Merge branch 'master' of https://github.com/Johni0702/mumble-web-proxy

Cargo.toml

@@ -6,20 +6,20 @@ edition = "2018"
 
 [dependencies]
 argparse = "0.2.2"
-bytes = "0.5"
+bytes = "1"
 byteorder = "1.2"
 futures = { version = "0.3", features = ["compat", "io-compat"] }
 toml = "0.5"
 serde = { version = "1.0", features = ["derive"] }
-tokio = { version = "0.2", features = ["full"] }
+tokio = { version = "1", features = ["full"] }
-tokio-util = { version = "0.3", features = ["codec"] }
+tokio-util = { version = "0.6", features = ["codec"] }
-tokio-tls = "0.3"
+tokio-native-tls = "0.3"
 native-tls = "0.2"
-mumble-protocol = { version = "0.3", features = ["webrtc-extensions"] }
+mumble-protocol = { version = "0.4", features = ["webrtc-extensions"] }
-tokio-tungstenite = "0.10"
+tokio-tungstenite = "0.13"
 http = "0.2"
-tungstenite = "0.10"
+tungstenite = "0.12"
-rtp = { git = "https://github.com/johni0702/rtp", rev = "1444b3c", features = ["rfc5764-openssl"] }
+rtp = { git = "https://github.com/johni0702/rtp", rev = "6c0223d", features = ["rfc5764-openssl"] }
-libnice = "0.2.1"
+libnice = "0.3"
 webrtc-sdp = "0.3"
 openssl = "0.10"

README.md

@@ -13,7 +13,7 @@ Note that it requires an extension to the Mumble protocol which has not yet been
 
 #### Prerequisites
 
-- Rust 1.39+ (e.g. via [rustup](https://rustup.rs/))
+- Rust 1.45+ (e.g. via [rustup](https://rustup.rs/))
 - libnice development headers (`libnice-devel` on Fedora, `libnice-dev` on Debian)
 - OpenSSL development headers (`openssl-devel` on Fedora, `libssl-dev` on Debian)
 - clang (`clang` on Fedora and Debian)

src/connection.rs

@@ -30,7 +30,7 @@ use std::task::Context;
 use std::task::Poll;
 use std::time::Duration;
 use tokio::io;
-use tokio::time::Delay;
+use tokio::time::Sleep;
 use webrtc_sdp::attribute_type::SdpAttribute;
 
 use crate::error::Error;
@@ -39,10 +39,10 @@ use crate::Config;
 type SessionId = u32;
 
 struct User {
-    session: u32,           // mumble session id
+    session: u32,                     // mumble session id
-    ssrc: u32,              // ssrc id
+    ssrc: u32,                        // ssrc id
-    active: bool,           // whether the user is currently transmitting audio
+    active: bool,                     // whether the user is currently transmitting audio
-    timeout: Option<Delay>, // assume end of transmission if silent until then
+    timeout: Option<Pin<Box<Sleep>>>, // assume end of transmission if silent until then
     start_voice_seq_num: u64,
     highest_voice_seq_num: u64,
     rtp_seq_num_offset: u32, // u32 because we also derive the timestamp from it
@@ -70,7 +70,7 @@ impl User {
     }
 
     fn set_active(&mut self, target: u8) -> Option<Frame> {
-        self.timeout = Some(tokio::time::delay_for(Duration::from_millis(400)));
+        self.timeout = Some(Box::pin(tokio::time::sleep(Duration::from_millis(400))));
 
         if self.active {
             None
@@ -587,8 +587,7 @@ impl Future for Connection {
             // (same applies to the other futures directly below it)
             for session in self.sessions.values_mut() {
                 if let Some(timeout) = &mut session.timeout {
-                    pin_mut!(timeout);
+                    if let Poll::Ready(()) = timeout.poll_unpin(cx) {
-                    if let Poll::Ready(()) = timeout.poll(cx) {
                         if let Some(frame) = session.set_inactive() {
                             self.outbound_buf.push_back(frame);
                         }

src/error.rs

@@ -37,12 +37,6 @@ impl From<native_tls::Error> for Error {
     }
 }
 
-impl From<tokio::time::Error> for Error {
-    fn from(e: tokio::time::Error) -> Self {
-        Error::Misc(Box::new(e))
-    }
-}
-
 impl From<rtp::Error> for Error {
     fn from(e: rtp::Error) -> Self {
         Error::Misc(Box::new(e))

src/main.rs

@@ -21,7 +21,7 @@ use std::net::Ipv6Addr;
 use std::net::ToSocketAddrs;
 use tokio::net::TcpListener;
 use tokio::net::TcpStream;
-use tokio_tls::TlsConnector;
+use tokio_native_tls::TlsConnector;
 use tokio_tungstenite::accept_hdr_async_with_config;
 use tokio_util::codec::Decoder;
 use tungstenite::handshake::server::{ErrorResponse, Request, Response};
@@ -179,7 +179,7 @@ async fn main() -> Result<(), Error> {
 
     println!("Binding to port {}", ws_port);
     let socket_addr = (Ipv6Addr::from(0), ws_port);
-    let mut server = TcpListener::bind(&socket_addr).await?;
+    let server = TcpListener::bind(&socket_addr).await?;
 
     println!("Waiting for client connections..");
     loop {
@@ -212,6 +212,7 @@ async fn main() -> Result<(), Error> {
             max_send_queue: Some(10), // can be fairly small as voice is using WebRTC instead
             max_message_size: Some(0x7f_ffff), // maximum size accepted by Murmur
             max_frame_size: Some(0x7f_ffff), // maximum size accepted by Murmur
+            accept_unmasked_frames: false, // browsers should comply with RFC 6455
         };
         fn header_callback(
             _req: &Request,