[WIP] Rewrite wish-server in Rust

This should solve our GC pauses :)

Works locally. I still need to implement the SQLite-backed
authentication and stuff though.
main
Charlotte Som 2023-03-13 15:48:33 +00:00
parent acb190dc18
commit 5df33492f4
4 changed files with 3235 additions and 0 deletions

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/target

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wish-server-rs/Cargo.lock generated Normal file

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[package]
name = "wish-server-rs"
version = "0.1.0"
edition = "2021"
# See more keys and their definitions at https://doc.rust-lang.org/cargo/reference/manifest.html
[dependencies]
axum = { version = "0.6.11", features = ["macros"] }
miette = { version = "5.5.0", features = ["fancy"] }
once_cell = "1.17.1"
tokio = { version = "1.26.0", features = ["full"] }
tower-http = { version = "0.4.0", features = ["trace", "cors"] }
tracing = "0.1.37"
tracing-subscriber = "0.3.16"
webrtc = "0.7.1"

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use std::{
collections::{hash_map::Entry, HashMap},
fmt::Display,
net::SocketAddr,
sync::{
atomic::{AtomicU64, Ordering},
Arc, Mutex,
},
};
use axum::{
extract::State,
http::{header, HeaderMap, Method, StatusCode},
response::{IntoResponse, Response},
routing, Router,
};
use miette::{miette, Context, IntoDiagnostic, Result};
use once_cell::sync::Lazy;
use tower_http::{
cors::{Any, CorsLayer},
trace::TraceLayer,
};
use tracing::{error, instrument};
use webrtc::{
api::{
interceptor_registry::register_default_interceptors, media_engine::MediaEngine,
setting_engine::SettingEngine, APIBuilder, API as WebRTC,
},
ice::network_type::NetworkType,
ice_transport::ice_server::RTCIceServer,
interceptor::registry::Registry as InterceptorRegistry,
peer_connection::{
configuration::RTCConfiguration, peer_connection_state::RTCPeerConnectionState,
sdp::session_description::RTCSessionDescription, RTCPeerConnection,
},
rtp_transceiver::rtp_codec::RTCRtpCodecCapability,
track::track_local::{track_local_static_rtp::TrackLocalStaticRTP, TrackLocalWriter},
Error,
};
fn create_rtc_config() -> RTCConfiguration {
RTCConfiguration {
ice_servers: vec![
RTCIceServer {
urls: vec!["stun:stun.cloudflare.com:3478".into()],
..Default::default()
},
RTCIceServer {
urls: vec!["stun:stun.l.google.com:19302".into()],
..Default::default()
},
],
..Default::default()
}
}
fn setup_webrtc() -> Result<WebRTC> {
let mut media_engine = MediaEngine::default();
media_engine
.register_default_codecs()
.into_diagnostic()
.wrap_err("Failed to register default media engine codecs.")?;
let interceptor_registry = InterceptorRegistry::new();
let interceptor_registry =
register_default_interceptors(interceptor_registry, &mut media_engine)
.into_diagnostic()
.wrap_err("Failed to register default interceptors.")?;
let mut setting_engine = SettingEngine::default();
setup_ice(&mut setting_engine)?;
let api = APIBuilder::new()
.with_media_engine(media_engine)
.with_interceptor_registry(interceptor_registry)
.with_setting_engine(setting_engine)
.build();
Ok(api)
}
fn setup_ice(setting_engine: &mut SettingEngine) -> Result<()> {
setting_engine.set_network_types(vec![
NetworkType::Tcp4,
NetworkType::Tcp6,
NetworkType::Udp4,
NetworkType::Udp6,
]);
// TODO: Set up UDP muxing?
Ok(())
}
#[derive(Clone)]
struct OngoingStream {
video_track: Arc<TrackLocalStaticRTP>,
audio_track: Arc<TrackLocalStaticRTP>,
viewer_count: Arc<AtomicU64>,
}
static STREAMS: Lazy<Mutex<HashMap<String, OngoingStream>>> =
Lazy::new(|| Mutex::new(HashMap::new()));
async fn get_ongoing_stream(channel: &str) -> Result<OngoingStream> {
let mut streams = STREAMS.lock().expect("Mutex was poisoned");
let stream = match streams.entry(channel.into()) {
Entry::Occupied(o) => o.into_mut(),
Entry::Vacant(v) => {
let video_track = Arc::new(TrackLocalStaticRTP::new(
RTCRtpCodecCapability {
mime_type: "video/h264".into(),
..Default::default()
},
"video".into(),
"pion".into(),
));
let audio_track = Arc::new(TrackLocalStaticRTP::new(
RTCRtpCodecCapability {
mime_type: "audio/opus".into(),
..Default::default()
},
"audio".into(),
"pion".into(),
));
v.insert(OngoingStream {
video_track,
audio_track,
viewer_count: Arc::new(AtomicU64::new(0)),
})
}
};
Ok(stream.clone())
}
async fn setup_whip_connection(
channel: &str,
webrtc: Arc<WebRTC>,
) -> Result<Arc<RTCPeerConnection>> {
let rtc_config = create_rtc_config();
let peer_connection = Arc::new(
webrtc
.new_peer_connection(rtc_config)
.await
.into_diagnostic()?,
);
let OngoingStream {
video_track,
audio_track,
..
} = get_ongoing_stream(channel).await?;
peer_connection.on_track(Box::new(move |track, _recv, _tx| {
let local_track = if track.codec().capability.mime_type.starts_with("audio/") {
&audio_track
} else {
&video_track
}
.clone();
tokio::spawn(async move {
let mut rtp_buf = vec![0u8; 1500];
while let Ok((rtp_read, _)) = track.read(&mut rtp_buf).await {
let (Ok(_) | Err(Error::ErrClosedPipe)) = local_track.write(&rtp_buf[..rtp_read]).await else { break };
}
Result::<()>::Ok(())
});
Box::pin(async {})
}));
Ok(peer_connection)
}
fn log_http_error<E: Display>(status_code: StatusCode, error: E) -> Response {
error!("{error}");
(status_code, format!("{}", error)).into_response()
}
#[instrument(skip_all)]
async fn handle_whip(
method: Method,
headers: HeaderMap,
State(webrtc): State<Arc<WebRTC>>,
offer: String,
) -> Response {
if method != Method::POST {
return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
}
let Some(auth) = headers.get(header::AUTHORIZATION) else { return (StatusCode::UNAUTHORIZED, "Authorization was not set").into_response() };
let auth = auth
.to_str()
.into_diagnostic()
.wrap_err("Failed to decode auth header")
.unwrap();
let auth = auth.strip_prefix("Bearer ").unwrap_or(auth);
let Some((channel, _key)) = auth.split_once(':') else { return (StatusCode::UNAUTHORIZED, "Invalid Authorization header").into_response() };
// TODO: Validate the stream key
let peer_connection = match setup_whip_connection(channel, webrtc)
.await
.wrap_err("Failed to initialize peer connection")
{
Ok(peer_connection) => peer_connection,
Err(e) => {
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
}
};
let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
if let Err(e) = peer_connection.set_remote_description(description).await {
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
};
let mut gather_complete = peer_connection.gathering_complete_promise().await;
let answer = match peer_connection.create_answer(None).await {
Ok(answer) => answer,
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
};
if let Err(e) = peer_connection.set_local_description(answer).await {
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
}
let _ = gather_complete.recv().await;
match peer_connection.local_description().await {
Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
None => log_http_error(
StatusCode::INTERNAL_SERVER_ERROR,
miette!("No local description exists!"),
),
}
}
async fn setup_whep_connection(
channel: &str,
webrtc: Arc<WebRTC>,
) -> Result<Arc<RTCPeerConnection>> {
let rtc_config = RTCConfiguration::default();
let peer_connection = Arc::new(
webrtc
.new_peer_connection(rtc_config)
.await
.into_diagnostic()?,
);
let OngoingStream {
video_track,
audio_track,
viewer_count,
} = get_ongoing_stream(channel).await?;
peer_connection.on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
if s == RTCPeerConnectionState::Connected {
viewer_count.fetch_add(1, Ordering::Relaxed);
}
if s == RTCPeerConnectionState::Disconnected {
viewer_count.fetch_sub(1, Ordering::Relaxed);
}
Box::pin(async {})
}));
peer_connection
.add_track(video_track)
.await
.into_diagnostic()
.wrap_err("Failed to add video track")?;
peer_connection
.add_track(audio_track)
.await
.into_diagnostic()
.wrap_err("Failed to add audio track")?;
Ok(peer_connection)
}
#[instrument(skip_all)]
async fn handle_whep(
method: Method,
headers: HeaderMap,
State(webrtc): State<Arc<WebRTC>>,
offer: String,
) -> Response {
if method != Method::POST {
return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
}
let channel = match headers
.get(header::AUTHORIZATION)
.ok_or(miette!("Authorization header was not set"))
.and_then(|h| {
h.to_str()
.into_diagnostic()
.wrap_err("Authorization header was malformed")
}) {
Ok(a) => a,
Err(e) => return log_http_error(StatusCode::BAD_REQUEST, e),
};
let channel = channel.strip_prefix("Bearer ").unwrap_or(channel);
let peer_connection = match setup_whep_connection(channel, webrtc).await {
Ok(p) => p,
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
};
let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
if let Err(e) = peer_connection.set_remote_description(description).await {
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
};
let mut gather_complete = peer_connection.gathering_complete_promise().await;
let answer = match peer_connection.create_answer(None).await {
Ok(answer) => answer,
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
};
if let Err(e) = peer_connection.set_local_description(answer).await {
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
}
let _ = gather_complete.recv().await;
match peer_connection.local_description().await {
Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
None => log_http_error(
StatusCode::INTERNAL_SERVER_ERROR,
miette!("No local description exists!"),
),
}
}
#[tokio::main]
async fn main() -> Result<()> {
let webrtc = Arc::new(setup_webrtc()?);
tracing_subscriber::fmt::init();
// TODO: CORS
let app = Router::new()
.route("/api/wish-server/whip", routing::any(handle_whip))
.route("/api/wish-server/whep", routing::any(handle_whep))
.layer(TraceLayer::new_for_http())
.layer(
CorsLayer::new()
.allow_methods(Any)
.allow_origin(Any)
.allow_headers(Any),
)
.with_state(webrtc);
let bind_addr: SocketAddr = "127.0.0.1:3001"
.parse()
.into_diagnostic()
.wrap_err("Couldn't parse bind address")?;
println!("Listening at http://{bind_addr}/ ...");
axum::Server::bind(&bind_addr)
.serve(app.into_make_service())
.await
.unwrap();
Ok(())
}