onClose);\n      signal?.addEventListener(\"abort\", onAbort, { once: true });\n\n      if (channel.readyState === \"open\") finish();\n      else if (channel.readyState === \"closed\") finish(new Error(\"WebRTC DataChannel is closed\"));\n    });\n  }\n\n  private sendRequest(\n    session: ClientSession,\n    request: TransportRequest,\n    requestedTimeout?: number,\n  ): Promise<TransportResponse> {\n    if (request.signal?.aborted) return Promise.reject(abortError(request.signal));\n    if (session.dataChannel.readyState !== \"open\") {\n      return Promise.reject(new Error(\"WebRTC DataChannel is not open\"));\n    }\n\n    const requestId = this.createId(`request${++this.requestSequence}`);\n    const timeoutMs =\n      requestedTimeout && requestedTimeout > 0 ? requestedTimeout : this.requestTimeoutMs;\n\n    return new Promise<TransportResponse>((resolve, reject) => {\n      const pending: PendingResponse = {\n        resolve,\n        reject,\n        timer: setTimeout(() => {\n          this.finishPending(session, requestId);\n          reject(new Error(`WebRTC request timed out after ${timeoutMs} ms`));\n        }, timeoutMs),\n        ...(request.signal ? { signal: request.signal } : {}),\n      };\n\n      if (request.signal) {\n        pending.abortListener = () => {\n          this.finishPending(session, requestId);\n          reject(abortError(request.signal));\n        };\n        request.signal.addEventListener(\"abort\", pending.abortListener, { once: true });\n      }\n\n      session.pending.set(requestId, pending);\n\n      try {\n        session.dataChannel.send(\n          JSON.stringify({\n            requestId,\n            method: request.method,\n            url: request.url,\n            headers: request.headers,\n            body: encodeWireBody(request.body),\n            protocol: request.protocol,\n          }),\n        );\n      } catch (error) {\n        this.finishPending(session, requestId);\n        reject(errorFromUnknown(error, \"Failed to send WebRTC request\"));\n      }\n    });\n  }\n\n  private finishPending(session: ClientSession, requestId: string): void {\n    const pending = session.pending.get(requestId);\n    if (!pending) return;\n    clearTimeout(pending.timer);\n    if (pending.signal && pending.abortListener) {\n      pending.signal.removeEventListener(\"abort\", pending.abortListener);\n    }\n    session.pending.delete(requestId);\n  }\n\n  private rejectAllPending(session: ClientSession, error: Error): void {\n    for (const [requestId, pending] of session.pending) {\n      this.finishPending(session, requestId);\n      pending.reject(error);\n    }\n  }\n\n  private async disposeClientSession(session: ClientSession, notifyRemote: boolean): Promise<void> {\n    if (session.closed) return;\n    session.closed = true;\n\n    if (this.sessions.get(session.peerUrl) === session) {\n      this.sessions.delete(session.peerUrl);\n    }\n    this.rejectAllPending(session, new Error(\"WebRTC session closed\"));\n\n    try {\n      session.dataChannel.close();\n    } catch {}\n    try {\n      await session.peerConnection.close();\n    } catch {}\n\n    if (notifyRemote) {\n      try {\n        const url = new URL(session.signalingUrl);\n        url.searchParams.set(\"sessionId\", session.id);\n        const controller = new AbortController();\n        const timer = setTimeout(() => controller.abort(), 2_000);\n        try {\n          await fetch(url, { method: \"DELETE\", signal: controller.signal });\n        } finally {\n          clearTimeout(timer);\n        }\n      } catch {}\n    }\n  }\n\n  private attachServerPeerEvents(\n    session: ServerSession,\n    onRequest: TransportListenOptions[\"onRequest\"],\n  ): void {\n    const pc = session.peerConnection;\n\n    pc.addEventListener(\"datachannel\", (event: any) => {\n      const channel = event.channel;\n      channel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n        void (async () => {\n          let requestId: string | undefined;\n          try {\n            const message = JSON.parse(String(messageEvent.data)) as {\n              requestId?: unknown;\n              method?: unknown;\n              url?: unknown;\n              headers?: unknown;\n              body?: unknown;\n              protocol?: unknown;\n            };\n\n            requestId = typeof message.requestId === \"string\" ? message.requestId : undefined;\n            if (\n              !requestId ||\n              typeof message.method !== \"string\" ||\n              typeof message.url !== \"string\"\n            ) {\n              throw new Error(\"Invalid WebRTC transport request\");\n            }\n\n            const request: TransportRequest = {\n              method: message.method,\n              url: message.url,\n              headers:\n                message.headers && typeof message.headers === \"object\"\n                  ? {\n                      ...(message.headers as TransportRequest[\"headers\"]),\n                      \"x-webrtc-session-id\": session.id,\n                    }\n                  : { \"x-webrtc-session-id\": session.id },\n              body: decodeWireBody(message.body),\n              protocol:\n                typeof message.protocol === \"string\"\n                  ? (message.protocol as TransportRequest[\"protocol\"])\n                  : \"webrtc\",\n            };\n\n            let response: TransportResponse;\n            if (!onRequest) {\n              response = {\n                status: 501,\n                statusText: \"Not Implemented\",\n                headers: { \"content-type\": \"application/json\" },\n                body: { error: \"No server request handler configured\" },\n                url: request.url,\n              };\n            } else {\n              response = await onRequest(request);\n            }\n\n            if (channel.readyState !== \"open\") return;\n            channel.send(\n              JSON.stringify({\n                requestId,\n                status: response.status,\n                statusText: response.statusText ?? \"\",\n                headers: response.headers,\n                body: encodeWireBody(response.body),\n                url: response.url ?? request.url,\n              }),\n            );\n          } catch (error) {\n            this.log(\"error\", \"[WebRtcTransport Server] Request failed:\", error);\n            if (requestId && channel.readyState === \"open\") {\n              try {\n                channel.send(\n                  JSON.stringify({\n                    requestId,\n                    status: 500,\n                    statusText: \"Internal Server Error\",\n                    headers: { \"content-type\": \"application/json\" },\n                    body: { error: errorFromUnknown(error, \"Request handler failed\").message },\n                  }),\n                );\n              } catch {}\n            }\n          }\n        })();\n      });\n    });\n\n    pc.addEventListener(\"track\", (event: any) => {\n      const track = event.track;\n      if (!session.remoteStream.getTracks().some((item: any) => item.id === track.id)) {\n        session.remoteStream.addTrack(track);\n      }\n      this.emitTrack({\n        side: \"server\",\n        sessionId: session.id,\n        peerConnection: pc,\n        track,\n        stream: session.remoteStream,\n        streams: event.streams,\n      });\n    });\n\n    pc.addEventListener(\"connectionstatechange\", () => {\n      if (pc.connectionState === \"failed\" || pc.connectionState === \"closed\") {\n        void this.closeServerSession(session.id);\n      }\n    });\n  }\n\n  private async attachConfiguredServerStream(session: ServerSession): Promise<void> {\n    if (session.closed) return;\n\n    if (!session.localStreamResolved) {\n      const configured = this.options.localStream;\n      const isFactory = typeof configured === \"function\";\n      const stream = isFactory ? configured(session.id) : configured;\n      session.localStreamResolved = true;\n\n      if (stream) {\n        session.localStream = stream;\n        session.ownsLocalStream = isFactory;\n      }\n    }\n\n    const stream = session.localStream;\n    if (!stream) return;\n\n    const pc = session.peerConnection;\n    for (const track of stream.getTracks()) {\n      if (session.closed || track.readyState !== \"live\" || session.localTrackIds.has(track.id!)) {\n        continue;\n      }\n      pc.addTrack(track, stream);\n      session.localTrackIds.add(track.id!);\n    }\n  }\n\n  private disposeServerLocalStream(session: ServerSession): void {\n    const stream = session.localStream;\n    const ownsStream = session.ownsLocalStream;\n\n    delete session.localStream;\n    session.ownsLocalStream = false;\n\n    if (!stream || !ownsStream) return;\n\n    try {\n      this.options.disposeLocalStream?.(session.id, stream);\n    } catch {\n    } finally {\n      for (const track of stream.getTracks()) {\n        if (track.readyState !== \"ended\") track.stop();\n      }\n    }\n  }\n\n  private async closeServerSession(sessionId: string): Promise<void> {\n    const session = this.serverSessions.get(sessionId);\n    if (!session || session.closed) return;\n\n    session.closed = true;\n    this.serverSessions.delete(sessionId);\n\n    try {\n      await session.peerConnection.close();\n    } catch {\n    } finally {\n      this.disposeServerLocalStream(session);\n    }\n  }\n\n  private emitTrack(event: WebRtcTrackEvent): void {\n    for (const listener of this.trackListeners) {\n      try {\n        listener(event);\n      } catch (error) {\n        this.log(\"error\", \"[WebRtcTransport] onTrack listener failed:\", error);\n      }\n    }\n  }\n\n  private toSignalingUrl(peerUrl: string): string {\n    if (peerUrl.startsWith(\"webrtcs://\")) return peerUrl.replace(/^webrtcs:\\/\\//, \"https://\");\n    if (peerUrl.startsWith(\"webrtc://\")) return peerUrl.replace(/^webrtc:\\/\\//, \"http://\");\n    throw new Error(`Unsupported WebRTC signaling URL: ${peerUrl}`);\n  }\n\n  private createId(prefix: string): string {\n    return `${prefix}-${crypto.randomUUID()}`;\n  }\n}\n"],
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-  "names": ["MAX_SIGNALING_BODY_BYTES", "abortError", "decodeWireBody", "encodeWireBody", "errorFromUnknown", "createServer", "WeriftMediaStream", "WeriftPeerConnection", "normalizeIceServers", "servers", "server", "urls", "url", "__name", "createWeriftPeerConnection", "configuration", "config", "createWeriftMediaStream", "WebRtcTransport", "options", "level", "message", "meta", "output", "listener", "sessionId", "session", "peerUrl", "request", "rtcRequest", "ownedSessions", "port", "host", "signalingPath", "char", "code", "closed", "closePromise", "pendingAuthorizations", "httpServer", "req", "res", "sendJson", "status", "body", "parsedUrl", "controller", "onDisconnect", "onAuthorizationAbort", "headers", "name", "value", "allowed", "raw", "chunk", "error", "pc", "payload", "answer", "localDescription", "resolve", "reject", "onError", "onListening", "onAbort", "closeServer", "stopped", "id", "existing", "inFlight", "connectionPromise", "signalingUrl", "channel", "trackListeners", "trackEndedListeners", "streamListeners", "event", "track", "stream", "kind", "sender", "s", "item", "pending", "operation", "changed", "receiveAudio", "receiveVideo", "ensureReceive", "enabled", "signal", "offer", "timer", "abortListener", "response", "text", "description", "previous", "active", "transport", "timeoutMs", "_resolve", "settled", "cleanup", "onStateChange", "finish", "onOpen", "onClose", "requestedTimeout", "requestId", "notifyRemote", "onRequest", "messageEvent", "configured", "isFactory", "ownsStream", "prefix"]
+  "sourcesContent": ["import type {\n  HyperTransport,\n  TransportListenOptions,\n  TransportRequest,\n  TransportResponse,\n  TransportServer,\n} from \"@hyperttp/types\";\nimport {\n  MAX_SIGNALING_BODY_BYTES,\n  type ClientSession,\n  type WebRtcServerSessionContext,\n  type WebRtcServerSessionEvent,\n  type PendingResponse,\n  type ServerSession,\n  type WebRtcSessionView,\n  type WebRtcTrackEvent,\n  type WebRtcTransportOptions as InternalWebRtcTransportOptions,\n  type WebRtcTransportRequest,\n} from \"./types/types.js\";\nimport { abortError, decodeWireBody, encodeWireBody, errorFromUnknown } from \"./utils/helper.js\";\nimport { createServer, type Server as HttpServer } from \"node:http\";\nimport {\n  MediaStream as WeriftMediaStream,\n  RTCPeerConnection as WeriftPeerConnection,\n  RTCDataChannel as WeriftDataChannel,\n  type MediaStreamTrack as WeriftMediaStreamTrack,\n} from \"werift\";\n\ndeclare module \"@hyperttp/types\" {\n  interface TransportServer {\n    readonly webrtc?: Pick<\n      WebRtcTransport,\n      \"getSession\" | \"getServerSession\" | \"onTrack\" | \"closeServerSession\" | \"onServerSession\"\n    >;\n  }\n}\n\ntype WeriftPeerConfig = NonNullable<ConstructorParameters<typeof WeriftPeerConnection>[0]>;\ntype WeriftIceServers = WeriftPeerConfig[\"iceServers\"];\n\nfunction normalizeIceServers(servers: RTCIceServer[] | undefined): WeriftIceServers {\n  const normalized = (servers ?? []).flatMap((server) => {\n    const urls = Array.isArray(server.urls) ? server.urls : [server.urls];\n    if (server.credential !== undefined && typeof server.credential !== \"string\") {\n      throw new TypeError(\"Werift currently requires ICE server credentials to be strings.\");\n    }\n    return urls.map((url) => ({\n      urls: url,\n      ...(server.username !== undefined ? { username: server.username } : {}),\n      ...(server.credential !== undefined ? { credential: server.credential } : {}),\n    }));\n  });\n  return normalized as unknown as WeriftIceServers;\n}\n\nfunction createWeriftPeerConnection(configuration: RTCConfiguration): WeriftPeerConnection {\n  const config = {\n    iceServers: normalizeIceServers(configuration.iceServers),\n    ...(configuration.iceTransportPolicy !== undefined\n      ? { iceTransportPolicy: configuration.iceTransportPolicy }\n      : {}),\n  } as unknown as WeriftPeerConfig;\n  return new WeriftPeerConnection(config);\n}\n\nfunction createWeriftMediaStream(): WeriftMediaStream {\n  return new WeriftMediaStream();\n}\n\nexport type WebRtcTransportOptions = InternalWebRtcTransportOptions;\n\nexport interface WebRtcServerSessionView {\n  readonly id: string;\n  readonly context: WebRtcServerSessionContext;\n  readonly peerConnection: WeriftPeerConnection;\n  readonly remoteStream: WeriftMediaStream;\n  readonly localStream?: WeriftMediaStream;\n}\n\nexport interface WebRtcTransportServer extends TransportServer {\n  readonly webrtc: Pick<\n    WebRtcTransport,\n    \"getSession\" | \"getServerSession\" | \"onTrack\" | \"closeServerSession\" | \"onServerSession\"\n  >;\n}\n\nexport class WebRtcTransport implements HyperTransport {\n  private readonly sessions = new Map<string, ClientSession>();\n  private readonly connecting = new Map<string, Promise<ClientSession>>();\n  private readonly serverSessions = new Map<string, ServerSession>();\n  private readonly sessionListeners = new Set<(event: WebRtcServerSessionEvent) => void>();\n  private readonly closingServerSessions = new Map<string, Promise<void>>();\n  private readonly trackListeners = new Set<(event: WebRtcTrackEvent) => void>();\n  private readonly serverClosers = new Set<() => Promise<void>>();\n  private requestSequence = 0;\n  private closing = false;\n  private readonly pendingPeers = new Set<WeriftPeerConnection>();\n  private readonly iceServers: RTCIceServer[];\n  private readonly iceGatheringTimeoutMs: number;\n  private readonly signalingTimeoutMs: number;\n  private readonly connectionTimeoutMs: number;\n  private readonly requestTimeoutMs: number;\n  readonly createMediaStream: () => WeriftMediaStream;\n  private readonly createPeerConnection: (configuration: RTCConfiguration) => WeriftPeerConnection;\n  private readonly options: InternalWebRtcTransportOptions;\n\n  constructor(options: WebRtcTransportOptions = {}) {\n    this.options = options as unknown as InternalWebRtcTransportOptions;\n    this.createPeerConnection = options.createPeerConnection ?? createWeriftPeerConnection;\n    this.createMediaStream =\n      (options.createMediaStream as unknown as (() => WeriftMediaStream) | undefined) ??\n      createWeriftMediaStream;\n    this.iceServers = options.iceServers ?? [{ urls: \"stun:stun.l.google.com:19302\" }];\n    this.iceGatheringTimeoutMs = options.iceGatheringTimeoutMs ?? 10_000;\n    this.signalingTimeoutMs = options.signalingTimeoutMs ?? 10_000;\n    this.connectionTimeoutMs = options.connectionTimeoutMs ?? 15_000;\n    this.requestTimeoutMs = options.requestTimeoutMs ?? 15_000;\n  }\n\n  private log(level: \"debug\" | \"info\" | \"warn\" | \"error\", message: string, meta?: unknown): void {\n    if (this.options.logger) {\n      this.options.logger(level, message, meta);\n      return;\n    }\n    if (!this.options.verbose && level !== \"error\") return;\n    const output = level === \"debug\" ? console.debug : console[level];\n    if (meta === undefined) output(message);\n    else output(message, meta);\n  }\n\n  onTrack(listener: (event: WebRtcTrackEvent) => void): () => void {\n    this.trackListeners.add(listener);\n    return () => this.trackListeners.delete(listener);\n  }\n\n  onServerSession(listener: (event: WebRtcServerSessionEvent) => void): () => void {\n    this.sessionListeners.add(listener);\n    return () => this.sessionListeners.delete(listener);\n  }\n\n  private emitServerSession(session: ServerSession, type: \"created\" | \"closed\") {\n    for (const listener of this.sessionListeners) {\n      try {\n        listener({ ...session.context, type });\n      } catch (error) {\n        this.log(\"error\", \"WebRTC session listener failed\", error);\n      }\n    }\n  }\n\n  getServerSession(sessionId: string): WebRtcServerSessionView | undefined {\n    const session = this.serverSessions.get(sessionId);\n    if (!session || session.closed) return undefined;\n    return {\n      id: session.id,\n      context: session.context,\n      peerConnection: session.peerConnection,\n      remoteStream: session.remoteStream,\n      ...(session.localStream ? { localStream: session.localStream } : {}),\n    };\n  }\n\n  getSession(peerUrl: string): WebRtcSessionView | undefined {\n    const session = this.sessions.get(peerUrl);\n    if (!session || session.closed) return undefined;\n\n    return {\n      id: session.id,\n      peerUrl: session.peerUrl,\n      peerConnection: session.peerConnection,\n      dataChannel: session.dataChannel,\n      remoteStream: session.remoteStream,\n      media: session.mediaController,\n      ...(session.localStream ? { localStream: session.localStream } : {}),\n    };\n  }\n\n  async execute(request: TransportRequest): Promise<TransportResponse> {\n    if (this.closing) throw new Error(\"WebRTC transport is closed\");\n    const rtcRequest = request as WebRtcTransportRequest;\n    const peerUrl = rtcRequest.peerUrl ?? request.url;\n\n    if (request.signal?.aborted) throw abortError(request.signal);\n\n    const session = await this.getOrCreateSession(peerUrl, rtcRequest);\n    await this.ensureMedia(session, rtcRequest);\n\n    if (request.signal?.aborted) throw abortError(request.signal);\n\n    return this.sendRequest(session, request, rtcRequest.timeout);\n  }\n\n  public async listen(options: TransportListenOptions): Promise<WebRtcTransportServer> {\n    if (this.closing) throw new Error(\"WebRTC transport is closed\");\n    const ownedSessions = new Set<string>();\n    const port = options.port ?? 3000;\n    const host = options.host ?? \"127.0.0.1\";\n    const signalingPath = options.signaling?.path ?? \"/signal\";\n    if (\n      !signalingPath.startsWith(\"/\") ||\n      signalingPath.startsWith(\"//\") ||\n      /[?#\\\\\\s]/u.test(signalingPath) ||\n      Array.from(signalingPath).some((char) => {\n        const code = char.codePointAt(0)!;\n        return code <= 0x1f || (code >= 0x7f && code <= 0x9f);\n      }) ||\n      new URL(signalingPath, \"http://localhost\").pathname !== signalingPath\n    ) {\n      throw new TypeError(\n        \"WebRTC signaling path must be an absolute literal pathname without query or fragment\",\n      );\n    }\n\n    if (options.signal?.aborted) {\n      throw new Error(\"WebRtcTransport: server startup aborted\");\n    }\n\n    let closed = false;\n    let closePromise: Promise<void> | undefined;\n    const pendingAuthorizations = new Set<AbortController>();\n\n    const httpServer: HttpServer = createServer((req, res) => {\n      void (async () => {\n        const sendJson = (status: number, body: unknown) => {\n          if (res.headersSent || res.destroyed) return;\n          res.writeHead(status, { \"content-type\": \"application/json\" });\n          res.end(JSON.stringify(body));\n        };\n\n        if (closed || this.closing) {\n          sendJson(503, { error: \"WebRTC server is closing\" });\n          return;\n        }\n\n        let parsedUrl: URL;\n        try {\n          parsedUrl = new URL(req.url ?? \"/\", \"http://localhost\");\n        } catch {\n          sendJson(400, { error: \"Invalid request URL\" });\n          return;\n        }\n\n        if (parsedUrl.pathname !== signalingPath) {\n          sendJson(404, { error: \"Signaling endpoint not found\" });\n          return;\n        }\n\n        let ownerId: string | undefined;\n        let authorization: unknown;\n        if ((req.method === \"POST\" || req.method === \"DELETE\") && options.signaling?.authorize) {\n          const controller = new AbortController();\n          const onDisconnect = () => controller.abort();\n          const onAuthorizationAbort = () => {\n            sendJson(503, { error: \"WebRTC signaling request aborted\" });\n          };\n          req.once(\"aborted\", onDisconnect);\n          res.once(\"close\", onDisconnect);\n          controller.signal.addEventListener(\"abort\", onAuthorizationAbort, { once: true });\n          pendingAuthorizations.add(controller);\n          try {\n            const headers: Record<string, string | string[]> = {};\n            for (const [name, value] of Object.entries(req.headers)) {\n              if (value !== undefined) headers[name] = value;\n            }\n            const allowed = await options.signaling.authorize({\n              protocol: \"webrtc\",\n              method: req.method,\n              url: req.url ?? \"/\",\n              headers,\n              signal: controller.signal,\n            });\n            if (controller.signal.aborted || req.aborted || res.destroyed) return;\n            if (closed || this.closing || options.signal?.aborted) {\n              sendJson(503, { error: \"WebRTC server is closing\" });\n              return;\n            }\n            if (typeof allowed === \"object\" && allowed !== null) {\n              if (typeof allowed.ownerId !== \"string\" || !allowed.ownerId.trim()) {\n                sendJson(403, { error: \"Invalid signaling identity\" });\n                return;\n              }\n              ownerId = allowed.ownerId;\n              authorization = allowed.context;\n            } else if (allowed !== true) {\n              sendJson(403, { error: \"Signaling forbidden\" });\n              return;\n            }\n          } catch {\n            sendJson(403, { error: \"Signaling forbidden\" });\n            return;\n          } finally {\n            pendingAuthorizations.delete(controller);\n            req.off(\"aborted\", onDisconnect);\n            res.off(\"close\", onDisconnect);\n            controller.signal.removeEventListener(\"abort\", onAuthorizationAbort);\n          }\n        }\n\n        if (req.method === \"DELETE\") {\n          const sessionId = parsedUrl.searchParams.get(\"sessionId\");\n          const existing = sessionId ? this.serverSessions.get(sessionId) : undefined;\n          if (\n            existing &&\n            (!ownedSessions.has(sessionId!) || existing.context.ownerId !== ownerId)\n          ) {\n            sendJson(403, { error: \"Signaling forbidden\" });\n            return;\n          }\n          if (sessionId && ownedSessions.has(sessionId)) {\n            await this.closeServerSession(sessionId);\n            ownedSessions.delete(sessionId);\n          }\n          res.writeHead(204).end();\n          return;\n        }\n\n        if (req.method !== \"POST\") {\n          sendJson(405, { error: \"Method not allowed\" });\n          return;\n        }\n\n        let raw = \"\";\n        try {\n          for await (const chunk of req) {\n            raw += Buffer.isBuffer(chunk) ? chunk.toString(\"utf8\") : String(chunk);\n            if (Buffer.byteLength(raw, \"utf8\") > MAX_SIGNALING_BODY_BYTES) {\n              sendJson(413, { error: \"Signaling payload too large\" });\n              return;\n            }\n          }\n        } catch (error) {\n          sendJson(400, { error: errorFromUnknown(error, \"Failed to read body\").message });\n          return;\n        }\n\n        let pc: WeriftPeerConnection | undefined;\n        let session: ServerSession | undefined;\n\n        try {\n          const payload = JSON.parse(raw) as {\n            sessionId?: unknown;\n            sdp?: RTCSessionDescriptionInit;\n          };\n\n          if (\n            typeof payload.sessionId !== \"string\" ||\n            payload.sessionId.length < 1 ||\n            payload.sessionId.length > 200 ||\n            payload.sdp?.type !== \"offer\" ||\n            typeof payload.sdp.sdp !== \"string\"\n          ) {\n            sendJson(400, { error: \"Expected sessionId and a valid SDP offer\" });\n            return;\n          }\n\n          if (closed || this.closing || options.signal?.aborted || req.aborted || res.destroyed) {\n            throw new Error(\"WebRTC signaling request aborted\");\n          }\n          const sessionId = payload.sessionId;\n          session = this.serverSessions.get(sessionId);\n          if (session && !ownedSessions.has(sessionId)) {\n            sendJson(409, { error: \"Session belongs to another listener\" });\n            return;\n          }\n          if (session && session.context.ownerId !== ownerId) {\n            sendJson(403, { error: \"Signaling forbidden\" });\n            return;\n          }\n          if (this.closingServerSessions.has(sessionId)) {\n            sendJson(409, { error: \"Session is closing\" });\n            return;\n          }\n          if (session?.negotiating) {\n            sendJson(409, { error: \"Session negotiation already in progress\" });\n            return;\n          }\n\n          if (!session || session.closed) {\n            pc = this.createPeerConnection({ iceServers: this.iceServers });\n            const controller = new AbortController();\n            session = {\n              controller,\n              context: Object.freeze({\n                sessionId,\n                ownerId,\n                authorization,\n                signal: controller.signal,\n              }),\n              release: () => ownedSessions.delete(sessionId),\n              id: sessionId,\n              peerConnection: pc,\n              remoteStream: this.createMediaStream(),\n              localTrackIds: new Set<string>(),\n              localStreamResolved: false,\n              ownsLocalStream: false,\n              closed: false,\n            };\n            ownedSessions.add(sessionId);\n            this.serverSessions.set(sessionId, session);\n            this.attachServerPeerEvents(session, options.onRequest);\n            this.emitServerSession(session, \"created\");\n          } else {\n            pc = session.peerConnection;\n          }\n\n          session.negotiating = true;\n          const activeSession = session;\n          const disconnect = () => {\n            if (!res.writableEnded) void this.closeServerSession(activeSession.id);\n          };\n          res.once(\"close\", disconnect);\n          const onSessionAbort = () => sendJson(503, { error: \"WebRTC session closed\" });\n          activeSession.controller.signal.addEventListener(\"abort\", onSessionAbort, { once: true });\n          res.once(\"finish\", () => {\n            activeSession.controller.signal.removeEventListener(\"abort\", onSessionAbort);\n            res.off(\"close\", disconnect);\n          });\n          await this.setRemoteDescription(pc, payload.sdp);\n          await this.attachConfiguredServerStream(session);\n          if (session.closed) throw new Error(\"WebRTC session closed\");\n\n          const answer = await pc.createAnswer();\n          await this.withDeadline(pc.setLocalDescription(answer), this.iceGatheringTimeoutMs);\n          if (!(await this.waitForIceGatheringComplete(pc, this.iceGatheringTimeoutMs)))\n            throw new Error(\"Full ICE gathering timed out; trickle ICE is not supported\");\n\n          const localDescription = pc.localDescription;\n          if (!localDescription?.sdp || localDescription.type !== \"answer\") {\n            throw new Error(\"Local SDP answer was not generated\");\n          }\n\n          if (closed || session.closed) throw new Error(\"WebRTC session closed\");\n          session.negotiating = false;\n          sendJson(200, localDescription);\n        } catch (error) {\n          this.log(\"error\", \"[WebRtcTransport Server] \u041E\u0448\u0438\u0431\u043A\u0430 \u0441\u0438\u0433\u043D\u0430\u043B\u0438\u0437\u0430\u0446\u0438\u0438:\", error);\n          if (session) {\n            await this.closeServerSession(session.id);\n          } else if (pc) {\n            try {\n              await pc.close();\n            } catch {}\n          }\n          sendJson(400, { error: errorFromUnknown(error, \"Signaling failed\").message });\n        }\n      })();\n    });\n\n    await new Promise<void>((resolve, reject) => {\n      const onError = (error: Error) => {\n        httpServer.off(\"listening\", onListening);\n        reject(error);\n      };\n      const onListening = () => {\n        httpServer.off(\"error\", onError);\n        resolve();\n      };\n      httpServer.once(\"error\", onError);\n      httpServer.once(\"listening\", onListening);\n      httpServer.listen(port, host);\n    });\n\n    this.log(\"info\", `[WebRtcTransport] Listening on http://${host}:${port}${signalingPath}`);\n\n    const onAbort = () => {\n      void closeServer().catch((error) => {\n        this.log(\"error\", \"[WebRtcTransport] Server shutdown failed:\", error);\n      });\n    };\n\n    const closeServer = (): Promise<void> => {\n      if (closePromise) return closePromise;\n      closed = true;\n      for (const controller of pendingAuthorizations) controller.abort();\n      options.signal?.removeEventListener(\"abort\", onAbort);\n      closePromise = (async () => {\n        const stopped = httpServer.listening\n          ? new Promise<void>((resolve, reject) => {\n              httpServer.close((error) => (error ? reject(error) : resolve()));\n            })\n          : Promise.resolve();\n        await Promise.all([...ownedSessions].map((id) => this.closeServerSession(id)));\n        ownedSessions.clear();\n        await stopped;\n        this.serverClosers.delete(closeServer);\n      })();\n      return closePromise;\n    };\n\n    options.signal?.addEventListener(\"abort\", onAbort, { once: true });\n    this.serverClosers.add(closeServer);\n    if (options.signal?.aborted && !closed) await closeServer();\n\n    return {\n      close: closeServer,\n      webrtc: {\n        getSession: this.getSession.bind(this),\n        getServerSession: this.getServerSession.bind(this),\n        onTrack: this.onTrack.bind(this),\n        closeServerSession: (id: string) =>\n          ownedSessions.has(id) ? this.closeServerSession(id) : Promise.resolve(),\n        onServerSession: this.onServerSession.bind(this),\n      },\n    };\n  }\n\n  async close(): Promise<void> {\n    this.closing = true;\n    await Promise.all([...this.pendingPeers].map((pc) => pc.close().catch(() => {})));\n    await Promise.allSettled(this.connecting.values());\n    await Promise.all(\n      [...this.sessions.values()].map((session) => this.disposeClientSession(session, true)),\n    );\n    await Promise.all([...this.serverClosers].map((closeServer) => closeServer()));\n  }\n\n  async destroy(): Promise<void> {\n    await this.close();\n  }\n\n  private async getOrCreateSession(\n    peerUrl: string,\n    request: WebRtcTransportRequest,\n  ): Promise<ClientSession> {\n    const existing = this.sessions.get(peerUrl);\n    if (\n      existing &&\n      !existing.closed &&\n      existing.dataChannel.readyState === \"open\" &&\n      existing.peerConnection.connectionState !== \"failed\" &&\n      existing.peerConnection.connectionState !== \"closed\"\n    ) {\n      return existing;\n    }\n\n    if (existing) {\n      await this.disposeClientSession(existing, true);\n    }\n\n    const inFlight = this.connecting.get(peerUrl);\n    if (inFlight) return inFlight;\n\n    const connectionPromise = this.createClientSession(peerUrl, request);\n    this.connecting.set(peerUrl, connectionPromise);\n\n    try {\n      const session = await connectionPromise;\n      if (this.closing || session.closed) {\n        await this.disposeClientSession(session, true);\n        throw new Error(\"WebRTC transport is closed\");\n      }\n      this.sessions.set(peerUrl, session);\n      return session;\n    } finally {\n      if (this.connecting.get(peerUrl) === connectionPromise) {\n        this.connecting.delete(peerUrl);\n      }\n    }\n  }\n\n  private async createClientSession(\n    peerUrl: string,\n    request: WebRtcTransportRequest,\n  ): Promise<ClientSession> {\n    const signalingUrl = this.toSignalingUrl(peerUrl);\n    const id = this.createId(\"session\");\n    const pc = this.createPeerConnection({\n      iceServers: request.iceServers ?? this.iceServers,\n    });\n    this.pendingPeers.add(pc);\n    const channel = pc.createDataChannel(request.channelLabel ?? \"hyperttp-webrtc-transport\", {\n      ordered: true,\n    });\n\n    const session: ClientSession = {\n      id,\n      peerUrl,\n      signalingUrl,\n      peerConnection: pc,\n      dataChannel: channel,\n      remoteStream: this.createMediaStream(),\n      ...(request.localStream ? { localStream: request.localStream } : {}),\n      localTrackIds: new Set<string>(),\n      pending: new Map<string, PendingResponse>(),\n      negotiation: Promise.resolve(),\n      closed: false,\n      mediaController: {} as any,\n    };\n\n    const trackListeners = new Set<\n      (e: { track: WeriftMediaStreamTrack; stream: WeriftMediaStream }) => void\n    >();\n    const trackEndedListeners = new Set<(e: { track: WeriftMediaStreamTrack }) => void>();\n    const streamListeners = new Set<(e: { stream: WeriftMediaStream }) => void>();\n\n    pc.addEventListener(\"track\", (event: WebRtcTrackEvent) => {\n      const { track } = event;\n      if (!session.remoteStream.getTracks().includes(track)) session.remoteStream.addTrack(track);\n      for (const listener of trackListeners) listener({ track, stream: session.remoteStream });\n      for (const listener of streamListeners) listener({ stream: session.remoteStream });\n      track.addEventListener(\"ended\", () => {\n        for (const listener of trackEndedListeners) listener({ track });\n      });\n    });\n\n    session.mediaController = {\n      localStream: session.localStream,\n      remoteStream: session.remoteStream,\n      addTrack: (track, stream) =>\n        this.queueNegotiation(session, async () => {\n          pc.addTrack(track, stream ?? session.localStream ?? this.createMediaStream());\n          if (track.id) session.localTrackIds.add(track.id);\n          await this.negotiate(session);\n        }),\n      replaceTrack: async (kind: \"audio\" | \"video\", track: any | null) => {\n        const sender = pc.getSenders().find((s: any) => s.track && s.track.kind === kind);\n        if (sender) {\n          await sender.replaceTrack(track);\n        }\n      },\n      removeTrack: (track) =>\n        this.queueNegotiation(session, async () => {\n          const sender = pc.getSenders().find((item) => item.track === track);\n          if (sender) {\n            pc.removeTrack(sender);\n            if (track.id) session.localTrackIds.delete(track.id);\n            await this.negotiate(session);\n          }\n        }),\n      close: async () => {\n        await this.disposeClientSession(session, true);\n      },\n      on(event: string, listener: any) {\n        if (event === \"track\") trackListeners.add(listener);\n        if (event === \"trackEnded\") trackEndedListeners.add(listener);\n        if (event === \"stream\") streamListeners.add(listener);\n        return () => {\n          trackListeners.delete(listener);\n          trackEndedListeners.delete(listener);\n          streamListeners.delete(listener);\n        };\n      },\n    };\n\n    try {\n      this.attachClientPeerEvents(session);\n      this.addLocalTracks(session, request.localStream);\n      this.addReceiveTransceivers(session, request.receiveAudio, request.receiveVideo);\n      await this.negotiate(session, request.signal);\n      await this.waitForChannelOpen(channel, this.connectionTimeoutMs, request.signal);\n      return session;\n    } catch (error) {\n      await this.disposeClientSession(session, true);\n      throw error;\n    } finally {\n      this.pendingPeers.delete(pc);\n    }\n  }\n\n  private attachClientPeerEvents(session: ClientSession): void {\n    const { peerConnection: pc, dataChannel: channel } = session;\n\n    pc.addEventListener(\"track\", (event: any) => {\n      const track = event.track;\n      if (!session.remoteStream.getTracks().some((item: any) => item.id === track.id)) {\n        session.remoteStream.addTrack(track);\n      }\n\n      this.emitTrack({\n        side: \"client\",\n        sessionId: session.id,\n        peerUrl: session.peerUrl,\n        peerConnection: pc,\n        track,\n        stream: session.remoteStream,\n        streams: event.streams,\n      });\n    });\n\n    pc.addEventListener(\"connectionstatechange\", () => {\n      if (pc.connectionState === \"failed\" || pc.connectionState === \"closed\") {\n        void this.disposeClientSession(session, true);\n      }\n    });\n\n    channel.addEventListener(\"message\", (event: MessageEvent) => {\n      let message: Record<string, unknown>;\n      try {\n        message = JSON.parse(String(event.data)) as Record<string, unknown>;\n      } catch {\n        return;\n      }\n\n      if (!message || typeof message !== \"object\" || typeof message.requestId !== \"string\") return;\n      const pending = session.pending.get(message.requestId);\n      if (!pending) return;\n\n      try {\n        const body = decodeWireBody(message.body);\n        this.finishPending(session, message.requestId);\n        pending.resolve({\n          status: typeof message.status === \"number\" ? message.status : 200,\n          statusText: typeof message.statusText === \"string\" ? message.statusText : \"OK\",\n          headers:\n            message.headers && typeof message.headers === \"object\"\n              ? (message.headers as TransportResponse[\"headers\"])\n              : {},\n          body,\n          url: typeof message.url === \"string\" ? message.url : session.peerUrl,\n        });\n      } catch (error) {\n        this.finishPending(session, message.requestId);\n        pending.reject(errorFromUnknown(error, \"Invalid WebRTC response\"));\n      }\n    });\n\n    channel.addEventListener(\"close\", () => {\n      this.rejectAllPending(session, new Error(\"WebRTC DataChannel closed\"));\n      void this.disposeClientSession(session, true);\n    });\n\n    channel.addEventListener(\"error\", () => {\n      this.rejectAllPending(session, new Error(\"WebRTC DataChannel error\"));\n    });\n  }\n\n  private queueNegotiation(session: ClientSession, operation: () => Promise<void>): Promise<void> {\n    const pending = session.negotiation\n      .catch(() => undefined)\n      .then(async () => {\n        if (session.closed) throw new Error(\"WebRTC session closed\");\n        try {\n          await operation();\n        } catch (error) {\n          await this.disposeClientSession(session, true);\n          throw error;\n        }\n      });\n    session.negotiation = pending;\n    return pending;\n  }\n\n  private async ensureMedia(\n    session: ClientSession,\n    request: WebRtcTransportRequest,\n  ): Promise<void> {\n    const needsPotentialChanges =\n      Boolean(request.localStream) ||\n      Boolean(request.receiveAudio) ||\n      Boolean(request.receiveVideo);\n    if (!needsPotentialChanges || session.closed) return;\n\n    await this.queueNegotiation(session, async () => {\n      let changed = this.addLocalTracks(session, request.localStream);\n      changed =\n        this.addReceiveTransceivers(session, request.receiveAudio, request.receiveVideo) || changed;\n\n      if (request.localStream) {\n        session.localStream = request.localStream;\n        (session.mediaController as { localStream?: WeriftMediaStream }).localStream =\n          request.localStream;\n      }\n      if (changed) await this.negotiate(session, request.signal);\n    });\n  }\n\n  private addLocalTracks(session: ClientSession, stream?: WeriftMediaStream): boolean {\n    if (!stream) return false;\n    let changed = false;\n\n    for (const track of stream.getTracks()) {\n      if (track.readyState !== \"live\" || session.localTrackIds.has(track.id!)) continue;\n      session.peerConnection.addTrack(track, stream);\n      session.localTrackIds.add(track.id!);\n      changed = true;\n    }\n\n    return changed;\n  }\n\n  private addReceiveTransceivers(\n    session: ClientSession,\n    receiveAudio?: boolean,\n    receiveVideo?: boolean,\n  ): boolean {\n    const pc = session.peerConnection;\n    let changed = false;\n\n    const ensureReceive = (kind: \"audio\" | \"video\", enabled: boolean | undefined) => {\n      if (!enabled) return;\n      const transceivers = pc\n        .getTransceivers()\n        .find(\n          (item: any) =>\n            item.receiver.track.kind === kind &&\n            (item.direction === \"recvonly\" || item.direction === \"sendrecv\"),\n        );\n      if (!transceivers) {\n        pc.addTransceiver(kind, { direction: \"recvonly\" });\n        changed = true;\n      }\n    };\n\n    ensureReceive(\"audio\", receiveAudio);\n    ensureReceive(\"video\", receiveVideo);\n\n    return changed;\n  }\n\n  private async negotiate(session: ClientSession, signal?: AbortSignal): Promise<void> {\n    if (signal?.aborted) throw abortError(signal);\n\n    const pc = session.peerConnection;\n    if (pc.signalingState !== \"stable\") {\n      throw new Error(`Cannot negotiate WebRTC session in state ${pc.signalingState}`);\n    }\n\n    const offer = await pc.createOffer();\n    await this.withDeadline(pc.setLocalDescription(offer), this.iceGatheringTimeoutMs, signal);\n    await this.waitForIceGatheringComplete(pc, this.iceGatheringTimeoutMs);\n\n    const localDescription = pc.localDescription;\n    if (!localDescription?.sdp || localDescription.type !== \"offer\") {\n      throw new Error(\"Local SDP offer was not generated\");\n    }\n\n    const controller = new AbortController();\n    const timer = setTimeout(() => controller.abort(), this.signalingTimeoutMs);\n    const abortListener = () => controller.abort(signal?.reason);\n    signal?.addEventListener(\"abort\", abortListener, { once: true });\n    if (signal?.aborted) abortListener();\n\n    try {\n      const response = await fetch(session.signalingUrl, {\n        method: \"POST\",\n        headers: { \"content-type\": \"application/json\" },\n        body: JSON.stringify({ sessionId: session.id, sdp: localDescription }),\n        signal: controller.signal,\n      });\n\n      if (!response.ok) {\n        const text = await response.text();\n        throw new Error(\n          `Signaling HTTP error: ${response.status} ${response.statusText}${text ? ` \u2014 ${text}` : \"\"}`,\n        );\n      }\n\n      const answer = (await response.json()) as RTCSessionDescriptionInit;\n      if (answer.type !== \"answer\" || typeof answer.sdp !== \"string\") {\n        throw new Error(\"Signaling server returned an invalid SDP answer\");\n      }\n\n      await this.setRemoteDescription(pc, answer);\n    } catch (error) {\n      if (signal?.aborted) throw abortError(signal);\n      if (controller.signal.aborted) {\n        throw new Error(`Signaling timed out after ${this.signalingTimeoutMs} ms`);\n      }\n      throw errorFromUnknown(error, \"WebRTC signaling failed\");\n    } finally {\n      clearTimeout(timer);\n      signal?.removeEventListener(\"abort\", abortListener);\n    }\n  }\n\n  private async setRemoteDescription(\n    pc: WeriftPeerConnection,\n    description: RTCSessionDescriptionInit,\n  ): Promise<void> {\n    const previous = new Set(pc.dtlsTransports);\n    try {\n      await pc.setRemoteDescription(description);\n    } finally {\n      // Werift can replace transports while applying BUNDLE without stopping the\n      // displaced ICE sockets. They are no longer reachable by pc.close().\n      const active = new Set(pc.dtlsTransports);\n      await Promise.all(\n        [...previous]\n          .filter((transport) => !active.has(transport))\n          .map((transport) => transport.stop()),\n      );\n    }\n  }\n\n  private async withDeadline<T>(\n    operation: Promise<T>,\n    timeoutMs: number,\n    signal?: AbortSignal,\n  ): Promise<T> {\n    let timer: ReturnType<typeof setTimeout> | undefined;\n    let onAbort: (() => void) | undefined;\n    try {\n      return await Promise.race([\n        operation,\n        new Promise<never>((_resolve, reject) => {\n          timer = setTimeout(\n            () => reject(new Error(`WebRTC operation timed out after ${timeoutMs} ms`)),\n            timeoutMs,\n          );\n          onAbort = () => reject(abortError(signal));\n          signal?.addEventListener(\"abort\", onAbort, { once: true });\n          if (signal?.aborted) onAbort();\n        }),\n      ]);\n    } finally {\n      clearTimeout(timer);\n      if (onAbort) signal?.removeEventListener(\"abort\", onAbort);\n    }\n  }\n\n  private async waitForIceGatheringComplete(\n    pc: WeriftPeerConnection,\n    timeoutMs: number,\n  ): Promise<boolean> {\n    if (pc.iceGatheringState === \"complete\") return true;\n\n    const completed = await new Promise<boolean>((resolve) => {\n      let settled = false;\n      let timer: ReturnType<typeof setTimeout>;\n\n      const cleanup = () => {\n        clearTimeout(timer);\n        pc.removeEventListener(\"icegatheringstatechange\", onStateChange);\n      };\n\n      const finish = (value: boolean) => {\n        if (settled) return;\n        settled = true;\n        cleanup();\n        resolve(value);\n      };\n\n      const onStateChange = () => {\n        if (pc.iceGatheringState === \"complete\") finish(true);\n      };\n\n      timer = setTimeout(() => finish(false), timeoutMs);\n      pc.addEventListener(\"icegatheringstatechange\", onStateChange);\n      onStateChange();\n    });\n\n    if (!completed) throw new Error(`ICE gathering timed out after ${timeoutMs} ms`);\n    return true;\n  }\n\n  private async waitForChannelOpen(\n    channel: WeriftDataChannel,\n    timeoutMs: number,\n    signal?: AbortSignal,\n  ): Promise<void> {\n    if (channel.readyState === \"open\") return;\n    if (signal?.aborted) throw abortError(signal);\n\n    await new Promise<void>((resolve, reject) => {\n      let settled = false;\n      const cleanup = () => {\n        clearTimeout(timer);\n        channel.removeEventListener(\"open\", onOpen);\n        channel.removeEventListener(\"error\", onError);\n        channel.removeEventListener(\"close\", onClose);\n        signal?.removeEventListener(\"abort\", onAbort);\n      };\n      const finish = (error?: Error) => {\n        if (settled) return;\n        settled = true;\n        cleanup();\n        if (error) reject(error);\n        else resolve();\n      };\n      const onOpen = () => finish();\n      const onError = () => finish(new Error(\"WebRTC DataChannel failed to open\"));\n      const onClose = () => finish(new Error(\"WebRTC DataChannel closed before opening\"));\n      const onAbort = () => finish(abortError(signal));\n      const timer = setTimeout(\n        () => finish(new Error(`DataChannel open timeout after ${timeoutMs} ms`)),\n        timeoutMs,\n      );\n\n      channel.addEventListener(\"open\", onOpen);\n      channel.addEventListener(\"error\", onError);\n      channel.addEventListener(\"close\", onClose);\n      signal?.addEventListener(\"abort\", onAbort, { once: true });\n\n      if (channel.readyState === \"open\") finish();\n      else if (channel.readyState === \"closed\") finish(new Error(\"WebRTC DataChannel is closed\"));\n    });\n  }\n\n  private sendRequest(\n    session: ClientSession,\n    request: TransportRequest,\n    requestedTimeout?: number,\n  ): Promise<TransportResponse> {\n    if (request.signal?.aborted) return Promise.reject(abortError(request.signal));\n    if (session.dataChannel.readyState !== \"open\") {\n      return Promise.reject(new Error(\"WebRTC DataChannel is not open\"));\n    }\n\n    const requestId = this.createId(`request${++this.requestSequence}`);\n    const timeoutMs =\n      requestedTimeout && requestedTimeout > 0 ? requestedTimeout : this.requestTimeoutMs;\n\n    return new Promise<TransportResponse>((resolve, reject) => {\n      const pending: PendingResponse = {\n        resolve,\n        reject,\n        timer: setTimeout(() => {\n          this.finishPending(session, requestId);\n          reject(new Error(`WebRTC request timed out after ${timeoutMs} ms`));\n        }, timeoutMs),\n        ...(request.signal ? { signal: request.signal } : {}),\n      };\n\n      if (request.signal) {\n        pending.abortListener = () => {\n          this.finishPending(session, requestId);\n          reject(abortError(request.signal));\n        };\n        request.signal.addEventListener(\"abort\", pending.abortListener, { once: true });\n      }\n\n      session.pending.set(requestId, pending);\n\n      try {\n        session.dataChannel.send(\n          JSON.stringify({\n            requestId,\n            method: request.method,\n            url: request.url,\n            headers: request.headers,\n            body: encodeWireBody(request.body),\n            protocol: request.protocol,\n          }),\n        );\n      } catch (error) {\n        this.finishPending(session, requestId);\n        reject(errorFromUnknown(error, \"Failed to send WebRTC request\"));\n      }\n    });\n  }\n\n  private finishPending(session: ClientSession, requestId: string): void {\n    const pending = session.pending.get(requestId);\n    if (!pending) return;\n    clearTimeout(pending.timer);\n    if (pending.signal && pending.abortListener) {\n      pending.signal.removeEventListener(\"abort\", pending.abortListener);\n    }\n    session.pending.delete(requestId);\n  }\n\n  private rejectAllPending(session: ClientSession, error: Error): void {\n    for (const [requestId, pending] of session.pending) {\n      this.finishPending(session, requestId);\n      pending.reject(error);\n    }\n  }\n\n  private async disposeClientSession(session: ClientSession, notifyRemote: boolean): Promise<void> {\n    if (session.closed) return;\n    session.closed = true;\n\n    if (this.sessions.get(session.peerUrl) === session) {\n      this.sessions.delete(session.peerUrl);\n    }\n    this.rejectAllPending(session, new Error(\"WebRTC session closed\"));\n\n    try {\n      session.dataChannel.close();\n    } catch {}\n    try {\n      await session.peerConnection.close();\n    } catch {}\n\n    if (notifyRemote) {\n      try {\n        const url = new URL(session.signalingUrl);\n        url.searchParams.set(\"sessionId\", session.id);\n        const controller = new AbortController();\n        const timer = setTimeout(() => controller.abort(), 2_000);\n        try {\n          await fetch(url, { method: \"DELETE\", signal: controller.signal });\n        } finally {\n          clearTimeout(timer);\n        }\n      } catch {}\n    }\n  }\n\n  private attachServerPeerEvents(\n    session: ServerSession,\n    onRequest: TransportListenOptions[\"onRequest\"],\n  ): void {\n    const pc = session.peerConnection;\n\n    pc.addEventListener(\"datachannel\", (event: any) => {\n      const channel = event.channel;\n      channel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n        void (async () => {\n          let requestId: string | undefined;\n          try {\n            const message = JSON.parse(String(messageEvent.data)) as {\n              requestId?: unknown;\n              method?: unknown;\n              url?: unknown;\n              headers?: unknown;\n              body?: unknown;\n              protocol?: unknown;\n            };\n\n            requestId = typeof message.requestId === \"string\" ? message.requestId : undefined;\n            if (\n              !requestId ||\n              typeof message.method !== \"string\" ||\n              typeof message.url !== \"string\"\n            ) {\n              throw new Error(\"Invalid WebRTC transport request\");\n            }\n\n            const request: TransportRequest = {\n              method: message.method,\n              url: message.url,\n              headers:\n                message.headers && typeof message.headers === \"object\"\n                  ? {\n                      ...(message.headers as TransportRequest[\"headers\"]),\n                      \"x-webrtc-session-id\": session.id,\n                    }\n                  : { \"x-webrtc-session-id\": session.id },\n              body: decodeWireBody(message.body),\n              protocol:\n                typeof message.protocol === \"string\"\n                  ? (message.protocol as TransportRequest[\"protocol\"])\n                  : \"webrtc\",\n            };\n\n            let response: TransportResponse;\n            if (!onRequest) {\n              response = {\n                status: 501,\n                statusText: \"Not Implemented\",\n                headers: { \"content-type\": \"application/json\" },\n                body: { error: \"No server request handler configured\" },\n                url: request.url,\n              };\n            } else {\n              response = await onRequest(request);\n            }\n\n            if (channel.readyState !== \"open\") return;\n            channel.send(\n              JSON.stringify({\n                requestId,\n                status: response.status,\n                statusText: response.statusText ?? \"\",\n                headers: response.headers,\n                body: encodeWireBody(response.body),\n                url: response.url ?? request.url,\n              }),\n            );\n          } catch (error) {\n            this.log(\"error\", \"[WebRtcTransport Server] Request failed:\", error);\n            if (requestId && channel.readyState === \"open\") {\n              try {\n                channel.send(\n                  JSON.stringify({\n                    requestId,\n                    status: 500,\n                    statusText: \"Internal Server Error\",\n                    headers: { \"content-type\": \"application/json\" },\n                    body: { error: errorFromUnknown(error, \"Request handler failed\").message },\n                  }),\n                );\n              } catch {}\n            }\n          }\n        })();\n      });\n    });\n\n    pc.addEventListener(\"track\", (event: any) => {\n      const track = event.track;\n      if (!session.remoteStream.getTracks().some((item: any) => item.id === track.id)) {\n        session.remoteStream.addTrack(track);\n      }\n      this.emitTrack({\n        side: \"server\",\n        sessionId: session.id,\n        peerConnection: pc,\n        track,\n        stream: session.remoteStream,\n        streams: event.streams,\n      });\n    });\n\n    pc.addEventListener(\"connectionstatechange\", () => {\n      if (pc.connectionState === \"failed\" || pc.connectionState === \"closed\") {\n        void this.closeServerSession(session.id);\n      }\n    });\n  }\n\n  private async attachConfiguredServerStream(session: ServerSession): Promise<void> {\n    if (session.closed) return;\n\n    if (!session.localStreamResolved) {\n      const configured = this.options.localStream;\n      const isFactory = typeof configured === \"function\";\n      const stream = isFactory ? await configured(session.id, session.context) : configured;\n      if (session.closed) {\n        if (stream && isFactory) {\n          session.localStream = stream;\n          session.ownsLocalStream = true;\n          this.disposeServerLocalStream(session);\n        }\n        return;\n      }\n      session.localStreamResolved = true;\n\n      if (stream) {\n        session.localStream = stream;\n        session.ownsLocalStream = isFactory;\n      }\n    }\n\n    const stream = session.localStream;\n    if (!stream) return;\n\n    const pc = session.peerConnection;\n    for (const track of stream.getTracks()) {\n      if (session.closed || track.readyState !== \"live\" || session.localTrackIds.has(track.id!)) {\n        continue;\n      }\n      pc.addTrack(track, stream);\n      session.localTrackIds.add(track.id!);\n    }\n  }\n\n  private disposeServerLocalStream(session: ServerSession): void {\n    const stream = session.localStream;\n    const ownsStream = session.ownsLocalStream;\n\n    delete session.localStream;\n    session.ownsLocalStream = false;\n\n    if (!stream || !ownsStream) return;\n\n    try {\n      this.options.disposeLocalStream?.(session.id, stream);\n    } catch {\n    } finally {\n      for (const track of stream.getTracks()) {\n        if (track.readyState !== \"ended\") track.stop();\n      }\n    }\n  }\n\n  /** Idempotent: concurrent closes share peer/resource cleanup. */\n  public closeServerSession(sessionId: string): Promise<void> {\n    const pending = this.closingServerSessions.get(sessionId);\n    if (pending) return pending;\n    const session = this.serverSessions.get(sessionId);\n    if (!session || session.closed) return Promise.resolve();\n    session.closed = true;\n    session.controller.abort();\n    session.release();\n    this.serverSessions.delete(sessionId);\n    const closing = Promise.resolve()\n      .then(async () => {\n        try {\n          await session.peerConnection.close();\n        } catch (error) {\n          this.log(\"warn\", \"WebRTC peer close failed\", error);\n        } finally {\n          this.disposeServerLocalStream(session);\n          this.emitServerSession(session, \"closed\");\n        }\n      })\n      .finally(() => this.closingServerSessions.delete(sessionId));\n    this.closingServerSessions.set(sessionId, closing);\n    return closing;\n  }\n\n  private emitTrack(event: WebRtcTrackEvent): void {\n    for (const listener of this.trackListeners) {\n      try {\n        listener(event);\n      } catch (error) {\n        this.log(\"error\", \"[WebRtcTransport] onTrack listener failed:\", error);\n      }\n    }\n  }\n\n  private toSignalingUrl(peerUrl: string): string {\n    if (peerUrl.startsWith(\"webrtcs://\")) return peerUrl.replace(/^webrtcs:\\/\\//, \"https://\");\n    if (peerUrl.startsWith(\"webrtc://\")) return peerUrl.replace(/^webrtc:\\/\\//, \"http://\");\n    throw new Error(`Unsupported WebRTC signaling URL: ${peerUrl}`);\n  }\n\n  private createId(prefix: string): string {\n    return `${prefix}-${crypto.randomUUID()}`;\n  }\n}\n"],
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+  "names": ["MAX_SIGNALING_BODY_BYTES", "abortError", "decodeWireBody", "encodeWireBody", "errorFromUnknown", "createServer", "WeriftMediaStream", "WeriftPeerConnection", "normalizeIceServers", "servers", "server", "urls", "url", "__name", "createWeriftPeerConnection", "configuration", "config", "createWeriftMediaStream", "WebRtcTransport", "options", "level", "message", "meta", "output", "listener", "session", "type", "error", "sessionId", "peerUrl", "request", "rtcRequest", "ownedSessions", "port", "host", "signalingPath", "char", "code", "closed", "closePromise", "pendingAuthorizations", "httpServer", "req", "res", "sendJson", "status", "body", "parsedUrl", "ownerId", "authorization", "controller", "onDisconnect", "onAuthorizationAbort", "headers", "name", "value", "allowed", "existing", "raw", "chunk", "pc", "payload", "activeSession", "disconnect", "onSessionAbort", "answer", "localDescription", "resolve", "reject", "onError", "onListening", "onAbort", "closeServer", "stopped", "id", "inFlight", "connectionPromise", "signalingUrl", "channel", "trackListeners", "trackEndedListeners", "streamListeners", "event", "track", "stream", "kind", "sender", "s", "item", "pending", "operation", "changed", "receiveAudio", "receiveVideo", "ensureReceive", "enabled", "signal", "offer", "timer", "abortListener", "response", "text", "description", "previous", "active", "transport", "timeoutMs", "_resolve", "settled", "cleanup", "onStateChange", "finish", "onOpen", "onClose", "requestedTimeout", "requestId", "notifyRemote", "onRequest", "messageEvent", "configured", "isFactory", "ownsStream", "closing", "prefix"]
 }

package/package.json CHANGED Viewed

@@ -1,6 +1,6 @@
 {
   "name": "@hyperttp/transport-webrtc",
-  "version": "0.2.1",
+  "version": "0.3.0",
   "description": "WebRTC transport for Hyperttp powered by Werift",
   "keywords": [
     "bun",
@@ -65,6 +65,6 @@
     "vitest": "^5.1.0"
   },
   "peerDependencies": {
-    "@hyperttp/types": "^0.5.3"
+    "@hyperttp/types": "^0.5.4"
   }
 }