Composer Reference Guide R2-2026-2.6.9642.28010

RTMP input

RTMP input properties for Script Engine. Receives a live video and audio stream over RTMP / RTMPS. Supports two modes: Client (pulls from an existing RTMP server — the same protocol used by most streaming services and many encoders for contribution feeds) and Server (binds to a local port and accepts an incoming push from an external RTMP sender). Includes auto-connect on load, automatic reconnect / re-listen on signal loss, configurable buffer length, and a delayed-start option for staggering multiple inputs. Optional NVIDIA NVDEC hardware decoding for H.264, HEVC, and AV1 (AV1 only on GPUs that support it — Ada Lovelace / RTX 40-series and newer); other codecs and GPUs fall back to software decoding.

Property Type Access Description
ShowAdvancedOptions bool get/set Show or hide the input's advanced settings in the editor UI. [default=false].
Mode RtmpMode get/set Whether the input pulls from a remote RTMP server or listens locally for an incoming push. [default=Client]. Client connects out to an existing RTMP server and pulls the stream — use this for CDNs, restream services, and any source that exposes a public ingest URL. Server binds the URL's host/port locally and waits for an external RTMP sender (OBS, vMix, hardware encoder, FFmpeg push, etc.) to connect to it — useful for direct contribution feeds without a separate RTMP server in between. Note: Server mode has no protocol-level authentication and no rtmps:// support. Any sender that can reach the port can publish. For per-stream auth, transport encryption, recording or multi-publisher fan-out, front Composer with nginx-rtmp / MediaMTX and pull from it in Client mode.
RemoteServerAddress string get/set RTMP / RTMPS URL the input uses. In Client mode this is the URL of the existing RTMP source to pull from, e.g. rtmp://host/app/streamkey. Both rtmp:// and rtmps:// are supported in client mode. In Server mode this is the local rtmp:// bind address the input listens on, e.g. rtmp://0.0.0.0:1935/live (use 0.0.0.0 to accept on all interfaces — narrow it to a specific NIC to restrict access). Server mode does not support rtmps://. Supports the @@HostName() macro, replaced with the machine's hostname at connect time (note the parentheses — bare @@HOSTNAME is not recognised here). Security: RTMP listen mode has no protocol-level authentication — any sender that can reach the port can publish. For real auth (per-stream keys, IP allow-lists, recording, transcoding) front Composer with a dedicated RTMP server such as nginx-rtmp or MediaMTX and pull from it in Client mode.
AutoConnect bool get/set Whether the input starts automatically as soon as the project loads. [default=true]. When true, the input invokes its connect logic immediately on load (equivalent to running ConnectCommand) — connecting out in Client mode, starting to listen in Server mode. Disable to require an explicit start, useful when the input should only come live on operator action or via a script trigger.
AutoReconnect bool get/set Whether the input automatically recovers after the stream ends. [default=true]. In Client mode: keeps retrying the source every ReconnectInterval until it comes back. In Server mode: re-binds and listens for a new sender after the current one disconnects. Useful for long-running productions where the stream may briefly drop. Disable for one-shot sessions that should not auto-recover.
ReconnectInterval int get/set Delay between automatic reconnect or re-listen attempts, in milliseconds. [min=1000, max=10000, default=5000]. Only used when AutoReconnect is true. Shorter intervals recover faster from transient blips at the cost of more retry churn; longer intervals reduce the retry rate. Applies equally to Client-mode reconnects and Server-mode re-listens.
RtmpBufferLength RtmpBufferLength get/set How much video and audio is buffered ahead of playback. [default=Medium]. Choose between Short, Medium, and Long. A longer buffer absorbs more network jitter and keeps playback steady on bumpy links, at the cost of more added latency. A shorter buffer is tighter on latency but more sensitive to network hiccups.
DelayedInitialConnect int get/set Wait this long before the input first starts, in milliseconds. [min=0, max=10000, default=0]. In Client mode: staggers several inputs so they don't all hit the source server at the same instant when the project loads. In Server mode: delays opening the listen socket on load — useful if you also need to stagger port binding. Only affects the very first attempt — subsequent reconnects / re-listens use ReconnectInterval.
UseNvdec bool get/set Whether to use NVIDIA's NVDEC GPU decoder instead of the libavcodec CPU decoder. [default=false]. NVDEC offloads H.264, HEVC, and AV1 decoding to dedicated silicon on the GPU, freeing CPU and typically sustaining higher resolutions and frame counts than software decoding. Applies only to incoming H.264, HEVC, and AV1 streams — other codecs always fall back to software decoding regardless of this setting. AV1 hardware decode requires a GPU that supports it (Ada Lovelace / RTX 40-series and newer); on older GPUs the connect will fail for AV1 streams while NVDEC is enabled. Requires a CUDA-capable NVIDIA GPU. The change takes effect on the next connect.
ConnectionStatus ConnectionStatus get Current connection state of the input (read-only). Reflects whether the input is Disconnected, Connecting, Connected, Reconnecting, or in an error state. Driven indirectly by ConnectCommand, DisconnectCommand, and AutoConnect at load time. Read this from a script to decide whether the stream is live; the value also drives the running indicator and the enabled state of the connect buttons.
ConnectCommand Command get Start the input — connect to the configured RTMP source (Client mode) or begin listening for an incoming push (Server mode).
DisconnectCommand Command get Stop the input — disconnect from the source (Client mode) or stop listening / drop the current sender (Server mode).
ComponentLog ComponentLog get Recent log lines from the input — connect, disconnect, reconnect, and error events (read-only). A short history of the most recent RTMP events, useful for diagnosing why a connection failed or why the input reconnected.
StreamTime TimeCode get Time elapsed since the stream started, as a time code (read-only). Counts up from 0:00:00:00 when the connection becomes live and resets on each new connection. Useful for displaying the stream's running duration.
Width int get Width of the video frames currently being received, in pixels (read-only).
Height int get Height of the video frames currently being received, in pixels (read-only).
NumAutoReconnectsPerformed int get Total number of automatic reconnects performed since the input was started (read-only). A rapidly growing counter is a strong signal that the network path or source is unstable. Resets only on full input dispose / re-create.
RtmpBufferState NetworkBufferingState get Whether the input is currently filling its receive buffer (read-only). Buffering means the input is waiting for enough data before it starts producing frames; Streaming means it's running normally. Sustained buffering states indicate the network can't keep up with the configured RtmpBufferLength.
NumFlushesPerformed int get Number of times the input has dropped its accumulated buffer to recover from drift (read-only). Buffer flushes happen when the input falls too far behind the source and needs to resync. A non-zero value usually indicates network jitter or a slow downstream consumer.
VideoBufferSizeFrames int get Number of video frames currently buffered ahead of playback (read-only, debug).
VideoBufferSizeMs int get How much video is currently buffered ahead of playback, in milliseconds (read-only, debug). A persistently low or zero value while connected suggests the network can't keep up with real-time playback and may produce visible stalls.
AudioBufferSizeMs int get How much audio is currently buffered ahead of playback, in milliseconds (read-only, debug). A persistently low or zero value may produce audible drop-outs.
PlaybackTime AVRational get Timestamp of the most recently rendered frame, relative to stream start (read-only, debug).
VideoTimeStamp TimeStamp get Timestamp of the most recently rendered video frame (read-only, debug).
AudioTimeStamp TimeStamp get Timestamp of the most recently rendered audio frame (read-only, debug).
VideoPacketTimeStamp TimeSpan get Timestamp of the most recently received video packet, before decoding (read-only, debug).
AudioPacketTimeStamp TimeSpan get Timestamp of the most recently received audio packet, before decoding (read-only, debug).
PacketTimeStampDiff int get Difference between the most recent video and audio packet timestamps (read-only, debug). A small steady value is normal; a value that grows or oscillates indicates A/V drift in the incoming stream.
ExpectedVideoPtsDelta double get Expected time between consecutive video frames at the source's frame rate, in milliseconds (read-only, debug). Useful for diagnosing pacing problems — if actual deltas drift far from this expected value, the upstream encoder isn't producing frames at a steady rate.
VideoTrackInfo Dictionary<string, object> get Metadata about the incoming video stream (read-only). A dictionary with technical details about the video — codec, resolution, frame rate, bit rate, and so on. Available once the source has been negotiated.
AudioTrackInfo Dictionary<string, object> get Metadata about the incoming audio stream (read-only). A dictionary with technical details about the audio — codec, sample rate, channel count, bit rate, and so on. Available once the source has been negotiated.

Inherits from: AbstractInput, AbstractAudioProcessing, AbstractAudioMetering.

See also: RTMP input in Inputs — user-facing introduction, screenshots, and section summaries.