Create a new video inference stream. Returns connection details for the chosen transport.
Workflow: Create stream → connect video source → receive results on WebSocket → send keepalives → close when done.
Transport options:
source. Response includes livekit.url and livekit.token for publishing video.source.type: 'webrtc' with an SDP offer. Response includes webrtc answer and turn_servers.source.type: 'livekit' with your room URL and token.Processing modes:
target_fps, clip_length_seconds, delay_seconds for temporal analysis (motion, actions).interval_seconds for static analysis (OCR, object detection).Limits: 5 concurrent streams per API key. Requires credits.
Documentation Index
Fetch the complete documentation index at: https://docs.overshoot.ai/llms.txt
Use this file to discover all available pages before exploring further.
Provide your API key in the Authorization header as 'Bearer <api_key>'
Create a new video inference stream.
Minimal example (frame mode, native transport):
{
"processing": { "interval_seconds": 2.0 },
"inference": { "prompt": "Describe what you see", "model": "Qwen/Qwen3.5-9B" }
}Clip mode processing config. Samples frames into short video clips for temporal analysis (motion, actions, events).
Provide target_fps directly, or the legacy fps + sampling_ratio pair (not both).
What the AI model should do with the video.
Legacy WebRTC source. Send an SDP offer and receive an SDP answer in the response.
(Legacy) WebRTC offer. Use source instead.
Processing mode. Auto-detected from processing config if omitted. 'clip' for video clips, 'frame' for single images.
clip, frame Optional client metadata for request tracking.
Stream created. Use stream_id for all subsequent operations.
Returned after successfully creating a stream. Contains connection details specific to the chosen transport.
Unique stream identifier. Use this for keepalive, close, prompt update, and WebSocket connection.
SDP answer for WebRTC sources. Null for native/livekit sources.
Stream lease info including ttl_seconds. Send keepalives before TTL expires.
TURN server credentials for WebRTC sources. Null for native/livekit sources.
LiveKit connection details (url + token) for native sources. Connect to this room to publish video.