Overview

Sinks consume upstream media and produce output - streams, files, or room publications.

Available types

Type	Description
livekit	Publish to a LiveKit room
jitsi	Publish to a Jitsi conference
daily	Publish to a Daily room
agora	Publish to an Agora channel
rtmp_push	Push RTMP to CDNs
srt_push	Push MPEG-TS over SRT
whip	Push via WHIP
segment	HLS segments to object storage
image	Periodic screenshots to storage
websocket	Send raw PCM or encoded audio access units

Shape

{
  "name": "out",
  "type": "rtmp_push",
  "inputs": ["mix_video", "mix_audio"],
  "config": {
    "encoding": { "videoCodec": "h264", "audioCodec": "aac" }
  }
}

An optional select on an object-form inputs entry filters which upstream tracks the sink receives on that edge.

Encoding

Many sinks accept an encoding block (EncodingConfig):

Field	Type	Default	Description
videoCodec	string	See below	Mux encoder opt-in: h264 | vp9 | av1. WebRTC sinks use it as a codec preference and also support vp8.
audioCodec	string	See below	Mux encoder opt-in or WebRTC codec preference: aac or opus.
audioProfile	string	—	AAC only: LC | HEv1 | HEv2. Ignored for Opus.
videoBitrateBps	int	0 (auto)	Target video bitrate. 0 = derive from resolution (see below). Clamped to 0–50_000_000 (50 Mbps).
audioBitrateBps	int	0 (auto)	Target audio bitrate. Clamped to 0–512_000 (512 kbps).
keyframeIntervalSec	int	2	Keyframe (GOP) interval in seconds (max 10). GOP frame count is keyframeIntervalSec × upstream fps: video_mixer canvas.fps when wired, otherwise measured from frame timestamps before encode starts.

For rtmp_push, srt_push, and segment, each codec must be explicit. An omitted codec means that raw media type is not encoded or sent. WebRTC sinks instead negotiate an available codec when the field is omitted; use select.mediaTypes to disable an RTC audio or video track.

When videoBitrateBps is 0, the engine picks a bitrate from output width × height and codec (H.264 baseline ladder; AV1 / VP9 use lower multipliers).

For compositor output cadence, see video_mixer canvas.fps.

avflow manages output audio sample rate internally. Audio encode rates snap to {8000, 16000, 32000, 44100, 48000} Hz from upstream media (fallback 48000).

For rtmp_push, srt_push, and segment, this block is a compatibility shortcut. The compiler expands it into video_encoder and audio_encoder nodes; the sink then only muxes and writes encoded packets. A missing codec does not encode that media, even when a matching raw track is present. Equivalent generated encoders are shared, and a Job may contain at most one video encoder after expansion. RTC sinks, including WHIP, keep encoding as an SDK codec/bitrate hint and accept raw media only.

Pricing

Video sinks are billed by encoded resolution × fps tier and codec. Audio is a separate additive line that uses the audio sink rate.

Sink pricing.