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Reactor handles the entire media pipeline between your model and connected clients. You yield frames and audio samples from Python. Reactor encodes, packetizes, and delivers them over WebRTC with adaptive bitrate and low latency.

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Transport

All media is delivered over WebRTC. Clients connect via standard SDP offer/answer negotiation, and frames stream over RTP with sub-50ms delivery latency. The runtime manages the full WebRTC lifecycle: ICE, DTLS, SRTP, and congestion control. Commands and messages between the client and model travel over a WebRTC data channel alongside the media streams.

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Video codecs

Reactor supports multiple video codecs, negotiated automatically with each client via SDP: The table is the runtime’s preference order: it takes the first of these that appears in the client’s offer. VP9 leads for its balance of quality and compression, with H.264 close behind for maximum device compatibility. Encoding is done in software.

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Audio codec

Audio is encoded with Opus at 48 kHz. Opus is the standard WebRTC audio codec and is supported by all modern browsers.

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Resolution

There are no hardcoded resolution limits. The runtime encodes and streams whatever resolution your model produces. If your model yields 720p frames, the stream is 720p. If it yields 4K, the stream is 4K.

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Bitrate

The runtime uses adaptive bitrate control based on network conditions. It monitors transport-level feedback and adjusts encoding bitrate dynamically, scaling between 500 kbps and 10 Mbps depending on available bandwidth.

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Multiple tracks

A model can output any combination of video and audio tracks. Each field on your Output class becomes a separate media track:
Track names (main_video, secondary_video, main_audio) are the identifiers clients use to subscribe to specific streams.

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When the model falls behind

Each connection plays out at the rate its chunks are tagged with. When a tick comes round and the next frame is still being generated, the client holds the frame it already has and the wire stays quiet, which keeps the bandwidth for real content. A single black frame marks each boundary: the start of a connection, or a flush(). That is what gives a client a clean opening frame. In the other direction, emit() throttles a model that generates faster than the playout rate. See Buffered latency.

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Packet loss recovery

The runtime enables RTP retransmission (RTX) by default. Lost packets are retransmitted from a short buffer (200 ms), keeping streams clean without adding latency for packets that arrive normally.