659 lines
25 KiB
Rust
659 lines
25 KiB
Rust
use futures::future::BoxFuture;
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use futures::pin_mut;
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use futures::ready;
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use futures::{Future, FutureExt, Sink, Stream};
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use libnice::ice;
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use mumble_protocol::control::msgs;
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use mumble_protocol::control::ControlPacket;
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use mumble_protocol::voice::VoicePacket;
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use mumble_protocol::voice::VoicePacketPayload;
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use mumble_protocol::Clientbound;
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use mumble_protocol::Serverbound;
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use openssl::asn1::Asn1Time;
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use openssl::hash::MessageDigest;
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use openssl::pkey::{PKey, Private};
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use openssl::rsa::Rsa;
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use openssl::ssl::{SslAcceptor, SslAcceptorBuilder, SslMethod};
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use openssl::x509::X509;
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use rtp::rfc3550::{
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RtcpCompoundPacket, RtcpPacket, RtcpPacketReader, RtcpPacketWriter, RtpFixedHeader, RtpPacket,
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RtpPacketReader, RtpPacketWriter,
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};
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use rtp::rfc5761::{MuxPacketReader, MuxPacketWriter, MuxedPacket};
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use rtp::rfc5764::DtlsSrtp;
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use rtp::traits::{ReadPacket, WritePacket};
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use std::collections::{BTreeMap, VecDeque};
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use std::ffi::CString;
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use std::net::IpAddr;
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use std::pin::Pin;
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use std::task::Context;
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use std::task::Poll;
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use std::time::Duration;
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use tokio::io;
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use tokio::time::Sleep;
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use webrtc_sdp::attribute_type::SdpAttribute;
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use crate::error::Error;
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use crate::Config;
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type SessionId = u32;
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struct User {
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session: u32, // mumble session id
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ssrc: u32, // ssrc id
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active: bool, // whether the user is currently transmitting audio
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timeout: Option<Pin<Box<Sleep>>>, // assume end of transmission if silent until then
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start_voice_seq_num: u64,
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highest_voice_seq_num: u64,
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rtp_seq_num_offset: u32, // u32 because we also derive the timestamp from it
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}
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impl User {
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fn set_inactive(&mut self) -> Option<Frame> {
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self.timeout = None;
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if self.active {
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self.active = false;
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self.rtp_seq_num_offset = self
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.rtp_seq_num_offset
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.wrapping_add((self.highest_voice_seq_num - self.start_voice_seq_num) as u32 + 1);
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self.start_voice_seq_num = 0;
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self.highest_voice_seq_num = 0;
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let mut msg = msgs::TalkingState::new();
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msg.set_session(self.session);
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Some(Frame::Client(msg.into()))
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} else {
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None
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}
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}
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fn set_active(&mut self, target: u8) -> Option<Frame> {
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self.timeout = Some(Box::pin(tokio::time::sleep(Duration::from_millis(400))));
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if self.active {
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None
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} else {
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self.active = true;
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let mut msg = msgs::TalkingState::new();
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msg.set_session(self.session);
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msg.set_target(target.into());
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Some(Frame::Client(msg.into()))
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}
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}
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}
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pub struct Connection {
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config: Config,
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inbound_client: Pin<Box<dyn Stream<Item = Result<ControlPacket<Serverbound>, Error>> + Send>>,
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outbound_client: Pin<Box<dyn Sink<ControlPacket<Clientbound>, Error = Error> + Send>>,
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inbound_server: Pin<Box<dyn Stream<Item = Result<ControlPacket<Clientbound>, Error>> + Send>>,
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outbound_server: Pin<Box<dyn Sink<ControlPacket<Serverbound>, Error = Error> + Send>>,
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outbound_buf: VecDeque<Frame>,
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ice: Option<(ice::Agent, ice::Stream)>,
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candidate_gathering_done: bool,
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dtls_srtp_future: Option<
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BoxFuture<'static, Result<DtlsSrtp<ice::StreamComponent, SslAcceptorBuilder>, io::Error>>,
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>,
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dtls_srtp: Option<DtlsSrtp<ice::StreamComponent, SslAcceptorBuilder>>,
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dtls_key: PKey<Private>,
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dtls_cert: X509,
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rtp_reader: MuxPacketReader<RtpPacketReader, RtcpPacketReader>,
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rtp_writer: MuxPacketWriter<RtpPacketWriter, RtcpPacketWriter>,
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target: Option<u8>, // only if client is talking
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next_ssrc: u32,
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free_ssrcs: Vec<u32>,
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sessions: BTreeMap<SessionId, User>,
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}
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impl Connection {
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pub fn new<CSi, CSt, SSi, SSt>(
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config: Config,
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client_sink: CSi,
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client_stream: CSt,
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server_sink: SSi,
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server_stream: SSt,
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) -> Self
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where
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CSi: Sink<ControlPacket<Clientbound>, Error = Error> + 'static + Send,
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CSt: Stream<Item = Result<ControlPacket<Serverbound>, Error>> + 'static + Send,
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SSi: Sink<ControlPacket<Serverbound>, Error = Error> + 'static + Send,
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SSt: Stream<Item = Result<ControlPacket<Clientbound>, Error>> + 'static + Send,
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{
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let rsa = Rsa::generate(2048).unwrap();
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let key = PKey::from_rsa(rsa).unwrap();
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let mut cert_builder = X509::builder().unwrap();
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cert_builder
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.set_not_after(&Asn1Time::days_from_now(1).unwrap())
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.unwrap();
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cert_builder
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.set_not_before(&Asn1Time::days_from_now(0).unwrap())
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.unwrap();
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cert_builder.set_pubkey(&key).unwrap();
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cert_builder.sign(&key, MessageDigest::sha256()).unwrap();
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let cert = cert_builder.build();
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Self {
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config,
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inbound_client: Box::pin(client_stream),
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outbound_client: Box::pin(client_sink),
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inbound_server: Box::pin(server_stream),
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outbound_server: Box::pin(server_sink),
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outbound_buf: VecDeque::new(),
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ice: None,
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candidate_gathering_done: false,
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dtls_srtp_future: None,
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dtls_srtp: None,
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dtls_key: key,
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dtls_cert: cert,
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rtp_reader: MuxPacketReader::new(RtpPacketReader, RtcpPacketReader),
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rtp_writer: MuxPacketWriter::new(RtpPacketWriter, RtcpPacketWriter),
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target: None,
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next_ssrc: 1,
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free_ssrcs: Vec::new(),
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sessions: BTreeMap::new(),
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}
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}
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fn supports_webrtc(&self) -> bool {
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self.ice.is_some()
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}
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fn allocate_ssrc(&mut self, session_id: SessionId) -> &mut User {
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let ssrc = self.free_ssrcs.pop().unwrap_or_else(|| {
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let ssrc = self.next_ssrc;
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self.next_ssrc += 1;
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if let Some(ref mut dtls_srtp) = self.dtls_srtp {
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dtls_srtp.add_incoming_unknown_ssrcs(1);
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dtls_srtp.add_outgoing_unknown_ssrcs(1);
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}
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ssrc
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});
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let user = User {
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session: session_id,
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ssrc,
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active: false,
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timeout: None,
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start_voice_seq_num: 0,
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highest_voice_seq_num: 0,
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rtp_seq_num_offset: 0,
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};
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self.sessions.insert(session_id, user);
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self.sessions.get_mut(&session_id).unwrap()
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}
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fn free_ssrc(&mut self, session_id: SessionId) {
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if let Some(user) = self.sessions.remove(&session_id) {
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self.free_ssrcs.push(user.ssrc)
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}
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}
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fn setup_ice(&mut self) -> Result<(), Error> {
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// Setup ICE agent
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let mut agent = ice::Agent::new_rfc5245();
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agent.set_software("mumble-web-proxy");
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agent.set_controlling_mode(true);
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// Setup ICE stream
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let mut stream = match {
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let mut builder = agent.stream_builder(1);
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if self.config.ice_min_port != 1 || self.config.ice_max_port != u16::max_value() {
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builder.set_port_range(self.config.ice_min_port, self.config.ice_max_port);
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}
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builder.build()
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} {
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Ok(stream) => stream,
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Err(err) => {
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return Err(io::Error::new(io::ErrorKind::Other, err).into());
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}
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};
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let component = stream.take_components().pop().expect("one component");
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// Send WebRTC details to the client
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let mut msg = msgs::WebRTC::new();
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msg.set_dtls_fingerprint(
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self.dtls_cert
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.digest(MessageDigest::sha256())
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.unwrap()
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.iter()
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.map(|byte| format!("{:02X}", byte))
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.collect::<Vec<_>>()
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.join(":"),
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);
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msg.set_ice_pwd(stream.get_local_pwd().to_owned());
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msg.set_ice_ufrag(stream.get_local_ufrag().to_owned());
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// Store ice agent and stream for later use
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self.ice = Some((agent, stream));
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// Prepare to accept the DTLS connection
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let mut acceptor = SslAcceptor::mozilla_intermediate(SslMethod::dtls()).unwrap();
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acceptor.set_certificate(&self.dtls_cert).unwrap();
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acceptor.set_private_key(&self.dtls_key).unwrap();
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// FIXME: verify remote fingerprint
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self.dtls_srtp_future = Some(DtlsSrtp::handshake(component, acceptor).boxed());
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self.outbound_buf.push_back(Frame::Client(msg.into()));
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Ok(())
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}
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fn gather_ice_candidates(mut self: Pin<&mut Self>, cx: &mut Context) -> bool {
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if self.candidate_gathering_done {
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return false;
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}
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let stream = match self.ice {
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Some((_, ref mut stream)) => stream,
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None => return false,
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};
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pin_mut!(stream);
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match stream.poll_next(cx) {
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Poll::Ready(Some(mut candidate)) => {
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println!("Local ice candidate: {}", candidate.to_string());
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// Map to public addresses (if configured)
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let config = &self.config;
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match (
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&mut candidate.address,
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config.ice_public_v4,
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config.ice_public_v6,
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) {
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(webrtc_sdp::address::Address::Ip(IpAddr::V4(addr)), Some(public), _) => {
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*addr = public;
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}
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(webrtc_sdp::address::Address::Ip(IpAddr::V6(addr)), _, Some(public)) => {
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*addr = public;
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}
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_ => {} // non configured
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};
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// Got a new candidate, send it to the client
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let mut msg = msgs::IceCandidate::new();
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msg.set_content(format!("candidate:{}", candidate.to_string()));
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let frame = Frame::Client(msg.into());
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self.outbound_buf.push_back(frame);
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true
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}
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Poll::Ready(None) => {
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self.candidate_gathering_done = true;
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false
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}
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_ => false,
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}
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}
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fn dispatch_outbound_frames(
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mut self: Pin<&mut Self>,
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cx: &mut Context,
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) -> Poll<Result<(), Error>> {
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// Make sure we can send any pending frames before trying to do so
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ready!(self.outbound_server.as_mut().poll_ready(cx)?);
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ready!(self.outbound_client.as_mut().poll_ready(cx)?);
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if let Some(ref mut dtls_srtp) = self.dtls_srtp {
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ready!(Pin::new(dtls_srtp).poll_ready(cx)?);
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}
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// Send out all pending frames
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while let Some(frame) = self.outbound_buf.pop_front() {
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match frame {
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Frame::Server(frame) => {
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self.outbound_server.as_mut().start_send(frame)?;
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ready!(self.outbound_server.as_mut().poll_ready(cx)?);
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}
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Frame::Client(frame) => {
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self.outbound_client.as_mut().start_send(frame)?;
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ready!(self.outbound_client.as_mut().poll_ready(cx)?);
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}
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Frame::Rtp(frame) => {
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let mut buf = Vec::new();
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self.rtp_writer.write_packet(&mut buf, &frame)?;
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if let Some(ref mut dtls_srtp) = self.dtls_srtp {
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pin_mut!(dtls_srtp);
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dtls_srtp.as_mut().start_send(&buf)?;
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ready!(dtls_srtp.poll_ready(cx)?);
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} else {
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// RTP not yet setup, just drop the frame
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}
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}
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}
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}
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// All frames have been sent (or queued), flush any buffers in the output path
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let _ = self.outbound_client.as_mut().poll_flush(cx)?;
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let _ = self.outbound_server.as_mut().poll_flush(cx)?;
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if let Some(ref mut dtls_srtp) = self.dtls_srtp {
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let _ = Pin::new(dtls_srtp).poll_flush(cx)?;
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}
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Poll::Ready(Ok(()))
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}
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fn handle_voice_packet(&mut self, packet: VoicePacket<Clientbound>) -> Result<(), Error> {
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let (target, session_id, seq_num, opus_data, last_bit) = match packet {
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VoicePacket::Audio {
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target,
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session_id,
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seq_num,
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payload: VoicePacketPayload::Opus(data, last_bit),
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..
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} => (target, session_id, seq_num, data, last_bit),
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_ => return Ok(()),
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};
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// NOTE: the mumble packet id increases by 1 per 10ms of audio contained
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// whereas rtp seq_num should increase by 1 per packet, regardless of audio,
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// but firefox seems to be just fine if we skip over rtp seq_nums.
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// NOTE: we rely on the srtp layer to prevent two-time-pads and by doing so,
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// allow for (reasonable) jitter of incoming voice packets.
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let user = match self.sessions.get_mut(&(session_id as u32)) {
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Some(s) => s,
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None => return Ok(()),
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};
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let rtp_ssrc = user.ssrc;
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let mut first_in_transmission = if user.active {
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false
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} else {
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user.start_voice_seq_num = seq_num;
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user.highest_voice_seq_num = seq_num;
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true
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};
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let offset = seq_num - user.start_voice_seq_num;
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let mut rtp_seq_num = user.rtp_seq_num_offset + offset as u32;
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if last_bit {
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if seq_num <= user.highest_voice_seq_num {
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// Horribly delayed end packet from a previous stream, just drop it
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// (or single packet stream which would be inaudible anyway)
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return Ok(());
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}
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// this is the last packet of this voice transmission -> reset counters
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// doing that will effectively trash any delayed packets but that's just
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// a flaw in the mumble protocol and there's nothing we can do about it.
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if let Some(frame) = user.set_inactive() {
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self.outbound_buf.push_back(frame);
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}
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} else if seq_num == user.highest_voice_seq_num && seq_num != user.start_voice_seq_num {
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// re-transmission, drop it
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return Ok(());
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} else if seq_num >= user.highest_voice_seq_num
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&& seq_num < user.highest_voice_seq_num + 100
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{
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// probably same voice transmission (also not too far in the future)
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user.highest_voice_seq_num = seq_num;
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if let Some(frame) = user.set_active(target) {
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self.outbound_buf.push_back(frame);
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}
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} else if seq_num < user.highest_voice_seq_num && seq_num + 100 > user.highest_voice_seq_num
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{
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// slightly delayed but probably same voice transmission
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if let Some(frame) = user.set_active(target) {
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self.outbound_buf.push_back(frame);
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}
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} else {
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// Either significant jitter (>2s) or we missed the end of the last
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// transmission. Since >2s jitter will break opus horribly anyway,
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// we assume the latter and start a new transmission
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if let Some(frame) = user.set_inactive() {
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self.outbound_buf.push_back(frame);
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}
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first_in_transmission = true;
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user.start_voice_seq_num = seq_num;
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user.highest_voice_seq_num = seq_num;
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rtp_seq_num = user.rtp_seq_num_offset;
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if let Some(frame) = user.set_active(target) {
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self.outbound_buf.push_back(frame);
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}
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};
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let rtp_time = 480 * rtp_seq_num;
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let rtp = RtpPacket {
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header: RtpFixedHeader {
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padding: false,
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marker: first_in_transmission,
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payload_type: 97,
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seq_num: rtp_seq_num as u16,
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timestamp: rtp_time as u32,
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ssrc: rtp_ssrc,
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csrc_list: Vec::new(),
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extension: None,
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},
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payload: opus_data.to_vec(),
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padding: Vec::new(),
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};
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let frame = Frame::Rtp(MuxedPacket::Rtp(rtp));
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self.outbound_buf.push_back(frame);
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Ok(())
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}
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fn process_packet_from_server(
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&mut self,
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packet: ControlPacket<Clientbound>,
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) -> Result<(), Error> {
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if !self.supports_webrtc() {
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self.outbound_buf.push_back(Frame::Client(packet));
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return Ok(());
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}
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match packet {
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ControlPacket::UDPTunnel(voice) => return self.handle_voice_packet(*voice),
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ControlPacket::UserState(mut message) => {
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let session_id = message.get_session();
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if !self.sessions.contains_key(&session_id) {
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let user = self.allocate_ssrc(session_id);
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message.set_ssrc(user.ssrc);
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}
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self.outbound_buf
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.push_back(Frame::Client((*message).into()));
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}
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ControlPacket::UserRemove(message) => {
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self.free_ssrc(message.get_session());
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self.outbound_buf
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.push_back(Frame::Client((*message).into()));
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}
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other => self.outbound_buf.push_back(Frame::Client(other)),
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};
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Ok(())
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}
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fn process_packet_from_client(
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&mut self,
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packet: ControlPacket<Serverbound>,
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) -> Result<(), Error> {
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match packet {
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ControlPacket::Authenticate(mut message) => {
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println!("MSG Authenticate: {:?}", {
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let mut message = message.clone();
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if message.get_password() != "" {
|
|
message.set_password("{{snip}}".to_string());
|
|
}
|
|
message
|
|
});
|
|
if message.get_webrtc() {
|
|
// strip webrtc support from the connection (we will be providing it)
|
|
message.clear_webrtc();
|
|
// and make sure opus is marked as supported
|
|
message.set_opus(true);
|
|
self.outbound_buf
|
|
.push_back(Frame::Server((*message).into()));
|
|
|
|
self.setup_ice()?;
|
|
} else {
|
|
self.outbound_buf
|
|
.push_back(Frame::Server((*message).into()));
|
|
}
|
|
}
|
|
ControlPacket::WebRTC(mut message) => {
|
|
println!("Got WebRTC: {:?}", message);
|
|
if let Some((_, stream)) = &mut self.ice {
|
|
if let (Ok(ufrag), Ok(pwd)) = (
|
|
CString::new(message.take_ice_ufrag()),
|
|
CString::new(message.take_ice_pwd()),
|
|
) {
|
|
stream.set_remote_credentials(ufrag, pwd);
|
|
}
|
|
// FIXME trigger ICE-restart if required
|
|
// FIXME store and use remote dtls fingerprint
|
|
}
|
|
}
|
|
ControlPacket::IceCandidate(mut message) => {
|
|
let candidate = message.take_content();
|
|
println!("Got ice candidate: {:?}", candidate);
|
|
if let Some((_, stream)) = &mut self.ice {
|
|
match format!("candidate:{}", candidate).parse() {
|
|
Ok(SdpAttribute::Candidate(candidate)) => {
|
|
stream.add_remote_candidate(candidate)
|
|
}
|
|
Ok(_) => unreachable!(),
|
|
Err(err) => {
|
|
return Err(io::Error::new(
|
|
io::ErrorKind::Other,
|
|
format!("Error parsing ICE candidate: {}", err),
|
|
)
|
|
.into());
|
|
}
|
|
}
|
|
}
|
|
}
|
|
ControlPacket::TalkingState(message) => {
|
|
self.target = if message.has_target() {
|
|
Some(message.get_target() as u8)
|
|
} else {
|
|
None
|
|
};
|
|
}
|
|
other => {
|
|
self.outbound_buf.push_back(Frame::Server(other));
|
|
}
|
|
};
|
|
Ok(())
|
|
}
|
|
|
|
fn process_rtp_packet(&mut self, buf: &[u8]) {
|
|
match self.rtp_reader.read_packet(&mut &buf[..]) {
|
|
Ok(MuxedPacket::Rtp(rtp)) => {
|
|
if let Some(target) = self.target {
|
|
// FIXME derive mumble seq_num from rtp timestamp to properly handle
|
|
// packet reordering and loss (done). But maybe keep it low?
|
|
let seq_num = rtp.header.timestamp / 480;
|
|
|
|
let voice_packet = VoicePacket::Audio {
|
|
_dst: std::marker::PhantomData::<Serverbound>,
|
|
target,
|
|
session_id: (),
|
|
seq_num: seq_num.into(),
|
|
payload: VoicePacketPayload::Opus(rtp.payload.into(), false),
|
|
position_info: None,
|
|
};
|
|
|
|
self.outbound_buf
|
|
.push_back(Frame::Server(voice_packet.into()));
|
|
}
|
|
}
|
|
Ok(MuxedPacket::Rtcp(_rtcp)) => {}
|
|
Err(_err) => {} // FIXME maybe not silently drop the error?
|
|
}
|
|
}
|
|
}
|
|
|
|
impl Future for Connection {
|
|
type Output = Result<(), Error>;
|
|
|
|
fn poll(mut self: Pin<&mut Self>, cx: &mut Context) -> Poll<Result<(), Error>> {
|
|
'poll: loop {
|
|
if let Some((ref mut agent, _)) = self.ice {
|
|
pin_mut!(agent);
|
|
let _ = agent.poll(cx);
|
|
}
|
|
|
|
ready!(self.as_mut().dispatch_outbound_frames(cx))?;
|
|
|
|
// Check/register voice timeouts
|
|
// Note that this must be ran if any new sessions are added or timeouts are
|
|
// modified as otherwise we may be blocking on IO and won't get notified of
|
|
// timeouts. In particular, this means that it has to always be called if
|
|
// we suspect that we may be blocking on inbound IO (outbound is less critical
|
|
// since any action taken as a result of timeouts will have to wait for it
|
|
// anyway), hence this being positioned above the code for incoming packets below.
|
|
// (same applies to the other futures directly below it)
|
|
for session in self.sessions.values_mut() {
|
|
if let Some(timeout) = &mut session.timeout {
|
|
if let Poll::Ready(()) = timeout.poll_unpin(cx) {
|
|
if let Some(frame) = session.set_inactive() {
|
|
self.outbound_buf.push_back(frame);
|
|
}
|
|
continue 'poll;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Poll ice stream for new candidates
|
|
if self.as_mut().gather_ice_candidates(cx) {
|
|
continue 'poll;
|
|
}
|
|
|
|
// Poll dtls_srtp future if required
|
|
if let Some(ref mut future) = self.dtls_srtp_future {
|
|
pin_mut!(future);
|
|
if let Poll::Ready(mut dtls_srtp) = future.poll(cx)? {
|
|
self.dtls_srtp_future = None;
|
|
|
|
println!("DTLS-SRTP connection established.");
|
|
|
|
dtls_srtp.add_incoming_unknown_ssrcs(self.next_ssrc as usize);
|
|
dtls_srtp.add_outgoing_unknown_ssrcs(self.next_ssrc as usize);
|
|
|
|
self.dtls_srtp = Some(dtls_srtp);
|
|
}
|
|
}
|
|
|
|
// Finally check for incoming packets
|
|
match self.inbound_server.as_mut().poll_next(cx)? {
|
|
Poll::Pending => {}
|
|
Poll::Ready(Some(frame)) => {
|
|
self.process_packet_from_server(frame)?;
|
|
continue 'poll;
|
|
}
|
|
Poll::Ready(None) => return Poll::Ready(Ok(())),
|
|
}
|
|
match self.inbound_client.as_mut().poll_next(cx)? {
|
|
Poll::Pending => {}
|
|
Poll::Ready(Some(frame)) => {
|
|
self.process_packet_from_client(frame)?;
|
|
continue 'poll;
|
|
}
|
|
Poll::Ready(None) => return Poll::Ready(Ok(())),
|
|
}
|
|
if let Some(ref mut dtls_srtp) = self.dtls_srtp {
|
|
pin_mut!(dtls_srtp);
|
|
match dtls_srtp.poll_next(cx)? {
|
|
Poll::Pending => {}
|
|
Poll::Ready(Some(frame)) => {
|
|
self.process_rtp_packet(&frame);
|
|
continue 'poll;
|
|
}
|
|
Poll::Ready(None) => return Poll::Ready(Ok(())),
|
|
}
|
|
}
|
|
|
|
return Poll::Pending;
|
|
}
|
|
}
|
|
}
|
|
|
|
#[derive(Clone)]
|
|
enum Frame {
|
|
Server(ControlPacket<Serverbound>),
|
|
Client(ControlPacket<Clientbound>),
|
|
Rtp(MuxedPacket<RtpPacket, RtcpCompoundPacket<RtcpPacket>>),
|
|
}
|