@hyperttp/transport-webrtc 0.2.0 → 0.2.1
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/browser.js +1 -1
- package/dist/browser.js.map +2 -2
- package/dist/pcm-audio.js +1 -1
- package/dist/pcm-audio.js.map +2 -2
- package/dist/utils/helper.js +1 -1
- package/dist/utils/helper.js.map +2 -2
- package/dist/webrtc.js +1 -1
- package/dist/webrtc.js.map +2 -2
- package/package.json +3 -3
package/dist/browser.js
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import{abortError as
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var g=Object.defineProperty;var l=(u,e)=>g(u,"name",{value:e,configurable:!0});import{abortError as p,decodeWireBody as m,encodeWireBody as w}from"./utils/helper.js";class y{static{l(this,"WebRtcTransport")}options;sessions=new Map;queues=new Map;trackListeners=new Set;disposals=new Set;closed=!1;closing;sequence=0;constructor(e={}){this.options=e}onTrack(e){return this.trackListeners.add(e),()=>{this.trackListeners.delete(e)}}getSession(e){const t=this.sessions.get(e);if(!(!t||t.controller.signal.aborted))return{id:t.id,peerUrl:t.peerUrl,peerConnection:t.peerConnection,dataChannel:t.dataChannel,remoteStream:t.remoteStream,...t.localStream?{localStream:t.localStream}:{}}}async execute(e){if(this.closed)throw new Error("WebRTC transport is closed");if(e.signal?.aborted)throw p(e.signal);const t=e.peerUrl??e.url,o=(this.queues.get(t)??Promise.resolve()).catch(()=>{}).then(async()=>{if(this.closed)throw new Error("WebRTC transport is closed");if(e.signal?.aborted)throw p(e.signal);let a=this.sessions.get(t);if(a&&(a.controller.signal.aborted||a.dataChannel.readyState!=="open"||["failed","closed"].includes(a.peerConnection.connectionState))&&(await this.dispose(a),a=void 0),this.closed)throw new Error("WebRTC transport is closed");a??=this.createSession(t,e);try{const c=this.configureMedia(a,e);return(!a.signaled||c)&&await this.negotiate(a,e.signal),await this.run(a,e.signal,this.options.connectionTimeoutMs??15e3,"DataChannel open",d=>this.waitFor(a.dataChannel,["open","close","error"],d,()=>a.dataChannel.readyState==="open",()=>["closing","closed"].includes(a.dataChannel.readyState))),a}catch(c){throw this.dispose(a),c}});this.queues.set(t,o);const n=l(()=>{this.queues.get(t)===o&&this.queues.delete(t)},"remove");o.then(n,n);const s=await this.abortable(o,e.signal);return this.sendRequest(s,e)}close(){if(this.closing)return this.closing;this.closed=!0;const e=[...this.queues.values()],t=[...this.sessions.values()];return this.closing=(async()=>{await Promise.all(t.map(r=>this.dispose(r))),await Promise.allSettled(e),await Promise.all(this.disposals),this.trackListeners.clear()})(),this.closing}destroy(){return this.close()}createSession(e,t){const r=new URL(e.replace(/^webrtc:/i,"http:").replace(/^webrtcs:/i,"https:"));if(r.protocol!=="http:"&&r.protocol!=="https:")throw new Error(`Unsupported WebRTC signaling protocol: ${r.protocol}`);r.pathname==="/"&&(r.pathname="/signal"),r.search="",r.hash="";const o=(this.options.createPeerConnection??(i=>new RTCPeerConnection(i)))({iceServers:t.iceServers??this.options.iceServers??[{urls:"stun:stun.l.google.com:19302"}]});let n;try{n={id:`session-${crypto.randomUUID()}`,peerUrl:e,signalingUrl:r.toString(),peerConnection:o,dataChannel:o.createDataChannel(t.channelLabel??"hyperttp-webrtc-transport",{ordered:!0}),remoteStream:(this.options.createMediaStream??(()=>new MediaStream))(),controller:new AbortController,pending:new Map,cleanup:[],signaled:!1}}catch(i){throw o.close(),i}this.sessions.set(e,n);const s=l(i=>{n.remoteStream.getTracks().includes(i.track)||n.remoteStream.addTrack(i.track);const h={side:"client",sessionId:n.id,peerUrl:e,peerConnection:o,track:i.track,stream:n.remoteStream,streams:i.streams};for(const f of this.trackListeners)try{f(h)}catch{}},"onTrack"),a=l(()=>{(o.connectionState==="failed"||o.connectionState==="closed")&&this.dispose(n)},"onState"),c=l(()=>{this.dispose(n)},"onClose"),d=l(i=>this.receive(n,i),"onMessage");return o.addEventListener("track",s),o.addEventListener("connectionstatechange",a),n.dataChannel.addEventListener("message",d),n.dataChannel.addEventListener("close",c),n.dataChannel.addEventListener("error",c),n.cleanup.push(()=>{o.removeEventListener("track",s),o.removeEventListener("connectionstatechange",a),n.dataChannel.removeEventListener("message",d),n.dataChannel.removeEventListener("close",c),n.dataChannel.removeEventListener("error",c)}),n}configureMedia(e,t){const r=e.peerConnection;let o=!1;if(t.localStream){for(const n of t.localStream.getTracks())n.readyState==="live"&&!r.getSenders().some(s=>s.track===n)&&(r.addTrack(n,t.localStream),o=!0);e.localStream=t.localStream}for(const n of["audio","video"]){if(!(n==="audio"?t.receiveAudio:t.receiveVideo))continue;const s=r.getTransceivers().find(a=>a.receiver.track.kind===n);s?(s.direction==="sendonly"||s.direction==="inactive")&&(s.direction=s.direction==="sendonly"?"sendrecv":"recvonly",o=!0):(r.addTransceiver(n,{direction:"recvonly"}),o=!0)}return o}async negotiate(e,t){const r=e.peerConnection;await this.run(e,t,this.options.signalingTimeoutMs??1e4,"SDP offer",async o=>{const n=await r.createOffer();if(o.aborted)throw p(o);await r.setLocalDescription(n)}),await this.run(e,t,this.options.iceGatheringTimeoutMs??1e4,"ICE gathering",o=>this.waitFor(r,["icegatheringstatechange"],o,()=>r.iceGatheringState==="complete")),await this.run(e,t,this.options.signalingTimeoutMs??1e4,"Signaling",async o=>{if(!r.localDescription?.sdp)throw new Error("Local SDP offer was not generated");e.signaled=!0;const n=await fetch(e.signalingUrl,{method:"POST",headers:{"content-type":"application/json"},body:JSON.stringify({sessionId:e.id,sdp:r.localDescription}),signal:o});if(!n.ok)throw new Error(`Signaling HTTP error: ${n.status} ${n.statusText}`);const s=await n.json();if(!s||typeof s!="object"||!("type"in s)||s.type!=="answer"||!("sdp"in s)||typeof s.sdp!="string")throw new Error("Signaling server returned an invalid SDP answer");if(o.aborted)throw p(o);await r.setRemoteDescription({type:"answer",sdp:s.sdp})})}receive(e,t){if(typeof t.data!="string")return;let r;try{const n=JSON.parse(t.data);if(!n||typeof n!="object")return;r=n}catch{return}if(typeof r.requestId!="string")return;const o=e.pending.get(r.requestId);if(o)try{o(void 0,{status:typeof r.status=="number"?r.status:200,statusText:typeof r.statusText=="string"?r.statusText:"OK",headers:r.headers&&typeof r.headers=="object"?r.headers:{},body:m(r.body),url:e.peerUrl})}catch(n){o(n)}}sendRequest(e,t){const r=t.timeout&&t.timeout>0?t.timeout:this.options.requestTimeoutMs??15e3;return this.run(e,t.signal,r,"WebRTC request",o=>new Promise((n,s)=>{if(e.dataChannel.readyState!=="open"){s(new Error("WebRTC DataChannel is not open"));return}const a=`request-${++this.sequence}`,c=l((i,h)=>{e.pending.delete(a),o.removeEventListener("abort",d),h?n(h):s(i??new Error("WebRTC session closed"))},"finish"),d=l(()=>c(p(o)),"onAbort");if(e.pending.set(a,c),o.addEventListener("abort",d,{once:!0}),o.aborted){d();return}try{e.dataChannel.send(JSON.stringify({requestId:a,method:t.method,url:t.url,headers:t.headers,body:w(t.body),protocol:t.protocol}))}catch(i){c(i)}}))}dispose(e){if(e.disposal)return e.disposal;const t=Promise.resolve().then(async()=>{e.controller.abort(new Error("WebRTC session closed")),this.sessions.get(e.peerUrl)===e&&this.sessions.delete(e.peerUrl);for(const s of e.cleanup)s();e.cleanup.length=0;for(const s of e.pending.values())s(new Error("WebRTC session closed"));try{e.dataChannel.close()}catch{}try{e.peerConnection.close()}catch{}if(!e.signaled)return;const r=new URL(e.signalingUrl);r.searchParams.set("sessionId",e.id);const o=new AbortController,n=setTimeout(()=>o.abort(),2e3);try{await this.abortable(fetch(r,{method:"DELETE",signal:o.signal}),o.signal)}catch{}finally{clearTimeout(n)}});return e.disposal=t,this.disposals.add(t),t.then(()=>this.disposals.delete(t)),t}async run(e,t,r,o,n){const s=new AbortController,a=[e.controller.signal,...t?[t]:[]],c=l(()=>{const i=a.find(h=>h.aborted);i&&s.abort(p(i))},"onAbort");for(const i of a)i.addEventListener("abort",c,{once:!0});const d=setTimeout(()=>s.abort(new Error(`${o} timed out after ${r} ms`)),r);try{if(c(),s.signal.aborted)throw p(s.signal);return await this.abortable(n(s.signal),s.signal)}finally{clearTimeout(d);for(const i of a)i.removeEventListener("abort",c)}}abortable(e,t){return t?new Promise((r,o)=>{const n=l(()=>t.removeEventListener("abort",s),"cleanup"),s=l(()=>{n(),o(p(t))},"onAbort");t.addEventListener("abort",s,{once:!0}),t.aborted&&s(),e.then(a=>{n(),r(a)},a=>{n(),o(a)})}):e}waitFor(e,t,r,o,n=()=>!1){return new Promise((s,a)=>{const c=l(h=>{for(const f of t)e.removeEventListener(f,i);r.removeEventListener("abort",d),h?a(h):s()},"finish"),d=l(()=>c(p(r)),"onAbort"),i=l(h=>{r.aborted?d():n()||h?.type==="error"?c(new Error("WebRTC connection failed")):o()&&c()},"check");for(const h of t)e.addEventListener(h,i);r.addEventListener("abort",d,{once:!0}),i()})}}export{y as WebRtcTransport};
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//# sourceMappingURL=browser.js.map
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package/dist/browser.js.map
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"version": 3,
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"sources": ["../src/browser.ts"],
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"sourcesContent": ["import type { HyperTransport, TransportRequest, TransportResponse } from \"@hyperttp/types\";\nimport { abortError, decodeWireBody, encodeWireBody } from \"./utils/helper.js\";\n\nexport type WebRtcTransportRequest = TransportRequest & {\n peerUrl?: string;\n iceServers?: RTCIceServer[];\n localStream?: MediaStream;\n receiveAudio?: boolean;\n receiveVideo?: boolean;\n channelLabel?: string;\n timeout?: number;\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 createPeerConnection?: (configuration: RTCConfiguration) => RTCPeerConnection;\n readonly createMediaStream?: () => MediaStream;\n}\n\nexport interface WebRtcTrackEvent {\n readonly side: \"client\";\n readonly sessionId: string;\n readonly peerUrl: string;\n readonly peerConnection: RTCPeerConnection;\n readonly track: MediaStreamTrack;\n readonly stream: MediaStream;\n readonly streams: readonly MediaStream[];\n}\n\nexport interface WebRtcSessionView {\n readonly id: string;\n readonly peerUrl: string;\n readonly peerConnection: RTCPeerConnection;\n readonly dataChannel: RTCDataChannel;\n readonly remoteStream: MediaStream;\n readonly localStream?: MediaStream;\n}\n\ninterface Session extends WebRtcSessionView {\n localStream?: MediaStream;\n signalingUrl: string;\n controller: AbortController;\n pending: Map<string, (error?: unknown, response?: TransportResponse) => void>;\n cleanup: (() => void)[];\n signaled: boolean;\n disposal?: Promise<void>;\n}\n\n/** Browser client transport. Caller-owned local media tracks are never stopped. */\nexport class WebRtcTransport implements HyperTransport {\n private readonly sessions = new Map<string, Session>();\n private readonly queues = new Map<string, Promise<Session>>();\n private readonly trackListeners = new Set<(event: WebRtcTrackEvent) => void>();\n private readonly disposals = new Set<Promise<void>>();\n private closed = false;\n private closing: Promise<void> | undefined;\n private sequence = 0;\n\n constructor(private readonly options: WebRtcTransportOptions = {}) {}\n\n onTrack(listener: (event: WebRtcTrackEvent) => void): () => void {\n this.trackListeners.add(listener);\n return () => {\n this.trackListeners.delete(listener);\n };\n }\n\n getSession(peerUrl: string): WebRtcSessionView | undefined {\n const session = this.sessions.get(peerUrl);\n if (!session || session.controller.signal.aborted) return undefined;\n return {\n id: session.id,\n peerUrl: session.peerUrl,\n peerConnection: session.peerConnection,\n dataChannel: session.dataChannel,\n remoteStream: session.remoteStream,\n ...(session.localStream ? { localStream: session.localStream } : {}),\n };\n }\n\n async execute(request: WebRtcTransportRequest): Promise<TransportResponse> {\n if (this.closed) throw new Error(\"WebRTC transport is closed\");\n if (request.signal?.aborted) throw abortError(request.signal);\n const peerUrl = request.peerUrl ?? request.url;\n // Queue before awaiting anything, including disposal of an unusable session.\n const previous = this.queues.get(peerUrl);\n const operation = (previous ?? Promise.resolve())\n .catch(() => undefined)\n .then(async () => {\n if (this.closed) throw new Error(\"WebRTC transport is closed\");\n if (request.signal?.aborted) throw abortError(request.signal);\n let session = this.sessions.get(peerUrl);\n if (\n session &&\n (session.controller.signal.aborted ||\n session.dataChannel.readyState !== \"open\" ||\n [\"failed\", \"closed\"].includes(session.peerConnection.connectionState))\n ) {\n await this.dispose(session);\n session = undefined;\n }\n if (this.closed) throw new Error(\"WebRTC transport is closed\");\n session ??= this.createSession(peerUrl, request);\n try {\n const changed = this.configureMedia(session, request);\n if (!session.signaled || changed) await this.negotiate(session, request.signal);\n await this.run(\n session,\n request.signal,\n this.options.connectionTimeoutMs ?? 15_000,\n \"DataChannel open\",\n (signal) =>\n this.waitFor(\n session!.dataChannel,\n [\"open\", \"close\", \"error\"],\n signal,\n () => session!.dataChannel.readyState === \"open\",\n () => [\"closing\", \"closed\"].includes(session!.dataChannel.readyState),\n ),\n );\n return session;\n } catch (error) {\n void this.dispose(session);\n throw error;\n }\n });\n this.queues.set(peerUrl, operation);\n const remove = () => {\n if (this.queues.get(peerUrl) === operation) this.queues.delete(peerUrl);\n };\n void operation.then(remove, remove);\n // A queued caller can abort immediately without interrupting another caller's setup.\n const session = await this.abortable(operation, request.signal);\n return this.sendRequest(session, request);\n }\n\n close(): Promise<void> {\n if (this.closing) return this.closing;\n this.closed = true;\n const connections = [...this.queues.values()];\n const sessions = [...this.sessions.values()];\n this.closing = (async () => {\n await Promise.all(sessions.map((session) => this.dispose(session)));\n await Promise.allSettled(connections);\n await Promise.all(this.disposals);\n this.trackListeners.clear();\n })();\n return this.closing;\n }\n\n destroy(): Promise<void> {\n return this.close();\n }\n\n private createSession(peerUrl: string, request: WebRtcTransportRequest): Session {\n const url = new URL(peerUrl.replace(/^webrtc:/i, \"http:\").replace(/^webrtcs:/i, \"https:\"));\n if (url.protocol !== \"http:\" && url.protocol !== \"https:\") {\n throw new Error(`Unsupported WebRTC signaling protocol: ${url.protocol}`);\n }\n if (url.pathname === \"/\") url.pathname = \"/signal\";\n url.search = \"\";\n url.hash = \"\";\n const pc = (this.options.createPeerConnection ?? ((config) => new RTCPeerConnection(config)))({\n iceServers: request.iceServers ??\n this.options.iceServers ?? [{ urls: \"stun:stun.l.google.com:19302\" }],\n });\n let session: Session;\n try {\n session = {\n id: `session-${crypto.randomUUID()}`,\n peerUrl,\n signalingUrl: url.toString(),\n peerConnection: pc,\n dataChannel: pc.createDataChannel(request.channelLabel ?? \"hyperttp-webrtc-transport\", {\n ordered: true,\n }),\n remoteStream: (this.options.createMediaStream ?? (() => new MediaStream()))(),\n controller: new AbortController(),\n pending: new Map(),\n cleanup: [],\n signaled: false,\n };\n } catch (error) {\n pc.close();\n throw error;\n }\n this.sessions.set(peerUrl, session);\n const onTrack = (event: RTCTrackEvent) => {\n if (!session.remoteStream.getTracks().includes(event.track))\n session.remoteStream.addTrack(event.track);\n const trackEvent: WebRtcTrackEvent = {\n side: \"client\",\n sessionId: session.id,\n peerUrl,\n peerConnection: pc,\n track: event.track,\n stream: session.remoteStream,\n streams: event.streams,\n };\n for (const listener of this.trackListeners) {\n try {\n listener(trackEvent);\n } catch {\n /* Consumers must not interrupt transport events. */\n }\n }\n };\n const onState = () => {\n if (pc.connectionState === \"failed\" || pc.connectionState === \"closed\")\n void this.dispose(session);\n };\n const onClose = () => {\n void this.dispose(session);\n };\n const onMessage = (event: MessageEvent) => this.receive(session, event);\n pc.addEventListener(\"track\", onTrack);\n pc.addEventListener(\"connectionstatechange\", onState);\n session.dataChannel.addEventListener(\"message\", onMessage);\n session.dataChannel.addEventListener(\"close\", onClose);\n session.dataChannel.addEventListener(\"error\", onClose);\n session.cleanup.push(() => {\n pc.removeEventListener(\"track\", onTrack);\n pc.removeEventListener(\"connectionstatechange\", onState);\n session.dataChannel.removeEventListener(\"message\", onMessage);\n session.dataChannel.removeEventListener(\"close\", onClose);\n session.dataChannel.removeEventListener(\"error\", onClose);\n });\n return session;\n }\n\n private configureMedia(session: Session, request: WebRtcTransportRequest): boolean {\n const pc = session.peerConnection;\n let changed = false;\n if (request.localStream) {\n for (const track of request.localStream.getTracks()) {\n if (\n track.readyState === \"live\" &&\n !pc.getSenders().some((sender) => sender.track === track)\n ) {\n pc.addTrack(track, request.localStream);\n changed = true;\n }\n }\n session.localStream = request.localStream;\n }\n for (const kind of [\"audio\", \"video\"] as const) {\n if (!(kind === \"audio\" ? request.receiveAudio : request.receiveVideo)) continue;\n const transceiver = pc.getTransceivers().find((item) => item.receiver.track.kind === kind);\n if (!transceiver) {\n pc.addTransceiver(kind, { direction: \"recvonly\" });\n changed = true;\n } else if (transceiver.direction === \"sendonly\" || transceiver.direction === \"inactive\") {\n transceiver.direction = transceiver.direction === \"sendonly\" ? \"sendrecv\" : \"recvonly\";\n changed = true;\n }\n }\n return changed;\n }\n\n private async negotiate(session: Session, signal?: AbortSignal): Promise<void> {\n const pc = session.peerConnection;\n await this.run(\n session,\n signal,\n this.options.signalingTimeoutMs ?? 10_000,\n \"SDP offer\",\n async (active) => {\n const offer = await pc.createOffer();\n if (active.aborted) throw abortError(active);\n await pc.setLocalDescription(offer);\n },\n );\n await this.run(\n session,\n signal,\n this.options.iceGatheringTimeoutMs ?? 10_000,\n \"ICE gathering\",\n (active) =>\n this.waitFor(\n pc,\n [\"icegatheringstatechange\"],\n active,\n () => pc.iceGatheringState === \"complete\",\n ),\n );\n await this.run(\n session,\n signal,\n this.options.signalingTimeoutMs ?? 10_000,\n \"Signaling\",\n async (active) => {\n if (!pc.localDescription?.sdp) throw new Error(\"Local SDP offer was not generated\");\n session.signaled = true;\n const response = await fetch(session.signalingUrl, {\n method: \"POST\",\n headers: { \"content-type\": \"application/json\" },\n body: JSON.stringify({ sessionId: session.id, sdp: pc.localDescription }),\n signal: active,\n });\n if (!response.ok)\n throw new Error(`Signaling HTTP error: ${response.status} ${response.statusText}`);\n const answer: unknown = await response.json();\n if (\n !answer ||\n typeof answer !== \"object\" ||\n !(\"type\" in answer) ||\n answer.type !== \"answer\" ||\n !(\"sdp\" in answer) ||\n typeof answer.sdp !== \"string\"\n ) {\n throw new Error(\"Signaling server returned an invalid SDP answer\");\n }\n if (active.aborted) throw abortError(active);\n await pc.setRemoteDescription({ type: \"answer\", sdp: answer.sdp });\n },\n );\n }\n\n private receive(session: Session, event: MessageEvent): void {\n if (typeof event.data !== \"string\") return;\n let message: Record<string, unknown>;\n try {\n const parsed: unknown = JSON.parse(event.data);\n if (!parsed || typeof parsed !== \"object\") return;\n message = parsed as Record<string, unknown>;\n } catch {\n return;\n }\n if (typeof message.requestId !== \"string\") return;\n const finish = session.pending.get(message.requestId);\n if (!finish) return;\n try {\n finish(undefined, {\n status: typeof message.status === \"number\" ? message.status : 200,\n statusText: typeof message.statusText === \"string\" ? message.statusText : \"OK\",\n headers:\n message.headers && typeof message.headers === \"object\"\n ? (message.headers as TransportResponse[\"headers\"])\n : {},\n body: decodeWireBody(message.body),\n url: session.peerUrl,\n });\n } catch (error) {\n finish(error);\n }\n }\n\n private sendRequest(\n session: Session,\n request: WebRtcTransportRequest,\n ): Promise<TransportResponse> {\n const timeout =\n request.timeout && request.timeout > 0\n ? request.timeout\n : (this.options.requestTimeoutMs ?? 15_000);\n return this.run(\n session,\n request.signal,\n timeout,\n \"WebRTC request\",\n (signal) =>\n new Promise<TransportResponse>((resolve, reject) => {\n if (session.dataChannel.readyState !== \"open\") {\n reject(new Error(\"WebRTC DataChannel is not open\"));\n return;\n }\n const requestId = `request-${++this.sequence}`;\n const finish = (error?: unknown, response?: TransportResponse) => {\n session.pending.delete(requestId);\n signal.removeEventListener(\"abort\", onAbort);\n if (response) resolve(response);\n else reject(error ?? new Error(\"WebRTC session closed\"));\n };\n const onAbort = () => finish(abortError(signal));\n session.pending.set(requestId, finish);\n signal.addEventListener(\"abort\", onAbort, { once: true });\n if (signal.aborted) {\n onAbort();\n return;\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 finish(error);\n }\n }),\n );\n }\n\n private dispose(session: Session): Promise<void> {\n if (session.disposal) return session.disposal;\n const disposal = Promise.resolve().then(async () => {\n session.controller.abort(new Error(\"WebRTC session closed\"));\n if (this.sessions.get(session.peerUrl) === session) this.sessions.delete(session.peerUrl);\n for (const cleanup of session.cleanup) cleanup();\n session.cleanup.length = 0;\n for (const finish of session.pending.values()) finish(new Error(\"WebRTC session closed\"));\n try {\n session.dataChannel.close();\n } catch {}\n try {\n session.peerConnection.close();\n } catch {}\n if (!session.signaled) return;\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 this.abortable(\n fetch(url, { method: \"DELETE\", signal: controller.signal }),\n controller.signal,\n );\n } catch {\n /* Remote cleanup is best effort. */\n } finally {\n clearTimeout(timer);\n }\n });\n session.disposal = disposal;\n this.disposals.add(disposal);\n void disposal.then(() => this.disposals.delete(disposal));\n return disposal;\n }\n\n private async run<T>(\n session: Session,\n signal: AbortSignal | undefined,\n timeout: number,\n label: string,\n task: (signal: AbortSignal) => Promise<T>,\n ): Promise<T> {\n const controller = new AbortController();\n const sources = [session.controller.signal, ...(signal ? [signal] : [])];\n const onAbort = () => {\n const source = sources.find((item) => item.aborted);\n if (source) controller.abort(abortError(source));\n };\n for (const source of sources) source.addEventListener(\"abort\", onAbort, { once: true });\n const timer = setTimeout(\n () => controller.abort(new Error(`${label} timed out after ${timeout} ms`)),\n timeout,\n );\n try {\n onAbort();\n if (controller.signal.aborted) throw abortError(controller.signal);\n return await this.abortable(task(controller.signal), controller.signal);\n } finally {\n clearTimeout(timer);\n for (const source of sources) source.removeEventListener(\"abort\", onAbort);\n }\n }\n\n private abortable<T>(promise: Promise<T>, signal?: AbortSignal): Promise<T> {\n if (!signal) return promise;\n return new Promise<T>((resolve, reject) => {\n const cleanup = () => signal.removeEventListener(\"abort\", onAbort);\n const onAbort = () => {\n cleanup();\n reject(abortError(signal));\n };\n signal.addEventListener(\"abort\", onAbort, { once: true });\n if (signal.aborted) onAbort();\n void promise.then(\n (value) => {\n cleanup();\n resolve(value);\n },\n (error: unknown) => {\n cleanup();\n reject(error);\n },\n );\n });\n }\n\n private waitFor(\n target: EventTarget,\n events: string[],\n signal: AbortSignal,\n ready: () => boolean,\n failed: () => boolean = () => false,\n ): Promise<void> {\n return new Promise<void>((resolve, reject) => {\n const finish = (error?: Error) => {\n for (const event of events) target.removeEventListener(event, check);\n signal.removeEventListener(\"abort\", onAbort);\n if (error) reject(error);\n else resolve();\n };\n const onAbort = () => finish(abortError(signal));\n const check = (event?: Event) => {\n if (signal.aborted) onAbort();\n else if (failed() || event?.type === \"error\") finish(new Error(\"WebRTC connection failed\"));\n else if (ready()) finish();\n };\n for (const event of events) target.addEventListener(event, check);\n signal.addEventListener(\"abort\", onAbort, { once: true });\n check();\n });\n }\n}\n"],
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5
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"mappings": "
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"names": ["abortError", "decodeWireBody", "encodeWireBody", "WebRtcTransport", "options", "listener", "peerUrl", "session", "request", "operation", "changed", "signal", "error", "remove", "connections", "sessions", "url", "pc", "config", "onTrack", "event", "trackEvent", "onState", "onClose", "onMessage", "track", "sender", "kind", "transceiver", "item", "active", "offer", "response", "answer", "message", "parsed", "finish", "timeout", "resolve", "reject", "requestId", "onAbort", "disposal", "cleanup", "controller", "timer", "label", "task", "sources", "source", "promise", "value", "target", "events", "ready", "failed", "check"]
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5
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|
|
6
|
+
"names": ["abortError", "decodeWireBody", "encodeWireBody", "WebRtcTransport", "__name", "options", "listener", "peerUrl", "session", "request", "operation", "changed", "signal", "error", "remove", "connections", "sessions", "url", "pc", "config", "onTrack", "event", "trackEvent", "onState", "onClose", "onMessage", "track", "sender", "kind", "transceiver", "item", "active", "offer", "response", "answer", "message", "parsed", "finish", "timeout", "resolve", "reject", "requestId", "onAbort", "disposal", "cleanup", "controller", "timer", "label", "task", "sources", "source", "promise", "value", "target", "events", "ready", "failed", "check"]
|
|
7
7
|
}
|
package/dist/pcm-audio.js
CHANGED
|
@@ -1,2 +1,2 @@
|
|
|
1
|
-
import{spawn as
|
|
1
|
+
var I=Object.defineProperty;var n=(u,c)=>I(u,"name",{value:c,configurable:!0});import{spawn as x}from"node:child_process";import{createSocket as A}from"node:dgram";import{MediaStream as R,MediaStreamTrack as $,RtpPacket as z}from"werift";async function K(u,c={}){c.signal?.throwIfAborted();const l=A("udp4");try{await new Promise((e,r)=>{l.once("error",r),l.bind(0,"127.0.0.1",()=>{l.removeListener("error",r),e()})}),c.signal?.throwIfAborted()}catch(e){try{l.close()}catch{}throw e}const k=l.address().port,o=x(c.ffmpegPath??"ffmpeg",["-hide_banner","-loglevel","error","-nostdin","-re","-f","s16le","-ar","48000","-ac","2","-i","pipe:0","-c:a","libopus","-frame_duration","20","-ar","48000","-ac","2","-payload_type","111","-f","rtp",`rtp://127.0.0.1:${k}?rtcpport=${k}&pkt_size=1200`],{stdio:["pipe","ignore","pipe"]}),p=new Set;let i=!1,P=!1,h="",y,S,v;const b=new Promise((e,r)=>{S=e,v=r});b.catch(()=>{});const f=new Set;function E(e){return new Promise((r,a)=>{const t=n(()=>r(void 0),"cancel");f.add(t),Promise.resolve(e).then(d=>{f.delete(t),r(d)},d=>{f.delete(t),a(d)}),i&&(f.delete(t),t())})}n(E,"interruptible");const L=new Promise(e=>o.once("close",()=>e()));let w;function s(e){if(w)return w;i=!0;for(const r of f)r();f.clear(),c.signal?.removeEventListener("abort",g);try{Promise.resolve(y?.()).catch(()=>{})}catch{}for(const r of p)r.stop();return p.clear(),o.stdin.destroy(),o.exitCode===null&&o.signalCode===null&&o.kill("SIGTERM"),w=(async()=>{const r=setTimeout(()=>{o.kill("SIGKILL")},1e3);try{await Promise.all([L,new Promise(a=>l.close(()=>a()))])}finally{clearTimeout(r)}})(),w.then(()=>e===void 0?S():v(e),v),w}n(s,"finish");function g(){s(c.signal?.reason??new Error("PCM audio source aborted"))}n(g,"onAbort"),l.on("error",e=>{s(e)}),o.stdin.on("error",e=>{i||s(e)}),o.stderr.on("data",e=>{h=(h+e.toString()).slice(-8192)}),o.on("error",e=>{s(e)}),o.on("close",(e,r)=>{i||(e===0&&P?s():s(new Error(`FFmpeg exited (${r??e}): ${h.trim()}`)))}),l.on("message",(e,r)=>{if(i||r.address!=="127.0.0.1"||e.length<12||e[0]>>>6!==2||(e[1]&127)!==111)return;let a;try{a=z.deSerialize(e)}catch{return}if(a.header.payloadType===111)for(const t of p)t.stopped||t.writeRtp(a)}),c.signal?.addEventListener("abort",g,{once:!0}),c.signal?.aborted&&g();try{if(await new Promise((e,r)=>{o.once("spawn",e),o.once("error",r)}),i)throw await b,new Error("PCM audio source closed during startup")}catch(e){throw await s(e),e}async function M(){let e;try{let r;if("getReader"in u){const t=u.getReader();r={next:n(()=>t.read(),"next")},y=n(()=>t.cancel(),"cancelInput"),e=n(()=>t.releaseLock(),"release")}else r=u[Symbol.asyncIterator](),y=n(()=>r.return?.(),"cancelInput");let a=0;for(;!i;){const t=await E(r.next());if(!t||i)return;if(t.done){if(a%4!==0)throw new Error("PCM input ends with an incomplete stereo s16le sample");P=!0,o.stdin.end();return}if(!(t.value instanceof Uint8Array))throw new TypeError("PCM chunks must be Uint8Array");const d=t.value;a=(a+d.byteLength)%4;for(let m=0;m<d.byteLength&&!i;m+=16384)await E(new Promise((T,_)=>{o.stdin.write(d.subarray(m,m+16384),C=>C?_(C):T())}))}}catch(r){i||s(r)}finally{try{e?.()}catch{}}}return n(M,"pump"),M(),{done:b,subscribe(){if(i)throw new Error("PCM audio source is closed");const e=new $({kind:"audio"}),r=new R;return r.addTrack(e),p.add(e),{stream:r,close(){p.delete(e),e.stop()}}},close:n(()=>s(),"close")}}n(K,"createPcmAudioSource");export{K as createPcmAudioSource};
|
|
2
2
|
//# sourceMappingURL=pcm-audio.js.map
|
package/dist/pcm-audio.js.map
CHANGED
|
@@ -2,6 +2,6 @@
|
|
|
2
2
|
"version": 3,
|
|
3
3
|
"sources": ["../src/pcm-audio.ts"],
|
|
4
4
|
"sourcesContent": ["import { spawn } from \"node:child_process\";\nimport { createSocket } from \"node:dgram\";\nimport { MediaStream, MediaStreamTrack, RtpPacket } from \"werift\";\n\nexport interface PcmAudioSubscription {\n stream: MediaStream;\n close(): void;\n}\n\nexport interface PcmAudioSource {\n subscribe(): PcmAudioSubscription;\n close(): Promise<void>;\n /** Resolves on EOF or explicit close; rejects on input, socket, or encoder failure. */\n done: Promise<void>;\n}\n\nexport interface PcmAudioOptions {\n ffmpegPath?: string;\n signal?: AbortSignal;\n}\n\n/**\n * Optional server-only helper. Requires an external FFmpeg executable with libopus\n * on PATH (or ffmpegPath). Input is raw 48 kHz, stereo, signed 16-bit little-endian\n * PCM; chunk boundaries need not align to samples. One input/encoder is shared,\n * paced by FFmpeg's -re, with no replay for late subscribers. RTP/RTCP stays on\n * IPv4 loopback. No peer connection or native wrtc dependency is created.\n *\n * Writes are bounded to 16 KiB, with one outstanding write and one input chunk.\n * Producers must bound their own chunks/queues. Closing requests iterator.return()\n * or reader.cancel(); arbitrary iterators must cooperate to release their resources\n * (a stuck next()/return() does not prevent this helper from closing).\n */\nexport async function createPcmAudioSource(\n input: AsyncIterable<Uint8Array> | ReadableStream<Uint8Array>,\n options: PcmAudioOptions = {},\n): Promise<PcmAudioSource> {\n options.signal?.throwIfAborted();\n const socket = createSocket(\"udp4\");\n try {\n await new Promise<void>((resolve, reject) => {\n socket.once(\"error\", reject);\n socket.bind(0, \"127.0.0.1\", () => {\n socket.removeListener(\"error\", reject);\n resolve();\n });\n });\n options.signal?.throwIfAborted();\n } catch (error) {\n try {\n socket.close();\n } catch {\n /* Bind may not have succeeded. */\n }\n throw error;\n }\n\n const port = socket.address().port;\n const child = spawn(\n options.ffmpegPath ?? \"ffmpeg\",\n [\n \"-hide_banner\",\n \"-loglevel\",\n \"error\",\n \"-nostdin\",\n \"-re\",\n \"-f\",\n \"s16le\",\n \"-ar\",\n \"48000\",\n \"-ac\",\n \"2\",\n \"-i\",\n \"pipe:0\",\n \"-c:a\",\n \"libopus\",\n \"-frame_duration\",\n \"20\",\n \"-ar\",\n \"48000\",\n \"-ac\",\n \"2\",\n \"-payload_type\",\n \"111\",\n \"-f\",\n \"rtp\",\n `rtp://127.0.0.1:${port}?rtcpport=${port}&pkt_size=1200`,\n ],\n { stdio: [\"pipe\", \"ignore\", \"pipe\"] },\n );\n\n const tracks = new Set<MediaStreamTrack>();\n let stopped = false;\n let inputEnded = false;\n let stderr = \"\";\n let cancelInput: (() => unknown) | undefined;\n let resolveDone!: () => void;\n let rejectDone!: (error: unknown) => void;\n const done = new Promise<void>((resolve, reject) => {\n resolveDone = resolve;\n rejectDone = reject;\n });\n // Callers may observe done later, or only call close().\n void done.catch(() => {});\n const pending = new Set<() => void>();\n function interruptible<T>(operation: PromiseLike<T>): Promise<T | undefined> {\n return new Promise((resolve, reject) => {\n const cancel = () => resolve(undefined);\n pending.add(cancel);\n void Promise.resolve(operation).then(\n (value) => {\n pending.delete(cancel);\n resolve(value);\n },\n (error) => {\n pending.delete(cancel);\n reject(error);\n },\n );\n if (stopped) {\n pending.delete(cancel);\n cancel();\n }\n });\n }\n const childClosed = new Promise<void>((resolve) => child.once(\"close\", () => resolve()));\n let cleanup: Promise<void> | undefined;\n\n function finish(error?: unknown): Promise<void> {\n if (cleanup) return cleanup;\n stopped = true;\n for (const cancel of pending) cancel();\n pending.clear();\n options.signal?.removeEventListener(\"abort\", onAbort);\n // Do not wait for a user iterator that ignores cancellation.\n try {\n void Promise.resolve(cancelInput?.()).catch(() => {});\n } catch {\n /* Best effort. */\n }\n for (const track of tracks) track.stop();\n tracks.clear();\n child.stdin.destroy();\n if (child.exitCode === null && child.signalCode === null) child.kill(\"SIGTERM\");\n cleanup = (async () => {\n const killTimer = setTimeout(() => {\n child.kill(\"SIGKILL\");\n }, 1_000);\n try {\n await Promise.all([\n childClosed,\n new Promise<void>((resolve) => socket.close(() => resolve())),\n ]);\n } finally {\n clearTimeout(killTimer);\n }\n })();\n void cleanup.then(() => (error === undefined ? resolveDone() : rejectDone(error)), rejectDone);\n return cleanup;\n }\n\n function onAbort() {\n void finish(options.signal?.reason ?? new Error(\"PCM audio source aborted\"));\n }\n socket.on(\"error\", (error) => {\n void finish(error);\n });\n child.stdin.on(\"error\", (error) => {\n if (!stopped) void finish(error);\n });\n child.stderr.on(\"data\", (chunk: Buffer) => {\n stderr = (stderr + chunk.toString()).slice(-8_192);\n });\n child.on(\"error\", (error) => {\n void finish(error);\n });\n child.on(\"close\", (code, signal) => {\n if (stopped) return;\n if (code === 0 && inputEnded) void finish();\n else void finish(new Error(`FFmpeg exited (${signal ?? code}): ${stderr.trim()}`));\n });\n socket.on(\"message\", (message, remote) => {\n if (stopped || remote.address !== \"127.0.0.1\" || message.length < 12) return;\n // Reject RTCP and malformed/non-RTP datagrams before parsing.\n if (message[0]! >>> 6 !== 2 || (message[1]! & 0x7f) !== 111) return;\n let packet: RtpPacket;\n try {\n packet = RtpPacket.deSerialize(message);\n } catch {\n return;\n }\n if (packet.header.payloadType !== 111) return;\n for (const track of tracks) {\n if (!track.stopped) track.writeRtp(packet);\n }\n });\n options.signal?.addEventListener(\"abort\", onAbort, { once: true });\n if (options.signal?.aborted) onAbort();\n\n // Surface executable startup failures from the factory, not just done.\n try {\n await new Promise<void>((resolve, reject) => {\n child.once(\"spawn\", resolve);\n child.once(\"error\", reject);\n });\n if (stopped) {\n await done;\n throw new Error(\"PCM audio source closed during startup\");\n }\n } catch (error) {\n await finish(error);\n throw error;\n }\n\n async function pump() {\n let release: (() => void) | undefined;\n try {\n let iterator: AsyncIterator<Uint8Array>;\n if (\"getReader\" in input) {\n const reader = input.getReader();\n iterator = { next: () => reader.read() };\n cancelInput = () => reader.cancel();\n release = () => reader.releaseLock();\n } else {\n iterator = input[Symbol.asyncIterator]();\n cancelInput = () => iterator.return?.();\n }\n let bytes = 0;\n while (!stopped) {\n const result = await interruptible(iterator.next());\n if (!result || stopped) return;\n if (result.done) {\n if (bytes % 4 !== 0)\n throw new Error(\"PCM input ends with an incomplete stereo s16le sample\");\n inputEnded = true;\n child.stdin.end();\n return;\n }\n if (!(result.value instanceof Uint8Array))\n throw new TypeError(\"PCM chunks must be Uint8Array\");\n const chunk = result.value;\n bytes = (bytes + chunk.byteLength) % 4;\n for (let offset = 0; offset < chunk.byteLength && !stopped; offset += 16_384) {\n await interruptible(\n new Promise<void>((resolve, reject) => {\n child.stdin.write(chunk.subarray(offset, offset + 16_384), (error) =>\n error ? reject(error) : resolve(),\n );\n }),\n );\n }\n }\n } catch (error) {\n if (!stopped) void finish(error);\n } finally {\n // A cancelled read may still be pending on a non-cooperative stream.\n try {\n release?.();\n } catch {\n /* Cancellation owns the pending read. */\n }\n }\n }\n void pump();\n\n return {\n done,\n subscribe() {\n if (stopped) throw new Error(\"PCM audio source is closed\");\n const track = new MediaStreamTrack({ kind: \"audio\" });\n const stream = new MediaStream();\n stream.addTrack(track);\n tracks.add(track);\n return {\n stream,\n close() {\n tracks.delete(track);\n track.stop();\n },\n };\n },\n close: () => finish(),\n };\n}\n"],
|
|
5
|
-
"mappings": "
|
|
6
|
-
"names": ["spawn", "createSocket", "MediaStream", "MediaStreamTrack", "RtpPacket", "createPcmAudioSource", "input", "options", "socket", "resolve", "reject", "error", "port", "child", "tracks", "stopped", "inputEnded", "stderr", "cancelInput", "resolveDone", "rejectDone", "done", "pending", "interruptible", "operation", "cancel", "value", "childClosed", "cleanup", "finish", "onAbort", "track", "killTimer", "chunk", "code", "signal", "message", "remote", "packet", "pump", "release", "iterator", "reader", "bytes", "result", "offset", "stream"]
|
|
5
|
+
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|
|
6
|
+
"names": ["spawn", "createSocket", "MediaStream", "MediaStreamTrack", "RtpPacket", "createPcmAudioSource", "input", "options", "socket", "resolve", "reject", "error", "port", "child", "tracks", "stopped", "inputEnded", "stderr", "cancelInput", "resolveDone", "rejectDone", "done", "pending", "interruptible", "operation", "cancel", "__name", "value", "childClosed", "cleanup", "finish", "onAbort", "track", "killTimer", "chunk", "code", "signal", "message", "remote", "packet", "pump", "release", "iterator", "reader", "bytes", "result", "offset", "stream"]
|
|
7
7
|
}
|
package/dist/utils/helper.js
CHANGED
|
@@ -1,2 +1,2 @@
|
|
|
1
|
-
import{WEBRTC_BYTES_MARKER as
|
|
1
|
+
var f=Object.defineProperty;var t=(e,r)=>f(e,"name",{value:r,configurable:!0});import{WEBRTC_BYTES_MARKER as i}from"../types/types.js";function a(e){let r="";for(let n=0;n<e.length;n+=32768)r+=String.fromCharCode(...e.subarray(n,n+32768));return btoa(r)}t(a,"bytesToBase64");function c(e){const r=atob(e),o=new Uint8Array(r.length);for(let n=0;n<r.length;n++)o[n]=r.charCodeAt(n);return o}t(c,"base64ToBytes");function h(e){let r;return e instanceof Uint8Array?r=e:e instanceof ArrayBuffer?r=new Uint8Array(e):ArrayBuffer.isView(e)&&(r=new Uint8Array(e.buffer,e.byteOffset,e.byteLength)),r?{[i]:!0,base64:a(r)}:e}t(h,"encodeWireBody");function l(e){if(typeof e!="object"||e===null||!(i in e))return e;const r=e;return r[i]!==!0||typeof r.base64!="string"?e:c(r.base64)}t(l,"decodeWireBody");function p(e,r){return e instanceof Error?e:new Error(r)}t(p,"errorFromUnknown");function x(e){return e?.reason instanceof Error?e.reason:new Error("WebRTC operation aborted")}t(x,"abortError");function A(e){return e?(e.match(/^a=candidate:/gm)??[]).length:0}t(A,"countIceCandidates");export{x as abortError,c as base64ToBytes,a as bytesToBase64,A as countIceCandidates,l as decodeWireBody,h as encodeWireBody,p as errorFromUnknown};
|
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2
2
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//# sourceMappingURL=helper.js.map
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package/dist/utils/helper.js.map
CHANGED
|
@@ -2,6 +2,6 @@
|
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|
2
2
|
"version": 3,
|
|
3
3
|
"sources": ["../../src/utils/helper.ts"],
|
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4
4
|
"sourcesContent": ["import { WEBRTC_BYTES_MARKER, type EncodedWebRtcBytes } from \"../types/types.js\";\n\nexport function bytesToBase64(bytes: Uint8Array): string {\n let binary = \"\";\n const chunkSize = 0x8000;\n\n for (let offset = 0; offset < bytes.length; offset += chunkSize) {\n binary += String.fromCharCode(...bytes.subarray(offset, offset + chunkSize));\n }\n\n return btoa(binary);\n}\n\nexport function base64ToBytes(base64: string): Uint8Array {\n const binary = atob(base64);\n const bytes = new Uint8Array(binary.length);\n\n for (let i = 0; i < binary.length; i++) bytes[i] = binary.charCodeAt(i);\n return bytes;\n}\n\nexport function encodeWireBody(body: unknown): unknown {\n let bytes: Uint8Array | undefined;\n\n if (body instanceof Uint8Array) {\n bytes = body;\n } else if (body instanceof ArrayBuffer) {\n bytes = new Uint8Array(body);\n } else if (ArrayBuffer.isView(body)) {\n bytes = new Uint8Array(body.buffer, body.byteOffset, body.byteLength);\n }\n\n if (!bytes) return body;\n\n const encoded: EncodedWebRtcBytes = {\n [WEBRTC_BYTES_MARKER]: true,\n base64: bytesToBase64(bytes),\n };\n\n return encoded;\n}\n\nexport function decodeWireBody(body: unknown): unknown {\n if (typeof body !== \"object\" || body === null || !(WEBRTC_BYTES_MARKER in body)) {\n return body;\n }\n\n const encoded = body as Partial<EncodedWebRtcBytes>;\n if (encoded[WEBRTC_BYTES_MARKER] !== true || typeof encoded.base64 !== \"string\") {\n return body;\n }\n\n return base64ToBytes(encoded.base64);\n}\n\nexport function errorFromUnknown(error: unknown, fallback: string): Error {\n return error instanceof Error ? error : new Error(fallback);\n}\n\nexport function abortError(signal?: AbortSignal): Error {\n if (signal?.reason instanceof Error) return signal.reason;\n return new Error(\"WebRTC operation aborted\");\n}\n\nexport function countIceCandidates(sdp: string | undefined): number {\n return sdp ? (sdp.match(/^a=candidate:/gm) ?? []).length : 0;\n}\n"],
|
|
5
|
-
"mappings": "
|
|
6
|
-
"names": ["WEBRTC_BYTES_MARKER", "bytesToBase64", "bytes", "binary", "offset", "base64ToBytes", "base64", "i", "encodeWireBody", "body", "decodeWireBody", "encoded", "errorFromUnknown", "error", "fallback", "abortError", "signal", "countIceCandidates", "sdp"]
|
|
5
|
+
"mappings": "+EAAA,OAAS,uBAAAA,MAAoD,oBAEvD,SAAUC,EAAcC,EAAiB,CAC7C,IAAIC,EAAS,GAGb,QAASC,EAAS,EAAGA,EAASF,EAAM,OAAQE,GAAU,MACpDD,GAAU,OAAO,aAAa,GAAGD,EAAM,SAASE,EAAQA,EAAS,KAAS,CAAC,EAG7E,OAAO,KAAKD,CAAM,CACpB,CATgBE,EAAAJ,EAAA,iBAWV,SAAUK,EAAcC,EAAc,CAC1C,MAAMJ,EAAS,KAAKI,CAAM,EACpBL,EAAQ,IAAI,WAAWC,EAAO,MAAM,EAE1C,QAASK,EAAI,EAAGA,EAAIL,EAAO,OAAQK,IAAKN,EAAMM,CAAC,EAAIL,EAAO,WAAWK,CAAC,EACtE,OAAON,CACT,CANgBG,EAAAC,EAAA,iBAQV,SAAUG,EAAeC,EAAa,CAC1C,IAAIR,EAUJ,OARIQ,aAAgB,WAClBR,EAAQQ,EACCA,aAAgB,YACzBR,EAAQ,IAAI,WAAWQ,CAAI,EAClB,YAAY,OAAOA,CAAI,IAChCR,EAAQ,IAAI,WAAWQ,EAAK,OAAQA,EAAK,WAAYA,EAAK,UAAU,GAGjER,EAE+B,CAClC,CAACF,CAAmB,EAAG,GACvB,OAAQC,EAAcC,CAAK,GAJVQ,CAQrB,CAnBgBL,EAAAI,EAAA,kBAqBV,SAAUE,EAAeD,EAAa,CAC1C,GAAI,OAAOA,GAAS,UAAYA,IAAS,MAAQ,EAAEV,KAAuBU,GACxE,OAAOA,EAGT,MAAME,EAAUF,EAChB,OAAIE,EAAQZ,CAAmB,IAAM,IAAQ,OAAOY,EAAQ,QAAW,SAC9DF,EAGFJ,EAAcM,EAAQ,MAAM,CACrC,CAXgBP,EAAAM,EAAA,kBAaV,SAAUE,EAAiBC,EAAgBC,EAAgB,CAC/D,OAAOD,aAAiB,MAAQA,EAAQ,IAAI,MAAMC,CAAQ,CAC5D,CAFgBV,EAAAQ,EAAA,oBAIV,SAAUG,EAAWC,EAAoB,CAC7C,OAAIA,GAAQ,kBAAkB,MAAcA,EAAO,OAC5C,IAAI,MAAM,0BAA0B,CAC7C,CAHgBZ,EAAAW,EAAA,cAKV,SAAUE,EAAmBC,EAAuB,CACxD,OAAOA,GAAOA,EAAI,MAAM,iBAAiB,GAAK,CAAA,GAAI,OAAS,CAC7D,CAFgBd,EAAAa,EAAA",
|
|
6
|
+
"names": ["WEBRTC_BYTES_MARKER", "bytesToBase64", "bytes", "binary", "offset", "__name", "base64ToBytes", "base64", "i", "encodeWireBody", "body", "decodeWireBody", "encoded", "errorFromUnknown", "error", "fallback", "abortError", "signal", "countIceCandidates", "sdp"]
|
|
7
7
|
}
|
package/dist/webrtc.js
CHANGED
|
@@ -1,2 +1,2 @@
|
|
|
1
|
-
import{MAX_SIGNALING_BODY_BYTES as W}from"./types/types.js";import{abortError as y,decodeWireBody as P,encodeWireBody as k,errorFromUnknown as b}from"./utils/helper.js";import{createServer as M}from"node:http";import{MediaStream as D,RTCPeerConnection as A}from"werift";function O(v){return(v??[]).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}:{}}))})}function x(v){const e={iceServers:O(v.iceServers),...v.iceTransportPolicy!==void 0?{iceTransportPolicy:v.iceTransportPolicy}:{}};return new A(e)}function $(){return new D}class F{sessions=new Map;connecting=new Map;serverSessions=new Map;trackListeners=new Set;serverClosers=new Set;requestSequence=0;closing=!1;pendingPeers=new Set;iceServers;iceGatheringTimeoutMs;signalingTimeoutMs;connectionTimeoutMs;requestTimeoutMs;createMediaStream;createPeerConnection;options;constructor(e={}){this.options=e,this.createPeerConnection=e.createPeerConnection??x,this.createMediaStream=e.createMediaStream??$,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 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,d=M((a,l)=>{(async()=>{const m=(h,S)=>{l.headersSent||l.destroyed||(l.writeHead(h,{"content-type":"application/json"}),l.end(JSON.stringify(S)))};if(s||this.closing){m(503,{error:"WebRTC server is closing"});return}let w;try{w=new URL(a.url??"/","http://localhost")}catch{m(400,{error:"Invalid request URL"});return}if(w.pathname!==n){m(404,{error:"Signaling endpoint not found"});return}if((a.method==="POST"||a.method==="DELETE")&&e.signaling?.authorize){const h=new AbortController,S=()=>h.abort(),E=()=>{m(503,{error:"WebRTC signaling request aborted"})};a.once("aborted",S),l.once("close",S),h.signal.addEventListener("abort",E,{once:!0}),c.add(h);try{const T={};for(const[I,L]of Object.entries(a.headers))L!==void 0&&(T[I]=L);const R=await e.signaling.authorize({protocol:"webrtc",method:a.method,url:a.url??"/",headers:T,signal:h.signal});if(h.signal.aborted||a.aborted||l.destroyed)return;if(s||this.closing||e.signal?.aborted){m(503,{error:"WebRTC server is closing"});return}if(!R){m(403,{error:"Signaling forbidden"});return}}catch{m(403,{error:"Signaling forbidden"});return}finally{c.delete(h),a.off("aborted",S),l.off("close",S),h.signal.removeEventListener("abort",E)}}if(a.method==="DELETE"){const h=w.searchParams.get("sessionId");h&&t.has(h)&&(await this.closeServerSession(h),t.delete(h)),l.writeHead(204).end();return}if(a.method!=="POST"){m(405,{error:"Method not allowed"});return}let C="";try{for await(const h of a)if(C+=Buffer.isBuffer(h)?h.toString("utf8"):String(h),Buffer.byteLength(C,"utf8")>W){m(413,{error:"Signaling payload too large"});return}}catch(h){m(400,{error:b(h,"Failed to read body").message});return}let p,u;try{const h=JSON.parse(C);if(typeof h.sessionId!="string"||h.sessionId.length<1||h.sessionId.length>200||h.sdp?.type!=="offer"||typeof h.sdp.sdp!="string"){m(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 S=h.sessionId;if(u=this.serverSessions.get(S),u&&!t.has(S)){m(409,{error:"Session belongs to another listener"});return}if(u?.negotiating){m(409,{error:"Session negotiation already in progress"});return}!u||u.closed?(p=this.createPeerConnection({iceServers:this.iceServers}),u={id:S,peerConnection:p,remoteStream:this.createMediaStream(),localTrackIds:new Set,localStreamResolved:!1,ownsLocalStream:!1,closed:!1},t.add(S),this.serverSessions.set(S,u),this.attachServerPeerEvents(u,e.onRequest)):p=u.peerConnection,u.negotiating=!0,await this.setRemoteDescription(p,h.sdp),await this.attachConfiguredServerStream(u);const E=await p.createAnswer();await this.withDeadline(p.setLocalDescription(E),this.iceGatheringTimeoutMs),await this.waitForIceGatheringComplete(p,this.iceGatheringTimeoutMs);const T=p.localDescription;if(!T?.sdp||T.type!=="answer")throw new Error("Local SDP answer was not generated");if(s||u.closed)throw new Error("WebRTC session closed");u.negotiating=!1,m(200,T)}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),u)await this.closeServerSession(u.id);else if(p)try{await p.close()}catch{}m(400,{error:b(h,"Signaling failed").message})}})()});await new Promise((a,l)=>{const m=C=>{d.off("listening",w),l(C)},w=()=>{d.off("error",m),a()};d.once("error",m),d.once("listening",w),d.listen(r,o)}),this.log("info",`[WebRtcTransport] Listening on http://${o}:${r}${n}`);const g=()=>{f().catch(a=>{this.log("error","[WebRtcTransport] Server shutdown failed:",a)})},f=()=>{if(i)return i;s=!0;for(const a of c)a.abort();return e.signal?.removeEventListener("abort",g),i=(async()=>{const a=d.listening?new Promise((l,m)=>{d.close(w=>w?m(w):l())}):Promise.resolve();await Promise.all([...t].map(l=>this.closeServerSession(l))),t.clear(),await a,this.serverClosers.delete(f)})(),i};return e.signal?.addEventListener("abort",g,{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,g=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 g)l({stream:i.remoteStream});a.addEventListener("ended",()=>{for(const l of d)l({track:a})})}),i.mediaController={localStream:i.localStream,remoteStream:i.remoteStream,addTrack:(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)}),replaceTrack:async(f,a)=>{const l=n.getSenders().find(m=>m.track&&m.track.kind===f);l&&await l.replaceTrack(a)},removeTrack: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))}),close:async()=>{await this.disposeClientSession(i,!0)},on(f,a){return f==="track"&&c.add(a),f==="trackEnded"&&d.add(a),f==="stream"&&g.add(a),()=>{c.delete(a),d.delete(a),g.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=P(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(b(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=(i,c)=>{if(!c)return;o.getTransceivers().find(g=>g.receiver.track.kind===i&&(g.direction==="recvonly"||g.direction==="sendrecv"))||(o.addTransceiver(i,{direction:"recvonly"}),n=!0)};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=()=>s.abort(t?.reason);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 g=await d.json();if(g.type!=="answer"||typeof g.sdp!="string")throw new Error("Signaling server returned an invalid SDP answer");await this.setRemoteDescription(r,g)}catch(d){throw t?.aborted?y(t):s.signal.aborted?new Error(`Signaling timed out after ${this.signalingTimeoutMs} ms`):b(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=()=>i(y(r)),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=()=>{clearTimeout(s),e.removeEventListener("icegatheringstatechange",d)},c=g=>{n||(n=!0,i(),o(g))},d=()=>{e.iceGatheringState==="complete"&&c(!0)};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 y(r);await new Promise((o,n)=>{let s=!1;const i=()=>{clearTimeout(l),e.removeEventListener("open",d),e.removeEventListener("error",g),e.removeEventListener("close",f),r?.removeEventListener("abort",a)},c=m=>{s||(s=!0,i(),m?n(m):o())},d=()=>c(),g=()=>c(new Error("WebRTC DataChannel failed to open")),f=()=>c(new Error("WebRTC DataChannel closed before opening")),a=()=>c(y(r)),l=setTimeout(()=>c(new Error(`DataChannel open timeout after ${t} ms`)),t);e.addEventListener("open",d),e.addEventListener("error",g),e.addEventListener("close",f),r?.addEventListener("abort",a,{once:!0}),e.readyState==="open"?c():e.readyState==="closed"&&c(new Error("WebRTC DataChannel is closed"))})}}sendRequest(e,t,r){if(t.signal?.aborted)return Promise.reject(y(t.signal));if(e.dataChannel.readyState!=="open")return Promise.reject(new Error("WebRTC DataChannel is not open"));const o=this.createId(`request${++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:k(t.body),protocol:t.protocol}))}catch(d){this.finishPending(e,o),i(b(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:P(c.body),protocol:typeof c.protocol=="string"?c.protocol:"webrtc"};let g;if(t?g=await t(d):g={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:g.status,statusText:g.statusText??"",headers:g.headers,body:k(g.body),url:g.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:b(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{F as WebRtcTransport};
|
|
1
|
+
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};
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CHANGED
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"sources": ["../src/webrtc.ts"],
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"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"],
|
|
5
|
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|
|
6
|
-
"names": ["MAX_SIGNALING_BODY_BYTES", "abortError", "decodeWireBody", "encodeWireBody", "errorFromUnknown", "createServer", "WeriftMediaStream", "WeriftPeerConnection", "normalizeIceServers", "servers", "server", "urls", "url", "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"]
|
|
5
|
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|
|
6
|
+
"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"]
|
|
7
7
|
}
|
package/package.json
CHANGED
|
@@ -1,6 +1,6 @@
|
|
|
1
1
|
{
|
|
2
2
|
"name": "@hyperttp/transport-webrtc",
|
|
3
|
-
"version": "0.2.
|
|
3
|
+
"version": "0.2.1",
|
|
4
4
|
"description": "WebRTC transport for Hyperttp powered by Werift",
|
|
5
5
|
"keywords": [
|
|
6
6
|
"bun",
|
|
@@ -59,12 +59,12 @@
|
|
|
59
59
|
"@types/bun": "^1.4.4",
|
|
60
60
|
"@types/node": "^26.6.6",
|
|
61
61
|
"esbuild": "^0.28.2",
|
|
62
|
-
"oxfmt": "^0.
|
|
62
|
+
"oxfmt": "^0.76.0",
|
|
63
63
|
"oxlint": "^1.90.0",
|
|
64
64
|
"typescript": "^7.0.2",
|
|
65
65
|
"vitest": "^5.1.0"
|
|
66
66
|
},
|
|
67
67
|
"peerDependencies": {
|
|
68
|
-
"@hyperttp/types": "^0.5.
|
|
68
|
+
"@hyperttp/types": "^0.5.3"
|
|
69
69
|
}
|
|
70
70
|
}
|