Async RTMP server and client library for Rust - Build live video streaming infrastructure with Tokio.
rtmp-rs is an RTMP server and client library written in Rust, built to handle the messy reality of live streaming. Encoders like OBS and FFmpeg all have their quirks, so rtmp-rs uses lenient parsing to roll with it instead of rejecting non-conforming streams. Late joiners get instant playback with keyframe caching, and smart backpressure handling keeps audio flowing for slow subscribers while selectively dropping video frames. If you're building a streaming service or relay and want RTMP ingest that just works, this might be what you're looking for.
- Async/Await - Built on Tokio for high-performance concurrent connections
- Zero-Copy - Uses
bytes::Bytesthroughout for efficient memory handling - Backpressure Handling - Slow subscribers drop video frames while audio keeps flowing, so viewers hear continuous sound instead of staring at a frozen frame
- Built-in Pub/Sub - Stream key routing works out of the box; no code required
- Late-Joiner GOP Cache - Buffers keyframes so viewers don't wait for the next IDR frame when joining mid-stream
- Lenient Parsing - Handles encoder quirks like empty app names and timestamp regression (OBS, Twitch compatible)
- Extensible - Optional
RtmpHandlercallbacks for custom auth and media processing
1. Add dependency:
cargo add rtmp-rs2. Create a server:
use rtmp_rs::{RtmpServer, ServerConfig, RtmpHandler, AuthResult};
use rtmp_rs::session::SessionContext;
use rtmp_rs::protocol::message::{ConnectParams, PublishParams};
struct MyHandler;
impl RtmpHandler for MyHandler {
async fn on_connect(&self, _ctx: &SessionContext, params: &ConnectParams) -> AuthResult {
println!("App: {}", params.app);
AuthResult::Accept
}
async fn on_publish(&self, _ctx: &SessionContext, params: &PublishParams) -> AuthResult {
println!("Stream key: {}", params.stream_key);
// Add your stream key validation here
AuthResult::Accept
}
// See RtmpHandler trait for other available callbacks
}
#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
let server = RtmpServer::new(ServerConfig::default(), MyHandler);
server.run().await?;
Ok(())
}3. Stream to it:
# OBS: Server: rtmp://localhost/live Stream Key: test
# ffmpeg:
ffmpeg -re -i input.mp4 -c copy -f flv rtmp://localhost/live/testuse rtmp_rs::client::{ClientConfig, ClientEvent, RtmpPuller};
#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
let config = ClientConfig::new("rtmp://server/live/stream_key");
let (puller, mut events) = RtmpPuller::new(config);
tokio::spawn(async move {
while let Some(event) = events.recv().await {
match event {
ClientEvent::VideoFrame { timestamp, data } => {
if data.is_keyframe() {
println!("Keyframe at {}ms", timestamp);
}
}
ClientEvent::AudioFrame { timestamp, .. } => {
println!("Audio at {}ms", timestamp);
}
ClientEvent::Disconnected => break,
_ => {}
}
}
});
puller.start().await?;
Ok(())
}The RtmpHandler trait provides optional hooks for custom logic. All callbacks have sensible defaults that accept connections and allow streams - you only override what you need:
use rtmp_rs::{RtmpHandler, AuthResult};
use rtmp_rs::session::SessionContext;
use rtmp_rs::protocol::message::PublishParams;
struct AuthHandler;
impl RtmpHandler for AuthHandler {
// Only override what you need - everything else uses defaults
async fn on_publish(&self, _ctx: &SessionContext, params: &PublishParams) -> AuthResult {
if validate_stream_key(¶ms.stream_key) {
AuthResult::Accept
} else {
AuthResult::Reject("Invalid stream key".into())
}
}
}Available callbacks:
| Callback | Use Case |
|---|---|
on_connection |
IP blocklist, rate limiting |
on_handshake_complete |
Post-handshake setup, before connect command, logging |
on_connect |
Validate app name, parse auth tokens from tcUrl |
on_disconnect |
Connection cleanup, logging |
on_fc_publish |
Early stream key validation (OBS sends this first) |
on_publish |
Main stream key authentication |
on_unpublish |
Publisher cleanup, notifications |
on_play |
Subscriber authorization |
on_pause |
Handle subscriber pause |
on_unpause |
Handle subscriber resume |
on_metadata |
Capture stream info (resolution, bitrate, codec) |
on_media_tag |
Raw FLV tag access, custom filtering |
on_video_frame |
Process H.264 NALUs |
on_audio_frame |
Process AAC frames |
on_keyframe |
Track GOP boundaries |
use std::time::Duration;
use rtmp_rs::ServerConfig;
let config = ServerConfig::default()
.bind("0.0.0.0:1935".parse()?)
.max_connections(1000)
.chunk_size(4096)
.connection_timeout(Duration::from_secs(10))
.idle_timeout(Duration::from_secs(60));# Stream (publish) with ffmpeg
ffmpeg -re -i test.mp4 -c copy -f flv rtmp://localhost/live/test_key
# Play with ffplay
ffplay rtmp://localhost/live/test_keyThis repo is a Rust rewrite of my RTMP Go server. Almost all of the code in this Rust version was written by AI (Claude Opus 4.5).
I recently had an idea that required an RTMP server, so I used it as an excuse to write some Rust and try out some agentic programming. This repo is partly an experiment to see how far I could get by vibecoding the entire thing with Claude Code. The answer? Far!
The whole thing took around 8 hours. It probably could have been faster if I auto-accepted edits without reading the code, but I like to review everything the agent generates. I started with Plan Mode to define the requirements, then moved on to implementation.
That said, there was a tricky timestamp bug that caused audio/video stuttering, and Claude kept hallucinating answers instead of helping. After a deep-dive on my own, I found the root cause. I also noticed some parts of the code that could be improved, but I decided to keep things as-is for now. Any future improvements I'll have Claude handle.
Licensed under MIT license