npm - @roboflow/inference-sdk - Versions diffs - 0.1.8 → 0.1.9 - Mend

@roboflow/inference-sdk 0.1.8 → 0.1.9

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package/dist/index.es.js +264 -243
package/dist/index.es.js.map +1 -1
package/dist/index.js +1 -1
package/dist/index.js.map +1 -1
package/dist/inference-api.d.ts +14 -0
package/dist/inference-api.d.ts.map +1 -1
package/dist/webrtc.d.ts +50 -0
package/dist/webrtc.d.ts.map +1 -1
package/package.json +1 -1

package/dist/index.js.map CHANGED Viewed

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1	- {"version":3,"file":"index.js","sources":["../src/inference-api.ts","../src/streams.ts","../src/video-upload.ts","../src/webrtc.ts"],"sourcesContent":["/*\n Base URL for the Roboflow API (used for TURN server configuration)\n * Can be overridden via environment variable in Node.js environments\n /\nconst RF_API_BASE_URL = typeof process !== \"undefined\" && process.env?.RF_API_BASE_URL\n ? process.env.RF_API_BASE_URL\n : \"https://api.roboflow.com\";\n\n/\n List of known Roboflow serverless API URLs where auto TURN config applies\n /\nconst ROBOFLOW_SERVERLESS_URLS = [\n \"https://serverless.roboflow.com\"\n];\n\nexport interface WebRTCWorkerConfig {\n imageInputName?: string;\n streamOutputNames?: string[];\n dataOutputNames?: string[];\n threadPoolWorkers?: number;\n /\n Workflow parameters to pass to the workflow execution\n /\n workflowsParameters?: Record<string, any>;\n /\n ICE servers for WebRTC connections (used for both client and server)\n /\n iceServers?: RTCIceServerConfig[];\n /\n Processing timeout in seconds (serverless only)\n * @default 600\n /\n processingTimeout?: number;\n /\n Requested compute plan (serverless only)\n * @example \"webrtc-gpu-small\"\n /\n requestedPlan?: string;\n /\n Requested region for processing (serverless only)\n * @example \"us\"\n /\n requestedRegion?: string;\n /\n Set to false for file upload mode (batch processing).\n * When false, server processes all frames sequentially instead of dropping frames.\n * @default true\n /\n realtimeProcessing?: boolean;\n}\n\n/\n ICE server configuration for WebRTC connections\n \n Use this to configure custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls.\n /\nexport interface RTCIceServerConfig {\n urls: string[];\n username?: string;\n credential?: string;\n}\n\nexport interface WebRTCOffer {\n sdp: string;\n type: string;\n}\n\nexport type WorkflowSpec = Record<string, any>;\n\nexport interface WebRTCWorkerResponse {\n status?: string;\n sdp: string;\n type: string;\n context?: {\n request_id: string \| null;\n pipeline_id: string \| null;\n };\n}\n\nexport interface WebRTCParams {\n workflowSpec?: WorkflowSpec;\n workspaceName?: string;\n workflowId?: string;\n imageInputName?: string;\n streamOutputNames?: string[];\n dataOutputNames?: string[];\n threadPoolWorkers?: number;\n /\n Workflow parameters to pass to the workflow execution\n /\n workflowsParameters?: Record<string, any>;\n /\n ICE servers for WebRTC connections (used for both client and server)\n \n Use this to specify custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls. The same configuration is\n * used for both the client-side RTCPeerConnection and sent to the\n * server via webrtc_config.\n \n @example\n * ```typescript\n * iceServers: [\n * { urls: [\"stun:stun.l.google.com:19302\"] },\n * { urls: [\"turn:turn.example.com:3478\"], username: \"user\", credential: \"pass\" }\n * ]\n * ```\n /\n iceServers?: RTCIceServerConfig[];\n /\n Processing timeout in seconds (serverless only)\n * @default 600\n /\n processingTimeout?: number;\n /\n Requested compute plan (serverless only)\n * @example \"webrtc-gpu-small\"\n /\n requestedPlan?: string;\n /\n Requested region for processing (serverless only)\n * @example \"us\"\n /\n requestedRegion?: string;\n /\n Set to false for file upload mode (batch processing).\n * When false, server processes all frames sequentially instead of dropping frames.\n * @default true\n /\n realtimeProcessing?: boolean;\n}\n\nexport interface Connector {\n connectWrtc(offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse>;\n /\n Fetch ICE servers (TURN configuration) for WebRTC connections\n * This should be called BEFORE creating the RTCPeerConnection to ensure\n * proper NAT traversal configuration.\n \n @returns Promise resolving to ICE server configuration, or null/undefined if not available\n /\n getIceServers?(): Promise<RTCIceServerConfig[] \| null>;\n _apiKey?: string;\n _serverUrl?: string;\n}\n\nexport class InferenceHTTPClient {\n private apiKey: string;\n private serverUrl: string;\n\n /\n @private\n * Use InferenceHTTPClient.init() instead\n /\n private constructor(apiKey: string, serverUrl: string = \"https://serverless.roboflow.com\") {\n this.apiKey = apiKey;\n this.serverUrl = serverUrl;\n }\n\n static init({ apiKey, serverUrl }: { apiKey: string; serverUrl?: string }): InferenceHTTPClient {\n if (!apiKey) {\n throw new Error(\"apiKey is required\");\n }\n return new InferenceHTTPClient(apiKey, serverUrl);\n }\n\n /\n Initialize a WebRTC worker pipeline\n \n @param params - Pipeline parameters\n * @param params.offer - WebRTC offer { sdp, type }\n * @param params.workflowSpec - Workflow specification\n * @param params.config - Additional configuration\n * @param params.config.imageInputName - Input image name (default: \"image\")\n * @param params.config.streamOutputNames - Output stream names for video (default: [])\n * @param params.config.dataOutputNames - Output data names (default: [\"string\"])\n * @param params.config.threadPoolWorkers - Thread pool workers (default: 4)\n * @returns Promise resolving to answer with SDP and pipeline ID\n \n @example\n * ```typescript\n * const answer = await client.initializeWebrtcWorker({\n * offer: { sdp, type },\n * workflowSpec: { ... },\n * config: {\n * imageInputName: \"image\",\n * streamOutputNames: [\"output_image\"]\n * }\n * });\n * ```\n /\n async initializeWebrtcWorker({\n offer,\n workflowSpec,\n workspaceName,\n workflowId,\n config = {}\n }: {\n offer: WebRTCOffer;\n workflowSpec?: WorkflowSpec;\n workspaceName?: string;\n workflowId?: string;\n config?: WebRTCWorkerConfig;\n }): Promise<WebRTCWorkerResponse> {\n if (!offer \|\| !offer.sdp \|\| !offer.type) {\n throw new Error(\"offer with sdp and type is required\");\n }\n\n // Validate that either workflowSpec OR (workspaceName + workflowId) is provided\n const hasWorkflowSpec = !!workflowSpec;\n const hasWorkspaceIdentifier = !!(workspaceName && workflowId);\n\n if (!hasWorkflowSpec && !hasWorkspaceIdentifier) {\n throw new Error(\"Either workflowSpec OR (workspaceName + workflowId) is required\");\n }\n if (hasWorkflowSpec && hasWorkspaceIdentifier) {\n throw new Error(\"Provide either workflowSpec OR (workspaceName + workflowId), not both\");\n }\n\n const {\n imageInputName = \"image\",\n streamOutputNames = [],\n dataOutputNames = [\"string\"],\n threadPoolWorkers = 4,\n workflowsParameters = {},\n iceServers,\n processingTimeout,\n requestedPlan,\n requestedRegion,\n realtimeProcessing = true\n } = config as any;\n\n // Build workflow_configuration based on what's provided\n const workflowConfiguration: any = {\n type: \"WorkflowConfiguration\",\n image_input_name: imageInputName,\n workflows_parameters: workflowsParameters,\n workflows_thread_pool_workers: threadPoolWorkers,\n cancel_thread_pool_tasks_on_exit: true,\n video_metadata_input_name: \"video_metadata\"\n };\n\n if (hasWorkflowSpec) {\n workflowConfiguration.workflow_specification = workflowSpec;\n } else {\n workflowConfiguration.workspace_name = workspaceName;\n workflowConfiguration.workflow_id = workflowId;\n }\n\n const payload: Record<string, any> = {\n workflow_configuration: workflowConfiguration,\n api_key: this.apiKey,\n webrtc_realtime_processing: realtimeProcessing,\n webrtc_offer: {\n sdp: offer.sdp,\n type: offer.type\n },\n webrtc_config: iceServers ? { iceServers } : null,\n stream_output: streamOutputNames,\n data_output: dataOutputNames\n };\n\n // Add serverless-specific fields if provided\n if (processingTimeout !== undefined) {\n payload.processing_timeout = processingTimeout;\n }\n if (requestedPlan !== undefined) {\n payload.requested_plan = requestedPlan;\n }\n if (requestedRegion !== undefined) {\n payload.requested_region = requestedRegion;\n }\n const response = await fetch(`${this.serverUrl}/initialise_webrtc_worker`, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify(payload)\n });\n\n if (!response.ok) {\n const errorText = await response.text().catch(() => \"\");\n throw new Error(`initialise_webrtc_worker failed (${response.status}): ${errorText}`);\n }\n\n const result = await response.json();\n\n return result;\n }\n\n async terminatePipeline({ pipelineId }: { pipelineId: string }): Promise<void> {\n if (!pipelineId) {\n throw new Error(\"pipelineId is required\");\n }\n\n await fetch(\n `${this.serverUrl}/inference_pipelines/${pipelineId}/terminate?api_key=${this.apiKey}`,\n {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" }\n }\n );\n }\n\n /\n Fetch TURN server configuration from Roboflow API\n \n This automatically fetches TURN server credentials for improved WebRTC\n * connectivity through firewalls and NAT. Only applicable when using\n * Roboflow serverless infrastructure.\n \n @returns Promise resolving to ICE server configuration, or null if not applicable\n \n @example\n * ```typescript\n * const client = InferenceHTTPClient.init({ apiKey: \"your-api-key\" });\n * const iceServers = await client.fetchTurnConfig();\n * // Returns: [{ urls: [\"turn:...\"], username: \"...\", credential: \"...\" }]\n * ```\n /\n async fetchTurnConfig(): Promise<RTCIceServerConfig[] \| null> {\n // // Only fetch TURN config for Roboflow serverless URLs\n if (!ROBOFLOW_SERVERLESS_URLS.includes(this.serverUrl)) {\n return null;\n }\n try {\n const response = await fetch(\n `${RF_API_BASE_URL}/webrtc_turn_config?api_key=${this.apiKey}`,\n {\n method: \"GET\",\n headers: { \"Content-Type\": \"application/json\" }\n }\n );\n\n if (!response.ok) {\n console.warn(`[RFWebRTC] Failed to fetch TURN config (${response.status}), using defaults`);\n return null;\n }\n\n const turnConfig = await response.json();\n\n // Handle 3 formats:\n // 1. Single server object: { urls, username, credential }\n // 2. Array of servers: [{ urls, username, credential }, ...]\n // 3. Object with iceServers: { iceServers: [...] }\n let iceServersRaw: any[];\n\n if (Array.isArray(turnConfig)) {\n // Format 2: array of servers\n iceServersRaw = turnConfig;\n } else if (turnConfig.iceServers && Array.isArray(turnConfig.iceServers)) {\n // Format 3: object with iceServers array\n iceServersRaw = turnConfig.iceServers;\n } else if (turnConfig.urls) {\n // Format 1: single server object - wrap in array\n iceServersRaw = [turnConfig];\n } else {\n console.warn(\"[RFWebRTC] Invalid TURN config format, using defaults\");\n return null;\n }\n\n // Normalize the ICE servers format\n const iceServers: RTCIceServerConfig[] = iceServersRaw.map((server: any) => ({\n urls: Array.isArray(server.urls) ? server.urls : [server.urls],\n username: server.username,\n credential: server.credential\n }));\n\n return iceServers;\n } catch (err) {\n console.warn(\"[RFWebRTC] Error fetching TURN config:\", err);\n return null;\n }\n }\n}\n\n/\n Connectors for establishing WebRTC connections to Roboflow\n /\nexport const connectors = {\n /\n Create a connector that uses API key directly\n \n WARNING: If you use this in the frontend, it will expose your API key. \n * Use only for demos/testing.\n * For production, use withProxyUrl() with a backend proxy.\n \n @param apiKey - Roboflow API key\n * @param options - Additional options\n * @param options.serverUrl - Custom Roboflow server URL\n * @returns Connector with connectWrtc method\n \n @example\n * ```typescript\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const answer = await connector.connectWrtc(offer, wrtcParams);\n * ```\n /\n withApiKey(apiKey: string, options: { serverUrl?: string } = {}): Connector {\n const { serverUrl } = options;\n\n // Warn if running in browser context\n if (typeof window !== 'undefined') {\n console.warn(\n '[Security Warning] Using API key directly in browser will expose it. ' +\n 'Use connectors.withProxyUrl() for production. ' +\n 'See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key'\n );\n }\n\n const client = InferenceHTTPClient.init({ apiKey, serverUrl });\n\n return {\n connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n console.debug(\"wrtcParams\", wrtcParams);\n const answer = await client.initializeWebrtcWorker({\n offer,\n workflowSpec: wrtcParams.workflowSpec,\n workspaceName: wrtcParams.workspaceName,\n workflowId: wrtcParams.workflowId,\n config: {\n imageInputName: wrtcParams.imageInputName,\n streamOutputNames: wrtcParams.streamOutputNames,\n dataOutputNames: wrtcParams.dataOutputNames,\n threadPoolWorkers: wrtcParams.threadPoolWorkers,\n workflowsParameters: wrtcParams.workflowsParameters,\n iceServers: wrtcParams.iceServers,\n processingTimeout: wrtcParams.processingTimeout,\n requestedPlan: wrtcParams.requestedPlan,\n requestedRegion: wrtcParams.requestedRegion,\n realtimeProcessing: wrtcParams.realtimeProcessing\n }\n });\n\n return answer;\n },\n\n /\n Fetch TURN server configuration for improved WebRTC connectivity\n /\n getIceServers: async (): Promise<RTCIceServerConfig[] \| null> => {\n return await client.fetchTurnConfig();\n },\n\n // Store apiKey for cleanup\n _apiKey: apiKey,\n _serverUrl: serverUrl\n };\n },\n\n /\n Create a connector that uses a backend proxy (recommended for production)\n \n Your backend receives the offer and wrtcParams, adds the secret API key,\n * and forwards to Roboflow. This keeps your API key secure.\n \n For improved WebRTC connectivity through firewalls, implement a separate\n * endpoint for TURN server configuration that calls `fetchTurnConfig()`.\n \n @param proxyUrl - Backend proxy endpoint URL for WebRTC initialization\n * @param options - Additional options\n * @param options.turnConfigUrl - Optional URL for fetching TURN server configuration\n * @returns Connector with connectWrtc and optional getIceServers methods\n \n @example\n * ```typescript\n * // Frontend: Create connector with TURN config endpoint\n * const connector = connectors.withProxyUrl('/api/init-webrtc', {\n * turnConfigUrl: '/api/turn-config'\n * });\n * ```\n \n @example\n * Backend implementation (Express) with TURN server support:\n * ```typescript\n * // Endpoint for TURN configuration (called first by SDK)\n * app.get('/api/turn-config', async (req, res) => {\n * const client = InferenceHTTPClient.init({\n * apiKey: process.env.ROBOFLOW_API_KEY\n * });\n * const iceServers = await client.fetchTurnConfig();\n * res.json({ iceServers });\n * });\n \n // Endpoint for WebRTC initialization\n * app.post('/api/init-webrtc', async (req, res) => {\n * const { offer, wrtcParams } = req.body;\n * const client = InferenceHTTPClient.init({\n * apiKey: process.env.ROBOFLOW_API_KEY\n * });\n \n const answer = await client.initializeWebrtcWorker({\n * offer,\n * workflowSpec: wrtcParams.workflowSpec,\n * workspaceName: wrtcParams.workspaceName,\n * workflowId: wrtcParams.workflowId,\n * config: {\n * imageInputName: wrtcParams.imageInputName,\n * streamOutputNames: wrtcParams.streamOutputNames,\n * dataOutputNames: wrtcParams.dataOutputNames,\n * threadPoolWorkers: wrtcParams.threadPoolWorkers,\n * workflowsParameters: wrtcParams.workflowsParameters,\n * iceServers: wrtcParams.iceServers,\n * processingTimeout: wrtcParams.processingTimeout,\n * requestedPlan: wrtcParams.requestedPlan,\n * requestedRegion: wrtcParams.requestedRegion\n * }\n * });\n \n res.json(answer);\n * });\n * ```\n /\n withProxyUrl(proxyUrl: string, options: { turnConfigUrl?: string } = {}): Connector {\n const { turnConfigUrl } = options;\n\n return {\n connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n const response = await fetch(proxyUrl, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify({\n offer,\n wrtcParams\n })\n });\n\n if (!response.ok) {\n const errorText = await response.text().catch(() => \"\");\n throw new Error(`Proxy request failed (${response.status}): ${errorText}`);\n }\n\n return await response.json();\n },\n\n /\n Fetch TURN server configuration from the proxy backend\n * Only available if turnConfigUrl was provided\n /\n getIceServers: turnConfigUrl\n ? async (): Promise<RTCIceServerConfig[] \| null> => {\n try {\n const response = await fetch(turnConfigUrl, {\n method: \"GET\",\n headers: { \"Content-Type\": \"application/json\" }\n });\n\n if (!response.ok) {\n console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${response.status})`);\n return null;\n }\n\n const data = await response.json();\n return data.iceServers \|\| null;\n } catch (err) {\n console.warn(\"[RFWebRTC] Error fetching TURN config from proxy:\", err);\n return null;\n }\n }\n : undefined\n };\n }\n};\n","/\n Get a camera stream with the given constraints.\n \n @param constraints - MediaStreamConstraints for getUserMedia\n * @returns Promise that resolves to MediaStream\n \n @example\n * ```typescript\n * const stream = await useCamera({\n * video: {\n * facingMode: { ideal: \"user\" },\n * width: { ideal: 1280 },\n * height: { ideal: 720 },\n * frameRate: { ideal: 30 }\n * },\n * audio: false\n * });\n * ```\n /\nexport async function useCamera(constraints: MediaStreamConstraints = { video: true }): Promise<MediaStream> {\n try {\n console.log(\"[RFStreams] requesting with\", constraints);\n const stream = await navigator.mediaDevices.getUserMedia(constraints);\n console.log(\"[RFStreams] got stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n return stream;\n } catch (err) {\n console.warn(\"[RFStreams] failed, falling back\", err);\n const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: false });\n console.log(\"[RFStreams] fallback stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n return stream;\n }\n}\n \nexport function stopStream(stream: MediaStream \| null \| undefined): void {\n if (stream) {\n stream.getTracks().forEach(track => track.stop());\n console.log(\"[RFStreams] Stream stopped\");\n }\n}\n","/\n Video file upload via WebRTC datachannel\n \n This module provides the FileUploader class for chunked file uploads\n * through WebRTC datachannels with backpressure handling.\n /\n\n/\n Configuration constants for file upload (matching Python SDK)\n /\nconst CHUNK_SIZE = 49152; // 49KB - safe for WebRTC\nconst BUFFER_LIMIT = 262144; // 256KB - backpressure threshold\nconst POLL_INTERVAL = 10; // 10ms buffer check interval\n\n/\n Helper to sleep for a given number of milliseconds\n /\nfunction sleep(ms: number): Promise<void> {\n return new Promise(resolve => setTimeout(resolve, ms));\n}\n\n/\n FileUploader handles chunked file upload with backpressure\n \n Uploads files through a WebRTC datachannel in 49KB chunks with\n * intelligent backpressure handling to prevent overwhelming the network.\n /\nexport class FileUploader {\n private file: File;\n private channel: RTCDataChannel;\n private totalChunks: number;\n private cancelled: boolean = false;\n\n constructor(file: File, channel: RTCDataChannel) {\n this.file = file;\n this.channel = channel;\n this.totalChunks = Math.ceil(file.size / CHUNK_SIZE);\n }\n\n /\n Cancel the upload\n /\n cancel(): void {\n this.cancelled = true;\n }\n\n /\n Upload the file in chunks with backpressure handling\n \n @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n /\n async upload(onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n const totalBytes = this.file.size;\n\n for (let chunkIndex = 0; chunkIndex < this.totalChunks; chunkIndex++) {\n // Check for cancellation\n if (this.cancelled) {\n throw new Error(\"Upload cancelled\");\n }\n\n // Check channel state\n if (this.channel.readyState !== \"open\") {\n throw new Error(\"Video upload interrupted\");\n }\n\n // Read chunk from file\n const start = chunkIndex CHUNK_SIZE;\n const end = Math.min(start + CHUNK_SIZE, totalBytes);\n const chunkBlob = this.file.slice(start, end);\n const chunkData = new Uint8Array(await chunkBlob.arrayBuffer());\n\n // Create message with 8-byte header (chunkIndex + totalChunks as uint32 LE)\n const message = new ArrayBuffer(8 + chunkData.length);\n const view = new DataView(message);\n view.setUint32(0, chunkIndex, true); // little-endian\n view.setUint32(4, this.totalChunks, true); // little-endian\n new Uint8Array(message, 8).set(chunkData);\n\n // Backpressure: wait for buffer to drain\n while (this.channel.bufferedAmount > BUFFER_LIMIT) {\n if (this.channel.readyState !== \"open\") {\n throw new Error(\"Video upload interrupted\");\n }\n await sleep(POLL_INTERVAL);\n }\n\n // Send chunk\n this.channel.send(message);\n\n // Report progress\n if (onProgress) {\n onProgress(end, totalBytes);\n }\n }\n }\n}\n","\nimport { InferenceHTTPClient, Connector, WebRTCParams, RTCIceServerConfig } from \"./inference-api\";\nimport { stopStream } from \"./streams\";\nimport { WebRTCOutputData } from \"./webrtc-types\";\nimport { FileUploader } from \"./video-upload\";\n\n// Re-export shared types\nexport type { WebRTCVideoMetadata, WebRTCOutputData } from \"./webrtc-types\";\n\n// Re-export FileUploader from video-upload\nexport { FileUploader } from \"./video-upload\";\n\n/*\n Binary protocol header size (frame_id + chunk_index + total_chunks)\n * Each field is 4 bytes uint32 little-endian\n /\nconst HEADER_SIZE = 12;\n\n/\n Reassembles chunked binary messages from the datachannel\n /\nexport class ChunkReassembler {\n private pendingFrames: Map<number, {\n chunks: Map<number, Uint8Array>;\n totalChunks: number;\n }> = new Map();\n\n /\n Process an incoming chunk and return the complete message if all chunks received\n /\n processChunk(frameId: number, chunkIndex: number, totalChunks: number, payload: Uint8Array): Uint8Array \| null {\n // Single chunk message - return immediately\n if (totalChunks === 1) {\n return payload;\n }\n\n // Multi-chunk message - accumulate\n if (!this.pendingFrames.has(frameId)) {\n this.pendingFrames.set(frameId, {\n chunks: new Map(),\n totalChunks\n });\n }\n\n const frame = this.pendingFrames.get(frameId)!;\n frame.chunks.set(chunkIndex, payload);\n\n // Check if all chunks received\n if (frame.chunks.size === totalChunks) {\n // Reassemble in order\n const totalLength = Array.from(frame.chunks.values()).reduce((sum, chunk) => sum + chunk.length, 0);\n const result = new Uint8Array(totalLength);\n let offset = 0;\n\n for (let i = 0; i < totalChunks; i++) {\n const chunk = frame.chunks.get(i)!;\n result.set(chunk, offset);\n offset += chunk.length;\n }\n\n this.pendingFrames.delete(frameId);\n return result;\n }\n\n return null;\n }\n\n /\n Clear all pending frames (for cleanup)\n /\n clear(): void {\n this.pendingFrames.clear();\n }\n}\n\n/\n Parse the binary header from a datachannel message\n /\nexport function parseBinaryHeader(buffer: ArrayBuffer): { frameId: number; chunkIndex: number; totalChunks: number; payload: Uint8Array } {\n const view = new DataView(buffer);\n const frameId = view.getUint32(0, true); // little-endian\n const chunkIndex = view.getUint32(4, true); // little-endian\n const totalChunks = view.getUint32(8, true); // little-endian\n const payload = new Uint8Array(buffer, HEADER_SIZE);\n\n return { frameId, chunkIndex, totalChunks, payload };\n}\n\nexport interface UseStreamOptions {\n disableInputStreamDownscaling?: boolean;\n}\n\nexport interface UseStreamParams {\n source: MediaStream;\n connector: Connector;\n wrtcParams: WebRTCParams;\n onData?: (data: WebRTCOutputData) => void;\n options?: UseStreamOptions;\n}\n\nasync function waitForIceGathering(pc: RTCPeerConnection, timeoutMs = 6000): Promise<void> {\n if (pc.iceGatheringState === \"complete\") return;\n\n let hasSrflx = false;\n\n // Track if we get a good candidate (srflx = public IP via STUN)\n const candidateHandler = (event: RTCPeerConnectionIceEvent) => {\n if (event.candidate && event.candidate.type === \"srflx\") {\n hasSrflx = true;\n }\n };\n pc.addEventListener(\"icecandidate\", candidateHandler);\n\n try {\n await Promise.race([\n new Promise<void>(resolve => {\n const check = () => {\n if (pc.iceGatheringState === \"complete\") {\n pc.removeEventListener(\"icegatheringstatechange\", check);\n resolve();\n }\n };\n pc.addEventListener(\"icegatheringstatechange\", check);\n }),\n new Promise<void>((resolve, reject) => {\n setTimeout(() => {\n if (!hasSrflx) {\n console.error(\"[ICE] timeout with NO srflx candidate! Connection may fail.\");\n reject(new Error(\"ICE gathering timeout without srflx candidate\"));\n } else {\n resolve();\n }\n }, timeoutMs);\n })\n ]);\n } finally {\n pc.removeEventListener(\"icecandidate\", candidateHandler);\n }\n}\n\nfunction setupRemoteStreamListener(pc: RTCPeerConnection): Promise<MediaStream> {\n return new Promise((resolve) => {\n pc.addEventListener(\"track\", (event: RTCTrackEvent) => {\n if (event.streams && event.streams[0]) {\n resolve(event.streams[0]);\n }\n });\n });\n}\n\nconst DEFAULT_ICE_SERVERS: RTCIceServerConfig[] = [\n { urls: [\"stun:stun.l.google.com:19302\"] }\n];\n\nasync function preparePeerConnection(\n localStream?: MediaStream,\n file?: File,\n customIceServers?: RTCIceServerConfig[]\n): Promise<{\n pc: RTCPeerConnection;\n offer: RTCSessionDescriptionInit;\n remoteStreamPromise: Promise<MediaStream>;\n dataChannel: RTCDataChannel;\n uploadChannel?: RTCDataChannel;\n}> {\n if (!localStream && !file \|\| (localStream && file)) {\n throw new Error(\"Either localStream or file must be provided, but not both\");\n }\n const iceServers = customIceServers ?? DEFAULT_ICE_SERVERS;\n\n const pc = new RTCPeerConnection({\n iceServers: iceServers as RTCIceServer[]\n });\n\n // Add transceiver for receiving remote video (BEFORE adding tracks - order matters!)\n try {\n pc.addTransceiver(\"video\", { direction: \"recvonly\" });\n } catch (err) {\n console.warn(\"[RFWebRTC] Could not add transceiver:\", err);\n }\n\n if (localStream) {\n // Add local tracks\n localStream.getVideoTracks().forEach(track => {\n try {\n // @ts-ignore - contentHint is not in all TypeScript definitions\n track.contentHint = \"detail\";\n } catch (e) {\n // Ignore if contentHint not supported\n }\n pc.addTrack(track, localStream);\n });\n }\n\n // Setup remote stream listener\n const remoteStreamPromise = setupRemoteStreamListener(pc);\n\n // Create control datachannel (named \"inference\" to match Python SDK)\n const dataChannel = pc.createDataChannel(\"inference\", {\n ordered: true\n });\n\n // Create upload datachannel for file uploads\n let uploadChannel: RTCDataChannel \| undefined;\n if (file) {\n uploadChannel = pc.createDataChannel(\"video_upload\");\n }\n\n // Create offer\n const offer = await pc.createOffer();\n await pc.setLocalDescription(offer);\n\n // Wait for ICE gathering\n await waitForIceGathering(pc);\n\n return {\n pc,\n offer: pc.localDescription!,\n remoteStreamPromise,\n dataChannel,\n uploadChannel\n };\n}\n\n/\n Disable input stream downscaling\n * @private\n /\nasync function disableInputStreamDownscaling(pc: RTCPeerConnection): Promise<void> {\n const sender = pc.getSenders().find(s => s.track && s.track.kind === \"video\");\n if (!sender) return;\n\n const params = sender.getParameters();\n params.encodings = params.encodings \|\| [{}];\n params.encodings[0].scaleResolutionDownBy = 1;\n\n try {\n await sender.setParameters(params);\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to set encoding parameters:\", err);\n }\n}\n\n/\n Helper to wait for datachannel to open\n /\nfunction waitForChannelOpen(channel: RTCDataChannel, timeoutMs = 30000): Promise<void> {\n return new Promise((resolve, reject) => {\n if (channel.readyState === \"open\") {\n resolve();\n return;\n }\n\n const openHandler = () => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n clearTimeout(timeout);\n resolve();\n };\n\n const errorHandler = () => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n clearTimeout(timeout);\n reject(new Error(\"Datachannel error\"));\n };\n\n const timeout = setTimeout(() => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n reject(new Error(\"Datachannel open timeout\"));\n }, timeoutMs);\n\n channel.addEventListener(\"open\", openHandler);\n channel.addEventListener(\"error\", errorHandler);\n });\n}\n\n/\n WebRTC Connection object\n \n Represents an active WebRTC connection to Roboflow for streaming inference\n * or file-based batch processing.\n /\nexport class RFWebRTCConnection {\n private pc: RTCPeerConnection;\n private _localStream?: MediaStream;\n private remoteStreamPromise: Promise<MediaStream>;\n private pipelineId: string \| null;\n private apiKey: string \| null;\n private dataChannel: RTCDataChannel;\n private reassembler: ChunkReassembler;\n private uploadChannel?: RTCDataChannel;\n private uploader?: FileUploader;\n private onComplete?: () => void;\n\n /* @private /\n constructor(\n pc: RTCPeerConnection,\n remoteStreamPromise: Promise<MediaStream>,\n pipelineId: string \| null,\n apiKey: string \| null,\n dataChannel: RTCDataChannel,\n options?: {\n localStream?: MediaStream;\n uploadChannel?: RTCDataChannel;\n onData?: (data: any) => void;\n onComplete?: () => void;\n }\n ) {\n this.pc = pc;\n this._localStream = options?.localStream;\n this.remoteStreamPromise = remoteStreamPromise;\n this.pipelineId = pipelineId;\n this.apiKey = apiKey;\n this.dataChannel = dataChannel;\n this.reassembler = new ChunkReassembler();\n this.uploadChannel = options?.uploadChannel;\n this.onComplete = options?.onComplete;\n\n // Set binary mode for datachannel\n this.dataChannel.binaryType = \"arraybuffer\";\n\n const onData = options?.onData;\n\n // Setup data channel event listeners\n if (onData) {\n this.dataChannel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n try {\n // Handle binary protocol with chunking\n if (messageEvent.data instanceof ArrayBuffer) {\n const { frameId, chunkIndex, totalChunks, payload } = parseBinaryHeader(messageEvent.data);\n const completePayload = this.reassembler.processChunk(frameId, chunkIndex, totalChunks, payload);\n\n if (completePayload) {\n // Decode UTF-8 JSON payload\n const decoder = new TextDecoder(\"utf-8\");\n const jsonString = decoder.decode(completePayload);\n const data = JSON.parse(jsonString);\n onData(data);\n }\n } else {\n // Fallback for string messages (shouldn't happen with new protocol)\n const data = JSON.parse(messageEvent.data);\n onData(data);\n }\n } catch (err) {\n console.error(\"[RFWebRTC] Failed to parse data channel message:\", err);\n }\n });\n\n this.dataChannel.addEventListener(\"error\", (error) => {\n console.error(\"[RFWebRTC] Data channel error:\", error);\n });\n }\n\n // Handle channel close - call onComplete when processing finishes\n this.dataChannel.addEventListener(\"close\", () => {\n this.reassembler.clear();\n if (this.onComplete) {\n this.onComplete();\n }\n });\n }\n\n /\n Get the remote stream (processed video from Roboflow)\n \n @returns Promise resolving to the remote MediaStream\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n /\n async remoteStream(): Promise<MediaStream> {\n return await this.remoteStreamPromise;\n }\n\n /\n Get the local stream (original camera)\n \n @returns The local MediaStream, or undefined if using file upload mode\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * const localStream = conn.localStream();\n * if (localStream) {\n * videoElement.srcObject = localStream;\n * }\n * ```\n /\n localStream(): MediaStream \| undefined {\n return this._localStream;\n }\n\n /\n Cleanup and close connection\n \n Terminates the pipeline on Roboflow, closes the peer connection,\n * and stops the local media stream (if applicable).\n \n @returns Promise that resolves when cleanup is complete\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * // ... use connection ...\n * await conn.cleanup(); // Clean up when done\n * ```\n /\n async cleanup(): Promise<void> {\n // Cancel any ongoing upload\n if (this.uploader) {\n this.uploader.cancel();\n }\n\n // Clear pending chunks\n this.reassembler.clear();\n\n // Terminate pipeline\n if (this.pipelineId && this.apiKey) {\n try {\n const client = InferenceHTTPClient.init({ apiKey: this.apiKey });\n await client.terminatePipeline({ pipelineId: this.pipelineId });\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to terminate pipeline:\", err);\n }\n }\n\n // Close peer connection\n if (this.pc && this.pc.connectionState !== \"closed\") {\n this.pc.close();\n }\n\n // Stop local stream if present\n if (this._localStream) {\n stopStream(this._localStream);\n }\n }\n\n /\n Start uploading a file through the connection\n \n @param file - The file to upload\n * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n * @returns Promise that resolves when upload is complete\n * @throws Error if no upload channel is available\n \n @example\n * ```typescript\n * await connection.startUpload(videoFile, (uploaded, total) => {\n * console.log(`Upload progress: ${(uploaded / total * 100).toFixed(1)}%`);\n * });\n * ```\n /\n async startUpload(file: File, onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n if (!this.uploadChannel) {\n throw new Error(\"No upload channel available. This connection was not created for file uploads.\");\n }\n\n // Wait for upload channel to open\n await waitForChannelOpen(this.uploadChannel);\n\n this.uploader = new FileUploader(file, this.uploadChannel);\n await this.uploader.upload(onProgress);\n }\n\n /\n Cancel any ongoing file upload\n /\n cancelUpload(): void {\n if (this.uploader) {\n this.uploader.cancel();\n }\n }\n\n /\n Reconfigure pipeline outputs at runtime\n \n Dynamically change stream and data outputs without restarting the connection.\n * Set a field to `null` to leave it unchanged, or to `null` value to enable all outputs,\n * or to `[]` to disable/auto-detect.\n \n @param config - Output configuration\n * @param config.streamOutput - Stream output names (null = unchanged, [] = auto-detect, [\"name\"] = specific output)\n * @param config.dataOutput - Data output names (null = unchanged, [] = disable, [\"name\"] = specific outputs, null value = all outputs)\n \n @example\n * ```typescript\n * // Change to different stream output\n * connection.reconfigureOutputs({\n * streamOutput: [\"annotated_image\"],\n * dataOutput: null // unchanged\n * });\n \n // Enable all data outputs\n * connection.reconfigureOutputs({\n * streamOutput: null, // unchanged\n * dataOutput: null // null value = all outputs\n * });\n \n // Disable all data outputs\n * connection.reconfigureOutputs({\n * streamOutput: null, // unchanged\n * dataOutput: [] // empty array = disable\n * });\n * ```\n /\n reconfigureOutputs(config: { streamOutput?: string[] \| null; dataOutput?: string[] \| null }): void {\n const message: any = {};\n\n if (config.streamOutput !== undefined) {\n message.stream_output = config.streamOutput;\n }\n\n if (config.dataOutput !== undefined) {\n message.data_output = config.dataOutput;\n }\n\n this.sendData(message);\n }\n\n /\n Send data through the data channel\n * @private\n /\n private sendData(data: any): void {\n if (this.dataChannel.readyState !== \"open\") {\n console.warn(\"[RFWebRTC] Data channel is not open. Current state:\", this.dataChannel.readyState);\n return;\n }\n\n try {\n const message = typeof data === \"string\" ? data : JSON.stringify(data);\n this.dataChannel.send(message);\n } catch (err) {\n console.error(\"[RFWebRTC] Failed to send data:\", err);\n }\n }\n}\n\n/\n Internal base function for establishing WebRTC connection\n * Used by both useStream and useVideoFile\n * @private\n /\ninterface BaseUseStreamParams {\n source: MediaStream \| File;\n connector: Connector;\n wrtcParams: WebRTCParams;\n onData?: (data: WebRTCOutputData) => void;\n onComplete?: () => void;\n onFileUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n options?: UseStreamOptions;\n}\n\nasync function baseUseStream({\n source,\n connector,\n wrtcParams,\n onData,\n onComplete,\n onFileUploadProgress,\n options = {}\n}: BaseUseStreamParams): Promise<RFWebRTCConnection> {\n // Validate connector\n if (!connector \|\| typeof connector.connectWrtc !== \"function\") {\n throw new Error(\"connector must have a connectWrtc method\");\n }\n\n const isFile = source instanceof File;\n const localStream = isFile ? undefined : source;\n const file = isFile ? source : undefined;\n\n // Step 1: Determine ICE servers to use\n // Priority: 1) User-provided in wrtcParams, 2) From connector.getIceServers(), 3) Defaults\n let iceServers = wrtcParams.iceServers;\n if ((!iceServers \|\| iceServers.length === 0) && connector.getIceServers) {\n try {\n const turnConfig = await connector.getIceServers();\n if (turnConfig && turnConfig.length > 0) {\n iceServers = turnConfig;\n console.log(\"[RFWebRTC] Using TURN servers from connector\");\n }\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to fetch TURN config, using defaults:\", err);\n }\n }\n\n // Step 2: Prepare peer connection and create offer\n const { pc, offer, remoteStreamPromise, dataChannel, uploadChannel } = await preparePeerConnection(\n localStream,\n file,\n iceServers\n );\n\n // Update wrtcParams with resolved iceServers so server also uses them\n // For file uploads, default to batch mode (realtimeProcessing: false)\n const resolvedWrtcParams = {\n ...wrtcParams,\n iceServers: iceServers,\n realtimeProcessing: wrtcParams.realtimeProcessing ?? !isFile\n };\n\n // Step 3: Call connector.connectWrtc to exchange SDP and get answer\n const answer = await connector.connectWrtc(\n { sdp: offer.sdp!, type: offer.type! },\n resolvedWrtcParams\n );\n\n // API returns sdp and type at root level\n const sdpAnswer = { sdp: answer.sdp, type: answer.type } as RTCSessionDescriptionInit;\n\n if (!sdpAnswer?.sdp \|\| !sdpAnswer?.type) {\n console.error(\"[RFWebRTC] Invalid answer from server:\", answer);\n throw new Error(\"connector.connectWrtc must return answer with sdp and type\");\n }\n\n const pipelineId = answer?.context?.pipeline_id \|\| null;\n\n // Step 4: Set remote description\n await pc.setRemoteDescription(sdpAnswer);\n\n // Step 5: Wait for connection to establish\n await new Promise<void>((resolve, reject) => {\n const checkState = () => {\n if (pc.connectionState === \"connected\") {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n resolve();\n } else if (pc.connectionState === \"failed\") {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n reject(new Error(\"WebRTC connection failed\"));\n }\n };\n\n pc.addEventListener(\"connectionstatechange\", checkState);\n checkState(); // Check immediately in case already connected\n\n // Timeout after 30 seconds\n setTimeout(() => {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n reject(new Error(\"WebRTC connection timeout after 30s\"));\n }, 30000);\n });\n\n // Step 6: Optimize quality for MediaStream (disable downsampling by default)\n if (localStream) {\n const shouldDisableDownscaling = options.disableInputStreamDownscaling !== false;\n if (shouldDisableDownscaling) {\n await disableInputStreamDownscaling(pc);\n }\n }\n\n // Get apiKey from connector if available (for cleanup)\n const apiKey = connector._apiKey \|\| null;\n\n // Step 7: Create connection object\n const connection = new RFWebRTCConnection(\n pc,\n remoteStreamPromise,\n pipelineId,\n apiKey,\n dataChannel,\n {\n localStream,\n uploadChannel,\n onData,\n onComplete\n }\n );\n\n // Step 8: Start file upload if applicable (runs in background)\n if (file && uploadChannel) {\n connection.startUpload(file, onFileUploadProgress).catch(err => {\n console.error(\"[RFWebRTC] Upload error:\", err);\n });\n }\n\n return connection;\n}\n\n/\n Main function to establish WebRTC streaming connection\n \n Creates a WebRTC connection to Roboflow for real-time inference on video streams.\n \n @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n \n @example\n * ```typescript\n * import { useStream } from 'inferencejs/webrtc';\n * import { connectors } from 'inferencejs/api';\n * import { useCamera } from 'inferencejs/streams';\n \n const connector = connectors.withApiKey(\"your-api-key\");\n * const stream = await useCamera({ video: { facingMode: { ideal: \"environment\" } } });\n * const conn = await useStream({\n * source: stream,\n * connector,\n * wrtcParams: {\n * workflowSpec: {\n * // Your workflow specification\n * },\n * imageInputName: \"image\",\n * streamOutputNames: [\"output\"],\n * dataOutputNames: [\"predictions\"]\n * },\n * onData: (data) => {\n * console.log(\"Inference results:\", data);\n * }\n * });\n \n const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n /\nexport async function useStream({\n source,\n connector,\n wrtcParams,\n onData,\n options = {}\n}: UseStreamParams): Promise<RFWebRTCConnection> {\n if (source instanceof File) {\n throw new Error(\"useStream requires a MediaStream. Use useVideoFile for File uploads.\");\n }\n\n return baseUseStream({\n source,\n connector,\n wrtcParams,\n onData,\n options\n });\n}\n\n/\n Parameters for useVideoFile function\n /\nexport interface UseVideoFileParams {\n /* The video file to upload /\n file: File;\n /* Connector for WebRTC signaling /\n connector: Connector;\n /* WebRTC parameters for the workflow /\n wrtcParams: WebRTCParams;\n /* Callback for inference results /\n onData?: (data: WebRTCOutputData) => void;\n /* Callback for upload progress /\n onUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n /* Callback when processing completes (datachannel closes) /\n onComplete?: () => void;\n}\n\n/\n Upload a video file for batch inference processing\n \n Creates a WebRTC connection to Roboflow for uploading a video file\n * and receiving inference results. The file is uploaded via datachannel\n * with intelligent backpressure handling.\n \n @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n \n @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n \n const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useVideoFile({\n * file: videoFile,\n * connector,\n * wrtcParams: {\n * workflowSpec: { ... },\n * imageInputName: \"image\",\n * dataOutputNames: [\"predictions\"]\n * },\n * onData: (data) => {\n * console.log(\"Inference results:\", data);\n * if (data.processing_complete) {\n * console.log(\"Processing complete!\");\n * }\n * },\n * onUploadProgress: (uploaded, total) => {\n * console.log(`Upload: ${(uploaded / total * 100).toFixed(1)}%`);\n * }\n * });\n \n // When done\n * await connection.cleanup();\n * ```\n */\nexport async function useVideoFile({\n file,\n connector,\n wrtcParams,\n onData,\n onUploadProgress,\n onComplete\n}: UseVideoFileParams): Promise<RFWebRTCConnection> {\n return baseUseStream({\n source: file,\n connector,\n wrtcParams: {\n ...wrtcParams,\n realtimeProcessing: wrtcParams.realtimeProcessing ?? true\n },\n onData,\n onComplete,\n onFileUploadProgress: onUploadProgress\n 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1	+ {"version":3,"file":"index.js","sources":["../src/inference-api.ts","../src/streams.ts","../src/video-upload.ts","../src/webrtc.ts"],"sourcesContent":["/*\n Base URL for the Roboflow API (used for TURN server configuration)\n * Can be overridden via environment variable in Node.js environments\n /\nconst RF_API_BASE_URL = typeof process !== \"undefined\" && process.env?.RF_API_BASE_URL\n ? process.env.RF_API_BASE_URL\n : \"https://api.roboflow.com\";\n\n/\n List of known Roboflow serverless API URLs where auto TURN config applies\n /\nconst ROBOFLOW_SERVERLESS_URLS = [\n \"https://serverless.roboflow.com\"\n];\n\nexport interface WebRTCWorkerConfig {\n imageInputName?: string;\n streamOutputNames?: string[];\n dataOutputNames?: string[];\n threadPoolWorkers?: number;\n /\n Workflow parameters to pass to the workflow execution\n /\n workflowsParameters?: Record<string, any>;\n /\n ICE servers for WebRTC connections (used for both client and server)\n /\n iceServers?: RTCIceServerConfig[];\n /\n Processing timeout in seconds (serverless only)\n * @default 600\n /\n processingTimeout?: number;\n /\n Requested compute plan (serverless only)\n * @example \"webrtc-gpu-small\"\n /\n requestedPlan?: string;\n /\n Requested region for processing (serverless only)\n * @example \"us\"\n /\n requestedRegion?: string;\n /\n Set to false for file upload mode (batch processing).\n * When false, server processes all frames sequentially instead of dropping frames.\n * @default true\n /\n realtimeProcessing?: boolean;\n /\n RTSP URL for server-side video capture.\n * When provided, the server captures video from this RTSP stream instead of receiving\n * video from the client. Supports credentials in URL format: rtsp://user:pass@host/stream\n * @example \"rtsp://camera.local/stream\"\n /\n rtspUrl?: string;\n}\n\n/\n ICE server configuration for WebRTC connections\n \n Use this to configure custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls.\n /\nexport interface RTCIceServerConfig {\n urls: string[];\n username?: string;\n credential?: string;\n}\n\nexport interface WebRTCOffer {\n sdp: string;\n type: string;\n}\n\nexport type WorkflowSpec = Record<string, any>;\n\nexport interface WebRTCWorkerResponse {\n status?: string;\n sdp: string;\n type: string;\n context?: {\n request_id: string \| null;\n pipeline_id: string \| null;\n };\n}\n\nexport interface WebRTCParams {\n workflowSpec?: WorkflowSpec;\n workspaceName?: string;\n workflowId?: string;\n imageInputName?: string;\n streamOutputNames?: string[];\n dataOutputNames?: string[];\n threadPoolWorkers?: number;\n /\n Workflow parameters to pass to the workflow execution\n /\n workflowsParameters?: Record<string, any>;\n /\n ICE servers for WebRTC connections (used for both client and server)\n \n Use this to specify custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls. The same configuration is\n * used for both the client-side RTCPeerConnection and sent to the\n * server via webrtc_config.\n \n @example\n * ```typescript\n * iceServers: [\n * { urls: [\"stun:stun.l.google.com:19302\"] },\n * { urls: [\"turn:turn.example.com:3478\"], username: \"user\", credential: \"pass\" }\n * ]\n * ```\n /\n iceServers?: RTCIceServerConfig[];\n /\n Processing timeout in seconds (serverless only)\n * @default 600\n /\n processingTimeout?: number;\n /\n Requested compute plan (serverless only)\n * @example \"webrtc-gpu-small\"\n /\n requestedPlan?: string;\n /\n Requested region for processing (serverless only)\n * @example \"us\"\n /\n requestedRegion?: string;\n /\n Set to false for file upload mode (batch processing).\n * When false, server processes all frames sequentially instead of dropping frames.\n * @default true\n /\n realtimeProcessing?: boolean;\n /\n RTSP URL for server-side video capture.\n * When provided, the server captures video from this RTSP stream instead of receiving\n * video from the client. Supports credentials in URL format: rtsp://user:pass@host/stream\n * @example \"rtsp://camera.local/stream\"\n /\n rtspUrl?: string;\n}\n\nexport interface Connector {\n connectWrtc(offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse>;\n /\n Fetch ICE servers (TURN configuration) for WebRTC connections\n * This should be called BEFORE creating the RTCPeerConnection to ensure\n * proper NAT traversal configuration.\n \n @returns Promise resolving to ICE server configuration, or null/undefined if not available\n /\n getIceServers?(): Promise<RTCIceServerConfig[] \| null>;\n _apiKey?: string;\n _serverUrl?: string;\n}\n\nexport class InferenceHTTPClient {\n private apiKey: string;\n private serverUrl: string;\n\n /\n @private\n * Use InferenceHTTPClient.init() instead\n /\n private constructor(apiKey: string, serverUrl: string = \"https://serverless.roboflow.com\") {\n this.apiKey = apiKey;\n this.serverUrl = serverUrl;\n }\n\n static init({ apiKey, serverUrl }: { apiKey: string; serverUrl?: string }): InferenceHTTPClient {\n if (!apiKey) {\n throw new Error(\"apiKey is required\");\n }\n return new InferenceHTTPClient(apiKey, serverUrl);\n }\n\n /\n Initialize a WebRTC worker pipeline\n \n @param params - Pipeline parameters\n * @param params.offer - WebRTC offer { sdp, type }\n * @param params.workflowSpec - Workflow specification\n * @param params.config - Additional configuration\n * @param params.config.imageInputName - Input image name (default: \"image\")\n * @param params.config.streamOutputNames - Output stream names for video (default: [])\n * @param params.config.dataOutputNames - Output data names (default: [\"string\"])\n * @param params.config.threadPoolWorkers - Thread pool workers (default: 4)\n * @returns Promise resolving to answer with SDP and pipeline ID\n \n @example\n * ```typescript\n * const answer = await client.initializeWebrtcWorker({\n * offer: { sdp, type },\n * workflowSpec: { ... },\n * config: {\n * imageInputName: \"image\",\n * streamOutputNames: [\"output_image\"]\n * }\n * });\n * ```\n /\n async initializeWebrtcWorker({\n offer,\n workflowSpec,\n workspaceName,\n workflowId,\n config = {}\n }: {\n offer: WebRTCOffer;\n workflowSpec?: WorkflowSpec;\n workspaceName?: string;\n workflowId?: string;\n config?: WebRTCWorkerConfig;\n }): Promise<WebRTCWorkerResponse> {\n if (!offer \|\| !offer.sdp \|\| !offer.type) {\n throw new Error(\"offer with sdp and type is required\");\n }\n\n // Validate that either workflowSpec OR (workspaceName + workflowId) is provided\n const hasWorkflowSpec = !!workflowSpec;\n const hasWorkspaceIdentifier = !!(workspaceName && workflowId);\n\n if (!hasWorkflowSpec && !hasWorkspaceIdentifier) {\n throw new Error(\"Either workflowSpec OR (workspaceName + workflowId) is required\");\n }\n if (hasWorkflowSpec && hasWorkspaceIdentifier) {\n throw new Error(\"Provide either workflowSpec OR (workspaceName + workflowId), not both\");\n }\n\n const {\n imageInputName = \"image\",\n streamOutputNames = [],\n dataOutputNames = [\"string\"],\n threadPoolWorkers = 4,\n workflowsParameters = {},\n iceServers,\n processingTimeout,\n requestedPlan,\n requestedRegion,\n realtimeProcessing = true,\n rtspUrl\n } = config as any;\n\n // Build workflow_configuration based on what's provided\n const workflowConfiguration: any = {\n type: \"WorkflowConfiguration\",\n image_input_name: imageInputName,\n workflows_parameters: workflowsParameters,\n workflows_thread_pool_workers: threadPoolWorkers,\n cancel_thread_pool_tasks_on_exit: true,\n video_metadata_input_name: \"video_metadata\"\n };\n\n if (hasWorkflowSpec) {\n workflowConfiguration.workflow_specification = workflowSpec;\n } else {\n workflowConfiguration.workspace_name = workspaceName;\n workflowConfiguration.workflow_id = workflowId;\n }\n\n const payload: Record<string, any> = {\n workflow_configuration: workflowConfiguration,\n api_key: this.apiKey,\n webrtc_realtime_processing: realtimeProcessing,\n webrtc_offer: {\n sdp: offer.sdp,\n type: offer.type\n },\n webrtc_config: iceServers ? { iceServers } : null,\n stream_output: streamOutputNames,\n data_output: dataOutputNames\n };\n\n // Add serverless-specific fields if provided\n if (processingTimeout !== undefined) {\n payload.processing_timeout = processingTimeout;\n }\n if (requestedPlan !== undefined) {\n payload.requested_plan = requestedPlan;\n }\n if (requestedRegion !== undefined) {\n payload.requested_region = requestedRegion;\n }\n // Add RTSP URL for server-side video capture\n if (rtspUrl) {\n payload.rtsp_url = rtspUrl;\n }\n const response = await fetch(`${this.serverUrl}/initialise_webrtc_worker`, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify(payload)\n });\n\n if (!response.ok) {\n const errorText = await response.text().catch(() => \"\");\n throw new Error(`initialise_webrtc_worker failed (${response.status}): ${errorText}`);\n }\n\n const result = await response.json();\n\n return result;\n }\n\n async terminatePipeline({ pipelineId }: { pipelineId: string }): Promise<void> {\n if (!pipelineId) {\n throw new Error(\"pipelineId is required\");\n }\n\n await fetch(\n `${this.serverUrl}/inference_pipelines/${pipelineId}/terminate?api_key=${this.apiKey}`,\n {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" }\n }\n );\n }\n\n /\n Fetch TURN server configuration from Roboflow API\n \n This automatically fetches TURN server credentials for improved WebRTC\n * connectivity through firewalls and NAT. Only applicable when using\n * Roboflow serverless infrastructure.\n \n @returns Promise resolving to ICE server configuration, or null if not applicable\n \n @example\n * ```typescript\n * const client = InferenceHTTPClient.init({ apiKey: \"your-api-key\" });\n * const iceServers = await client.fetchTurnConfig();\n * // Returns: [{ urls: [\"turn:...\"], username: \"...\", credential: \"...\" }]\n * ```\n /\n async fetchTurnConfig(): Promise<RTCIceServerConfig[] \| null> {\n // // Only fetch TURN config for Roboflow serverless URLs\n if (!ROBOFLOW_SERVERLESS_URLS.includes(this.serverUrl)) {\n return null;\n }\n try {\n const response = await fetch(\n `${RF_API_BASE_URL}/webrtc_turn_config?api_key=${this.apiKey}`,\n {\n method: \"GET\",\n headers: { \"Content-Type\": \"application/json\" }\n }\n );\n\n if (!response.ok) {\n console.warn(`[RFWebRTC] Failed to fetch TURN config (${response.status}), using defaults`);\n return null;\n }\n\n const turnConfig = await response.json();\n\n // Handle 3 formats:\n // 1. Single server object: { urls, username, credential }\n // 2. Array of servers: [{ urls, username, credential }, ...]\n // 3. Object with iceServers: { iceServers: [...] }\n let iceServersRaw: any[];\n\n if (Array.isArray(turnConfig)) {\n // Format 2: array of servers\n iceServersRaw = turnConfig;\n } else if (turnConfig.iceServers && Array.isArray(turnConfig.iceServers)) {\n // Format 3: object with iceServers array\n iceServersRaw = turnConfig.iceServers;\n } else if (turnConfig.urls) {\n // Format 1: single server object - wrap in array\n iceServersRaw = [turnConfig];\n } else {\n console.warn(\"[RFWebRTC] Invalid TURN config format, using defaults\");\n return null;\n }\n\n // Normalize the ICE servers format\n const iceServers: RTCIceServerConfig[] = iceServersRaw.map((server: any) => ({\n urls: Array.isArray(server.urls) ? server.urls : [server.urls],\n username: server.username,\n credential: server.credential\n }));\n\n return iceServers;\n } catch (err) {\n console.warn(\"[RFWebRTC] Error fetching TURN config:\", err);\n return null;\n }\n }\n}\n\n/\n Connectors for establishing WebRTC connections to Roboflow\n /\nexport const connectors = {\n /\n Create a connector that uses API key directly\n \n WARNING: If you use this in the frontend, it will expose your API key. \n * Use only for demos/testing.\n * For production, use withProxyUrl() with a backend proxy.\n \n @param apiKey - Roboflow API key\n * @param options - Additional options\n * @param options.serverUrl - Custom Roboflow server URL\n * @returns Connector with connectWrtc method\n \n @example\n * ```typescript\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const answer = await connector.connectWrtc(offer, wrtcParams);\n * ```\n /\n withApiKey(apiKey: string, options: { serverUrl?: string } = {}): Connector {\n const { serverUrl } = options;\n\n // Warn if running in browser context\n if (typeof window !== 'undefined') {\n console.warn(\n '[Security Warning] Using API key directly in browser will expose it. ' +\n 'Use connectors.withProxyUrl() for production. ' +\n 'See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key'\n );\n }\n\n const client = InferenceHTTPClient.init({ apiKey, serverUrl });\n\n return {\n connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n console.debug(\"wrtcParams\", wrtcParams);\n const answer = await client.initializeWebrtcWorker({\n offer,\n workflowSpec: wrtcParams.workflowSpec,\n workspaceName: wrtcParams.workspaceName,\n workflowId: wrtcParams.workflowId,\n config: {\n imageInputName: wrtcParams.imageInputName,\n streamOutputNames: wrtcParams.streamOutputNames,\n dataOutputNames: wrtcParams.dataOutputNames,\n threadPoolWorkers: wrtcParams.threadPoolWorkers,\n workflowsParameters: wrtcParams.workflowsParameters,\n iceServers: wrtcParams.iceServers,\n processingTimeout: wrtcParams.processingTimeout,\n requestedPlan: wrtcParams.requestedPlan,\n requestedRegion: wrtcParams.requestedRegion,\n realtimeProcessing: wrtcParams.realtimeProcessing,\n rtspUrl: wrtcParams.rtspUrl\n }\n });\n\n return answer;\n },\n\n /\n Fetch TURN server configuration for improved WebRTC connectivity\n /\n getIceServers: async (): Promise<RTCIceServerConfig[] \| null> => {\n return await client.fetchTurnConfig();\n },\n\n // Store apiKey for cleanup\n _apiKey: apiKey,\n _serverUrl: serverUrl\n };\n },\n\n /\n Create a connector that uses a backend proxy (recommended for production)\n \n Your backend receives the offer and wrtcParams, adds the secret API key,\n * and forwards to Roboflow. This keeps your API key secure.\n \n For improved WebRTC connectivity through firewalls, implement a separate\n * endpoint for TURN server configuration that calls `fetchTurnConfig()`.\n \n @param proxyUrl - Backend proxy endpoint URL for WebRTC initialization\n * @param options - Additional options\n * @param options.turnConfigUrl - Optional URL for fetching TURN server configuration\n * @returns Connector with connectWrtc and optional getIceServers methods\n \n @example\n * ```typescript\n * // Frontend: Create connector with TURN config endpoint\n * const connector = connectors.withProxyUrl('/api/init-webrtc', {\n * turnConfigUrl: '/api/turn-config'\n * });\n * ```\n \n @example\n * Backend implementation (Express) with TURN server support:\n * ```typescript\n * // Endpoint for TURN configuration (called first by SDK)\n * app.get('/api/turn-config', async (req, res) => {\n * const client = InferenceHTTPClient.init({\n * apiKey: process.env.ROBOFLOW_API_KEY\n * });\n * const iceServers = await client.fetchTurnConfig();\n * res.json({ iceServers });\n * });\n \n // Endpoint for WebRTC initialization\n * app.post('/api/init-webrtc', async (req, res) => {\n * const { offer, wrtcParams } = req.body;\n * const client = InferenceHTTPClient.init({\n * apiKey: process.env.ROBOFLOW_API_KEY\n * });\n \n const answer = await client.initializeWebrtcWorker({\n * offer,\n * workflowSpec: wrtcParams.workflowSpec,\n * workspaceName: wrtcParams.workspaceName,\n * workflowId: wrtcParams.workflowId,\n * config: {\n * imageInputName: wrtcParams.imageInputName,\n * streamOutputNames: wrtcParams.streamOutputNames,\n * dataOutputNames: wrtcParams.dataOutputNames,\n * threadPoolWorkers: wrtcParams.threadPoolWorkers,\n * workflowsParameters: wrtcParams.workflowsParameters,\n * iceServers: wrtcParams.iceServers,\n * processingTimeout: wrtcParams.processingTimeout,\n * requestedPlan: wrtcParams.requestedPlan,\n * requestedRegion: wrtcParams.requestedRegion\n * }\n * });\n \n res.json(answer);\n * });\n * ```\n /\n withProxyUrl(proxyUrl: string, options: { turnConfigUrl?: string } = {}): Connector {\n const { turnConfigUrl } = options;\n\n return {\n connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n const response = await fetch(proxyUrl, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify({\n offer,\n wrtcParams\n })\n });\n\n if (!response.ok) {\n const errorText = await response.text().catch(() => \"\");\n throw new Error(`Proxy request failed (${response.status}): ${errorText}`);\n }\n\n return await response.json();\n },\n\n /\n Fetch TURN server configuration from the proxy backend\n * Only available if turnConfigUrl was provided\n /\n getIceServers: turnConfigUrl\n ? async (): Promise<RTCIceServerConfig[] \| null> => {\n try {\n const response = await fetch(turnConfigUrl, {\n method: \"GET\",\n headers: { \"Content-Type\": \"application/json\" }\n });\n\n if (!response.ok) {\n console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${response.status})`);\n return null;\n }\n\n const data = await response.json();\n return data.iceServers \|\| null;\n } catch (err) {\n console.warn(\"[RFWebRTC] Error fetching TURN config from proxy:\", err);\n return null;\n }\n }\n : undefined\n };\n }\n};\n","/\n Get a camera stream with the given constraints.\n \n @param constraints - MediaStreamConstraints for getUserMedia\n * @returns Promise that resolves to MediaStream\n \n @example\n * ```typescript\n * const stream = await useCamera({\n * video: {\n * facingMode: { ideal: \"user\" },\n * width: { ideal: 1280 },\n * height: { ideal: 720 },\n * frameRate: { ideal: 30 }\n * },\n * audio: false\n * });\n * ```\n /\nexport async function useCamera(constraints: MediaStreamConstraints = { video: true }): Promise<MediaStream> {\n try {\n console.log(\"[RFStreams] requesting with\", constraints);\n const stream = await navigator.mediaDevices.getUserMedia(constraints);\n console.log(\"[RFStreams] got stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n return stream;\n } catch (err) {\n console.warn(\"[RFStreams] failed, falling back\", err);\n const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: false });\n console.log(\"[RFStreams] fallback stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n return stream;\n }\n}\n \nexport function stopStream(stream: MediaStream \| null \| undefined): void {\n if (stream) {\n stream.getTracks().forEach(track => track.stop());\n console.log(\"[RFStreams] Stream stopped\");\n }\n}\n","/\n Video file upload via WebRTC datachannel\n \n This module provides the FileUploader class for chunked file uploads\n * through WebRTC datachannels with backpressure handling.\n /\n\n/\n Configuration constants for file upload (matching Python SDK)\n /\nconst CHUNK_SIZE = 49152; // 49KB - safe for WebRTC\nconst BUFFER_LIMIT = 262144; // 256KB - backpressure threshold\nconst POLL_INTERVAL = 10; // 10ms buffer check interval\n\n/\n Helper to sleep for a given number of milliseconds\n /\nfunction sleep(ms: number): Promise<void> {\n return new Promise(resolve => setTimeout(resolve, ms));\n}\n\n/\n FileUploader handles chunked file upload with backpressure\n \n Uploads files through a WebRTC datachannel in 49KB chunks with\n * intelligent backpressure handling to prevent overwhelming the network.\n /\nexport class FileUploader {\n private file: File;\n private channel: RTCDataChannel;\n private totalChunks: number;\n private cancelled: boolean = false;\n\n constructor(file: File, channel: RTCDataChannel) {\n this.file = file;\n this.channel = channel;\n this.totalChunks = Math.ceil(file.size / CHUNK_SIZE);\n }\n\n /\n Cancel the upload\n /\n cancel(): void {\n this.cancelled = true;\n }\n\n /\n Upload the file in chunks with backpressure handling\n \n @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n /\n async upload(onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n const totalBytes = this.file.size;\n\n for (let chunkIndex = 0; chunkIndex < this.totalChunks; chunkIndex++) {\n // Check for cancellation\n if (this.cancelled) {\n throw new Error(\"Upload cancelled\");\n }\n\n // Check channel state\n if (this.channel.readyState !== \"open\") {\n throw new Error(\"Video upload interrupted\");\n }\n\n // Read chunk from file\n const start = chunkIndex CHUNK_SIZE;\n const end = Math.min(start + CHUNK_SIZE, totalBytes);\n const chunkBlob = this.file.slice(start, end);\n const chunkData = new Uint8Array(await chunkBlob.arrayBuffer());\n\n // Create message with 8-byte header (chunkIndex + totalChunks as uint32 LE)\n const message = new ArrayBuffer(8 + chunkData.length);\n const view = new DataView(message);\n view.setUint32(0, chunkIndex, true); // little-endian\n view.setUint32(4, this.totalChunks, true); // little-endian\n new Uint8Array(message, 8).set(chunkData);\n\n // Backpressure: wait for buffer to drain\n while (this.channel.bufferedAmount > BUFFER_LIMIT) {\n if (this.channel.readyState !== \"open\") {\n throw new Error(\"Video upload interrupted\");\n }\n await sleep(POLL_INTERVAL);\n }\n\n // Send chunk\n this.channel.send(message);\n\n // Report progress\n if (onProgress) {\n onProgress(end, totalBytes);\n }\n }\n }\n}\n","\nimport { InferenceHTTPClient, Connector, WebRTCParams, RTCIceServerConfig } from \"./inference-api\";\nimport { stopStream } from \"./streams\";\nimport { WebRTCOutputData } from \"./webrtc-types\";\nimport { FileUploader } from \"./video-upload\";\n\n// Re-export shared types\nexport type { WebRTCVideoMetadata, WebRTCOutputData } from \"./webrtc-types\";\n\n// Re-export FileUploader from video-upload\nexport { FileUploader } from \"./video-upload\";\n\n/*\n Binary protocol header size (frame_id + chunk_index + total_chunks)\n * Each field is 4 bytes uint32 little-endian\n /\nconst HEADER_SIZE = 12;\n\n/\n Reassembles chunked binary messages from the datachannel\n /\nexport class ChunkReassembler {\n private pendingFrames: Map<number, {\n chunks: Map<number, Uint8Array>;\n totalChunks: number;\n }> = new Map();\n\n /\n Process an incoming chunk and return the complete message if all chunks received\n /\n processChunk(frameId: number, chunkIndex: number, totalChunks: number, payload: Uint8Array): Uint8Array \| null {\n // Single chunk message - return immediately\n if (totalChunks === 1) {\n return payload;\n }\n\n // Multi-chunk message - accumulate\n if (!this.pendingFrames.has(frameId)) {\n this.pendingFrames.set(frameId, {\n chunks: new Map(),\n totalChunks\n });\n }\n\n const frame = this.pendingFrames.get(frameId)!;\n frame.chunks.set(chunkIndex, payload);\n\n // Check if all chunks received\n if (frame.chunks.size === totalChunks) {\n // Reassemble in order\n const totalLength = Array.from(frame.chunks.values()).reduce((sum, chunk) => sum + chunk.length, 0);\n const result = new Uint8Array(totalLength);\n let offset = 0;\n\n for (let i = 0; i < totalChunks; i++) {\n const chunk = frame.chunks.get(i)!;\n result.set(chunk, offset);\n offset += chunk.length;\n }\n\n this.pendingFrames.delete(frameId);\n return result;\n }\n\n return null;\n }\n\n /\n Clear all pending frames (for cleanup)\n /\n clear(): void {\n this.pendingFrames.clear();\n }\n}\n\n/\n Parse the binary header from a datachannel message\n /\nexport function parseBinaryHeader(buffer: ArrayBuffer): { frameId: number; chunkIndex: number; totalChunks: number; payload: Uint8Array } {\n const view = new DataView(buffer);\n const frameId = view.getUint32(0, true); // little-endian\n const chunkIndex = view.getUint32(4, true); // little-endian\n const totalChunks = view.getUint32(8, true); // little-endian\n const payload = new Uint8Array(buffer, HEADER_SIZE);\n\n return { frameId, chunkIndex, totalChunks, payload };\n}\n\nexport interface UseStreamOptions {\n disableInputStreamDownscaling?: boolean;\n}\n\nexport interface UseStreamParams {\n source: MediaStream;\n connector: Connector;\n wrtcParams: WebRTCParams;\n onData?: (data: WebRTCOutputData) => void;\n options?: UseStreamOptions;\n}\n\nasync function waitForIceGathering(pc: RTCPeerConnection, timeoutMs = 6000): Promise<void> {\n if (pc.iceGatheringState === \"complete\") return;\n\n let hasSrflx = false;\n\n // Track if we get a good candidate (srflx = public IP via STUN)\n const candidateHandler = (event: RTCPeerConnectionIceEvent) => {\n if (event.candidate && event.candidate.type === \"srflx\") {\n hasSrflx = true;\n }\n };\n pc.addEventListener(\"icecandidate\", candidateHandler);\n\n try {\n await Promise.race([\n new Promise<void>(resolve => {\n const check = () => {\n if (pc.iceGatheringState === \"complete\") {\n pc.removeEventListener(\"icegatheringstatechange\", check);\n resolve();\n }\n };\n pc.addEventListener(\"icegatheringstatechange\", check);\n }),\n new Promise<void>((resolve, reject) => {\n setTimeout(() => {\n if (!hasSrflx) {\n console.error(\"[ICE] timeout with NO srflx candidate! Connection may fail.\");\n reject(new Error(\"ICE gathering timeout without srflx candidate\"));\n } else {\n resolve();\n }\n }, timeoutMs);\n })\n ]);\n } finally {\n pc.removeEventListener(\"icecandidate\", candidateHandler);\n }\n}\n\nfunction setupRemoteStreamListener(pc: RTCPeerConnection): Promise<MediaStream> {\n return new Promise((resolve) => {\n pc.addEventListener(\"track\", (event: RTCTrackEvent) => {\n if (event.streams && event.streams[0]) {\n resolve(event.streams[0]);\n }\n });\n });\n}\n\nconst DEFAULT_ICE_SERVERS: RTCIceServerConfig[] = [\n { urls: [\"stun:stun.l.google.com:19302\"] }\n];\n\nasync function preparePeerConnection(\n localStream?: MediaStream,\n file?: File,\n customIceServers?: RTCIceServerConfig[],\n rtspUrl?: string\n): Promise<{\n pc: RTCPeerConnection;\n offer: RTCSessionDescriptionInit;\n remoteStreamPromise: Promise<MediaStream>;\n dataChannel: RTCDataChannel;\n uploadChannel?: RTCDataChannel;\n}> {\n // Validate: exactly one source type must be provided\n const hasLocalStream = !!localStream;\n const hasFile = !!file;\n const hasRtspUrl = !!rtspUrl;\n const sourceCount = [hasLocalStream, hasFile, hasRtspUrl].filter(Boolean).length;\n\n if (sourceCount !== 1) {\n throw new Error(\"Exactly one of localStream, file, or rtspUrl must be provided\");\n }\n\n const iceServers = customIceServers ?? DEFAULT_ICE_SERVERS;\n\n const pc = new RTCPeerConnection({\n iceServers: iceServers as RTCIceServer[]\n });\n\n // Add transceiver for receiving remote video (BEFORE adding tracks - order matters!)\n try {\n pc.addTransceiver(\"video\", { direction: \"recvonly\" });\n } catch (err) {\n console.warn(\"[RFWebRTC] Could not add transceiver:\", err);\n }\n\n if (localStream) {\n // Add local tracks\n localStream.getVideoTracks().forEach(track => {\n try {\n // @ts-ignore - contentHint is not in all TypeScript definitions\n track.contentHint = \"detail\";\n } catch (e) {\n // Ignore if contentHint not supported\n }\n pc.addTrack(track, localStream);\n });\n }\n // Note: For RTSP, no local tracks are added (receive-only mode)\n\n // Setup remote stream listener\n const remoteStreamPromise = setupRemoteStreamListener(pc);\n\n // Create control datachannel (named \"inference\" to match Python SDK)\n const dataChannel = pc.createDataChannel(\"inference\", {\n ordered: true\n });\n\n // Create upload datachannel for file uploads (not needed for RTSP)\n let uploadChannel: RTCDataChannel \| undefined;\n if (file) {\n uploadChannel = pc.createDataChannel(\"video_upload\");\n }\n\n // Create offer\n const offer = await pc.createOffer();\n await pc.setLocalDescription(offer);\n\n // Wait for ICE gathering\n await waitForIceGathering(pc);\n\n return {\n pc,\n offer: pc.localDescription!,\n remoteStreamPromise,\n dataChannel,\n uploadChannel\n };\n}\n\n/\n Disable input stream downscaling\n * @private\n /\nasync function disableInputStreamDownscaling(pc: RTCPeerConnection): Promise<void> {\n const sender = pc.getSenders().find(s => s.track && s.track.kind === \"video\");\n if (!sender) return;\n\n const params = sender.getParameters();\n params.encodings = params.encodings \|\| [{}];\n params.encodings[0].scaleResolutionDownBy = 1;\n\n try {\n await sender.setParameters(params);\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to set encoding parameters:\", err);\n }\n}\n\n/\n Helper to wait for datachannel to open\n /\nfunction waitForChannelOpen(channel: RTCDataChannel, timeoutMs = 30000): Promise<void> {\n return new Promise((resolve, reject) => {\n if (channel.readyState === \"open\") {\n resolve();\n return;\n }\n\n const openHandler = () => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n clearTimeout(timeout);\n resolve();\n };\n\n const errorHandler = () => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n clearTimeout(timeout);\n reject(new Error(\"Datachannel error\"));\n };\n\n const timeout = setTimeout(() => {\n channel.removeEventListener(\"open\", openHandler);\n channel.removeEventListener(\"error\", errorHandler);\n reject(new Error(\"Datachannel open timeout\"));\n }, timeoutMs);\n\n channel.addEventListener(\"open\", openHandler);\n channel.addEventListener(\"error\", errorHandler);\n });\n}\n\n/\n WebRTC Connection object\n \n Represents an active WebRTC connection to Roboflow for streaming inference\n * or file-based batch processing.\n /\nexport class RFWebRTCConnection {\n private pc: RTCPeerConnection;\n private _localStream?: MediaStream;\n private remoteStreamPromise: Promise<MediaStream>;\n private pipelineId: string \| null;\n private apiKey: string \| null;\n private dataChannel: RTCDataChannel;\n private reassembler: ChunkReassembler;\n private uploadChannel?: RTCDataChannel;\n private uploader?: FileUploader;\n private onComplete?: () => void;\n\n /* @private /\n constructor(\n pc: RTCPeerConnection,\n remoteStreamPromise: Promise<MediaStream>,\n pipelineId: string \| null,\n apiKey: string \| null,\n dataChannel: RTCDataChannel,\n options?: {\n localStream?: MediaStream;\n uploadChannel?: RTCDataChannel;\n onData?: (data: any) => void;\n onComplete?: () => void;\n }\n ) {\n this.pc = pc;\n this._localStream = options?.localStream;\n this.remoteStreamPromise = remoteStreamPromise;\n this.pipelineId = pipelineId;\n this.apiKey = apiKey;\n this.dataChannel = dataChannel;\n this.reassembler = new ChunkReassembler();\n this.uploadChannel = options?.uploadChannel;\n this.onComplete = options?.onComplete;\n\n // Set binary mode for datachannel\n this.dataChannel.binaryType = \"arraybuffer\";\n\n const onData = options?.onData;\n\n // Setup data channel event listeners\n if (onData) {\n this.dataChannel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n try {\n // Handle binary protocol with chunking\n if (messageEvent.data instanceof ArrayBuffer) {\n const { frameId, chunkIndex, totalChunks, payload } = parseBinaryHeader(messageEvent.data);\n const completePayload = this.reassembler.processChunk(frameId, chunkIndex, totalChunks, payload);\n\n if (completePayload) {\n // Decode UTF-8 JSON payload\n const decoder = new TextDecoder(\"utf-8\");\n const jsonString = decoder.decode(completePayload);\n const data = JSON.parse(jsonString);\n onData(data);\n }\n } else {\n // Fallback for string messages (shouldn't happen with new protocol)\n const data = JSON.parse(messageEvent.data);\n onData(data);\n }\n } catch (err) {\n console.error(\"[RFWebRTC] Failed to parse data channel message:\", err);\n }\n });\n\n this.dataChannel.addEventListener(\"error\", (error) => {\n console.error(\"[RFWebRTC] Data channel error:\", error);\n });\n }\n\n // Handle channel close - call onComplete when processing finishes\n this.dataChannel.addEventListener(\"close\", () => {\n this.reassembler.clear();\n if (this.onComplete) {\n this.onComplete();\n }\n });\n }\n\n /\n Get the remote stream (processed video from Roboflow)\n \n @returns Promise resolving to the remote MediaStream\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n /\n async remoteStream(): Promise<MediaStream> {\n return await this.remoteStreamPromise;\n }\n\n /\n Get the local stream (original camera)\n \n @returns The local MediaStream, or undefined if using file upload mode\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * const localStream = conn.localStream();\n * if (localStream) {\n * videoElement.srcObject = localStream;\n * }\n * ```\n /\n localStream(): MediaStream \| undefined {\n return this._localStream;\n }\n\n /\n Cleanup and close connection\n \n Terminates the pipeline on Roboflow, closes the peer connection,\n * and stops the local media stream (if applicable).\n \n @returns Promise that resolves when cleanup is complete\n \n @example\n * ```typescript\n * const conn = await useStream({ ... });\n * // ... use connection ...\n * await conn.cleanup(); // Clean up when done\n * ```\n /\n async cleanup(): Promise<void> {\n // Cancel any ongoing upload\n if (this.uploader) {\n this.uploader.cancel();\n }\n\n // Clear pending chunks\n this.reassembler.clear();\n\n // Terminate pipeline\n if (this.pipelineId && this.apiKey) {\n try {\n const client = InferenceHTTPClient.init({ apiKey: this.apiKey });\n await client.terminatePipeline({ pipelineId: this.pipelineId });\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to terminate pipeline:\", err);\n }\n }\n\n // Close peer connection\n if (this.pc && this.pc.connectionState !== \"closed\") {\n this.pc.close();\n }\n\n // Stop local stream if present\n if (this._localStream) {\n stopStream(this._localStream);\n }\n }\n\n /\n Start uploading a file through the connection\n \n @param file - The file to upload\n * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n * @returns Promise that resolves when upload is complete\n * @throws Error if no upload channel is available\n \n @example\n * ```typescript\n * await connection.startUpload(videoFile, (uploaded, total) => {\n * console.log(`Upload progress: ${(uploaded / total * 100).toFixed(1)}%`);\n * });\n * ```\n /\n async startUpload(file: File, onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n if (!this.uploadChannel) {\n throw new Error(\"No upload channel available. This connection was not created for file uploads.\");\n }\n\n // Wait for upload channel to open\n await waitForChannelOpen(this.uploadChannel);\n\n this.uploader = new FileUploader(file, this.uploadChannel);\n await this.uploader.upload(onProgress);\n }\n\n /\n Cancel any ongoing file upload\n /\n cancelUpload(): void {\n if (this.uploader) {\n this.uploader.cancel();\n }\n }\n\n /\n Reconfigure pipeline outputs at runtime\n \n Dynamically change stream and data outputs without restarting the connection.\n * Set a field to `null` to leave it unchanged, or to `null` value to enable all outputs,\n * or to `[]` to disable/auto-detect.\n \n @param config - Output configuration\n * @param config.streamOutput - Stream output names (null = unchanged, [] = auto-detect, [\"name\"] = specific output)\n * @param config.dataOutput - Data output names (null = unchanged, [] = disable, [\"name\"] = specific outputs, null value = all outputs)\n \n @example\n * ```typescript\n * // Change to different stream output\n * connection.reconfigureOutputs({\n * streamOutput: [\"annotated_image\"],\n * dataOutput: null // unchanged\n * });\n \n // Enable all data outputs\n * connection.reconfigureOutputs({\n * streamOutput: null, // unchanged\n * dataOutput: null // null value = all outputs\n * });\n \n // Disable all data outputs\n * connection.reconfigureOutputs({\n * streamOutput: null, // unchanged\n * dataOutput: [] // empty array = disable\n * });\n * ```\n /\n reconfigureOutputs(config: { streamOutput?: string[] \| null; dataOutput?: string[] \| null }): void {\n const message: any = {};\n\n if (config.streamOutput !== undefined) {\n message.stream_output = config.streamOutput;\n }\n\n if (config.dataOutput !== undefined) {\n message.data_output = config.dataOutput;\n }\n\n this.sendData(message);\n }\n\n /\n Send data through the data channel\n * @private\n /\n private sendData(data: any): void {\n if (this.dataChannel.readyState !== \"open\") {\n console.warn(\"[RFWebRTC] Data channel is not open. Current state:\", this.dataChannel.readyState);\n return;\n }\n\n try {\n const message = typeof data === \"string\" ? data : JSON.stringify(data);\n this.dataChannel.send(message);\n } catch (err) {\n console.error(\"[RFWebRTC] Failed to send data:\", err);\n }\n }\n}\n\n/\n Internal base function for establishing WebRTC connection\n * Used by both useStream and useVideoFile\n * @private\n /\ninterface BaseUseStreamParams {\n source?: MediaStream \| File;\n rtspUrl?: string;\n connector: Connector;\n wrtcParams: WebRTCParams;\n onData?: (data: WebRTCOutputData) => void;\n onComplete?: () => void;\n onFileUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n options?: UseStreamOptions;\n}\n\nasync function baseUseStream({\n source,\n rtspUrl,\n connector,\n wrtcParams,\n onData,\n onComplete,\n onFileUploadProgress,\n options = {}\n}: BaseUseStreamParams): Promise<RFWebRTCConnection> {\n // Validate connector\n if (!connector \|\| typeof connector.connectWrtc !== \"function\") {\n throw new Error(\"connector must have a connectWrtc method\");\n }\n\n // Determine source type\n const isRtsp = !!rtspUrl;\n const isFile = !isRtsp && source instanceof File;\n const localStream = !isRtsp && !isFile && source ? (source as MediaStream) : undefined;\n const file = isFile ? (source as File) : undefined;\n\n // Step 1: Determine ICE servers to use\n // Priority: 1) User-provided in wrtcParams, 2) From connector.getIceServers(), 3) Defaults\n let iceServers = wrtcParams.iceServers;\n if ((!iceServers \|\| iceServers.length === 0) && connector.getIceServers) {\n try {\n const turnConfig = await connector.getIceServers();\n if (turnConfig && turnConfig.length > 0) {\n iceServers = turnConfig;\n console.log(\"[RFWebRTC] Using TURN servers from connector\");\n }\n } catch (err) {\n console.warn(\"[RFWebRTC] Failed to fetch TURN config, using defaults:\", err);\n }\n }\n\n // Step 2: Prepare peer connection and create offer\n const { pc, offer, remoteStreamPromise, dataChannel, uploadChannel } = await preparePeerConnection(\n localStream,\n file,\n iceServers,\n rtspUrl\n );\n\n // Update wrtcParams with resolved iceServers so server also uses them\n // For file uploads, default to batch mode (realtimeProcessing: false)\n // For RTSP, default to realtime processing\n const resolvedWrtcParams = {\n ...wrtcParams,\n iceServers: iceServers,\n realtimeProcessing: wrtcParams.realtimeProcessing ?? !isFile,\n rtspUrl: rtspUrl\n };\n\n // Step 3: Call connector.connectWrtc to exchange SDP and get answer\n const answer = await connector.connectWrtc(\n { sdp: offer.sdp!, type: offer.type! },\n resolvedWrtcParams\n );\n\n // API returns sdp and type at root level\n const sdpAnswer = { sdp: answer.sdp, type: answer.type } as RTCSessionDescriptionInit;\n\n if (!sdpAnswer?.sdp \|\| !sdpAnswer?.type) {\n console.error(\"[RFWebRTC] Invalid answer from server:\", answer);\n throw new Error(\"connector.connectWrtc must return answer with sdp and type\");\n }\n\n const pipelineId = answer?.context?.pipeline_id \|\| null;\n\n // Step 4: Set remote description\n await pc.setRemoteDescription(sdpAnswer);\n\n // Step 5: Wait for connection to establish\n await new Promise<void>((resolve, reject) => {\n const checkState = () => {\n if (pc.connectionState === \"connected\") {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n resolve();\n } else if (pc.connectionState === \"failed\") {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n reject(new Error(\"WebRTC connection failed\"));\n }\n };\n\n pc.addEventListener(\"connectionstatechange\", checkState);\n checkState(); // Check immediately in case already connected\n\n // Timeout after 30 seconds\n setTimeout(() => {\n pc.removeEventListener(\"connectionstatechange\", checkState);\n reject(new Error(\"WebRTC connection timeout after 30s\"));\n }, 30000);\n });\n\n // Step 6: Optimize quality for MediaStream (disable downsampling by default)\n if (localStream) {\n const shouldDisableDownscaling = options.disableInputStreamDownscaling !== false;\n if (shouldDisableDownscaling) {\n await disableInputStreamDownscaling(pc);\n }\n }\n\n // Get apiKey from connector if available (for cleanup)\n const apiKey = connector._apiKey \|\| null;\n\n // Step 7: Create connection object\n const connection = new RFWebRTCConnection(\n pc,\n remoteStreamPromise,\n pipelineId,\n apiKey,\n dataChannel,\n {\n localStream,\n uploadChannel,\n onData,\n onComplete\n }\n );\n\n // Step 8: Start file upload if applicable (runs in background)\n if (file && uploadChannel) {\n connection.startUpload(file, onFileUploadProgress).catch(err => {\n console.error(\"[RFWebRTC] Upload error:\", err);\n });\n }\n\n return connection;\n}\n\n/\n Main function to establish WebRTC streaming connection\n \n Creates a WebRTC connection to Roboflow for real-time inference on video streams.\n \n @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n \n @example\n * ```typescript\n * import { useStream } from 'inferencejs/webrtc';\n * import { connectors } from 'inferencejs/api';\n * import { useCamera } from 'inferencejs/streams';\n \n const connector = connectors.withApiKey(\"your-api-key\");\n * const stream = await useCamera({ video: { facingMode: { ideal: \"environment\" } } });\n * const conn = await useStream({\n * source: stream,\n * connector,\n * wrtcParams: {\n * workflowSpec: {\n * // Your workflow specification\n * },\n * imageInputName: \"image\",\n * streamOutputNames: [\"output\"],\n * dataOutputNames: [\"predictions\"]\n * },\n * onData: (data) => {\n * console.log(\"Inference results:\", data);\n * }\n * });\n \n const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n /\nexport async function useStream({\n source,\n connector,\n wrtcParams,\n onData,\n options = {}\n}: UseStreamParams): Promise<RFWebRTCConnection> {\n if (source instanceof File) {\n throw new Error(\"useStream requires a MediaStream. Use useVideoFile for File uploads.\");\n }\n\n return baseUseStream({\n source,\n connector,\n wrtcParams,\n onData,\n options\n });\n}\n\n/\n Parameters for useVideoFile function\n /\nexport interface UseVideoFileParams {\n /* The video file to upload /\n file: File;\n /* Connector for WebRTC signaling /\n connector: Connector;\n /* WebRTC parameters for the workflow /\n wrtcParams: WebRTCParams;\n /* Callback for inference results /\n onData?: (data: WebRTCOutputData) => void;\n /* Callback for upload progress /\n onUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n /* Callback when processing completes (datachannel closes) /\n onComplete?: () => void;\n}\n\n/\n Upload a video file for batch inference processing\n \n Creates a WebRTC connection to Roboflow for uploading a video file\n * and receiving inference results. The file is uploaded via datachannel\n * with intelligent backpressure handling.\n \n @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n \n @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n \n const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useVideoFile({\n * file: videoFile,\n * connector,\n * wrtcParams: {\n * workflowSpec: { ... },\n * imageInputName: \"image\",\n * dataOutputNames: [\"predictions\"]\n * },\n * onData: (data) => {\n * console.log(\"Inference results:\", data);\n * if (data.processing_complete) {\n * console.log(\"Processing complete!\");\n * }\n * },\n * onUploadProgress: (uploaded, total) => {\n * console.log(`Upload: ${(uploaded / total * 100).toFixed(1)}%`);\n * }\n * });\n \n // When done\n * await connection.cleanup();\n * ```\n /\nexport async function useVideoFile({\n file,\n connector,\n wrtcParams,\n onData,\n onUploadProgress,\n onComplete\n}: UseVideoFileParams): Promise<RFWebRTCConnection> {\n return baseUseStream({\n source: file,\n connector,\n wrtcParams: {\n ...wrtcParams,\n realtimeProcessing: wrtcParams.realtimeProcessing ?? true\n },\n onData,\n onComplete,\n onFileUploadProgress: onUploadProgress\n });\n}\n\n/\n Parameters for useRtspStream function\n /\nexport interface UseRtspStreamParams {\n /* RTSP URL for server-side video capture (e.g., \"rtsp://camera.local/stream\") /\n rtspUrl: string;\n /* Connector for WebRTC signaling /\n connector: Connector;\n /* WebRTC parameters for the workflow /\n wrtcParams: WebRTCParams;\n /* Callback for inference results /\n onData?: (data: WebRTCOutputData) => void;\n}\n\n/\n Connect to an RTSP stream for inference processing\n \n Creates a WebRTC connection where the server captures video from an RTSP URL\n * and sends processed video back to the client. This is a receive-only mode -\n * no video is sent from the browser to the server.\n \n @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n \n @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n \n const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useRtspStream({\n * rtspUrl: \"rtsp://camera.local/stream\",\n * connector,\n * wrtcParams: {\n * workflowSpec: { ... },\n * imageInputName: \"image\",\n * dataOutputNames: [\"predictions\"]\n * },\n * onData: (data) => {\n * console.log(\"Inference results:\", data);\n * }\n * });\n \n // Get processed video stream from server\n * const remoteStream = await connection.remoteStream();\n * videoElement.srcObject = remoteStream;\n \n // When done\n * await connection.cleanup();\n * ```\n */\nexport async function useRtspStream({\n rtspUrl,\n connector,\n wrtcParams,\n onData\n}: UseRtspStreamParams): Promise<RFWebRTCConnection> {\n // Validate RTSP URL format\n if (!rtspUrl.startsWith(\"rtsp://\") && !rtspUrl.startsWith(\"rtsps://\")) {\n throw new Error(\"Invalid RTSP URL: must start with rtsp:// or rtsps://\");\n }\n\n return baseUseStream({\n rtspUrl,\n connector,\n wrtcParams,\n onData\n 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package/dist/inference-api.d.ts CHANGED Viewed

@@ -32,6 +32,13 @@ export interface WebRTCWorkerConfig {
      * @default true
      */
     realtimeProcessing?: boolean;
+    /**
+     * RTSP URL for server-side video capture.
+     * When provided, the server captures video from this RTSP stream instead of receiving
+     * video from the client. Supports credentials in URL format: rtsp://user:pass@host/stream
+     * @example "rtsp://camera.local/stream"
+     */
+    rtspUrl?: string;
 }
 /**
  * ICE server configuration for WebRTC connections
@@ -108,6 +115,13 @@ export interface WebRTCParams {
      * @default true
      */
     realtimeProcessing?: boolean;
+    /**
+     * RTSP URL for server-side video capture.
+     * When provided, the server captures video from this RTSP stream instead of receiving
+     * video from the client. Supports credentials in URL format: rtsp://user:pass@host/stream
+     * @example "rtsp://camera.local/stream"
+     */
+    rtspUrl?: string;
 }
 export interface Connector {
     connectWrtc(offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse>;

package/dist/inference-api.d.ts.map CHANGED Viewed

	@@ -1 +1 @@
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package/dist/webrtc.d.ts CHANGED Viewed

@@ -258,4 +258,54 @@ export interface UseVideoFileParams {
  * ```
  */
 export declare function useVideoFile({ file, connector, wrtcParams, onData, onUploadProgress, onComplete }: UseVideoFileParams): Promise<RFWebRTCConnection>;
+/**
+ * Parameters for useRtspStream function
+ */
+export interface UseRtspStreamParams {
+    /** RTSP URL for server-side video capture (e.g., "rtsp://camera.local/stream") */
+    rtspUrl: string;
+    /** Connector for WebRTC signaling */
+    connector: Connector;
+    /** WebRTC parameters for the workflow */
+    wrtcParams: WebRTCParams;
+    /** Callback for inference results */
+    onData?: (data: WebRTCOutputData) => void;
+}
+/**
+ * Connect to an RTSP stream for inference processing
+ *
+ * Creates a WebRTC connection where the server captures video from an RTSP URL
+ * and sends processed video back to the client. This is a receive-only mode -
+ * no video is sent from the browser to the server.
+ *
+ * @param params - Connection parameters
+ * @returns Promise resolving to RFWebRTCConnection
+ *
+ * @example
+ * ```typescript
+ * import { connectors, webrtc } from './index.ts';
+ *
+ * const connector = connectors.withApiKey("your-api-key");
+ * const connection = await webrtc.useRtspStream({
+ *   rtspUrl: "rtsp://camera.local/stream",
+ *   connector,
+ *   wrtcParams: {
+ *     workflowSpec: { ... },
+ *     imageInputName: "image",
+ *     dataOutputNames: ["predictions"]
+ *   },
+ *   onData: (data) => {
+ *     console.log("Inference results:", data);
+ *   }
+ * });
+ *
+ * // Get processed video stream from server
+ * const remoteStream = await connection.remoteStream();
+ * videoElement.srcObject = remoteStream;
+ *
+ * // When done
+ * await connection.cleanup();
+ * ```
+ */
+export declare function useRtspStream({ rtspUrl, connector, wrtcParams, onData }: UseRtspStreamParams): Promise<RFWebRTCConnection>;
 //# sourceMappingURL=webrtc.d.ts.map

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package/package.json CHANGED Viewed

@@ -1,6 +1,6 @@
 {
   "name": "@roboflow/inference-sdk",
-  "version": "0.1.8",
+  "version": "0.1.9",
   "description": "Lightweight client for Roboflow's hosted inference API with WebRTC streaming support",
   "keywords": [
     "roboflow",