File
Blob: firmware/crates/radio-webrtc/src/audio/tests.rs
| 1 | //! Authenticated RTP over an in-memory network with public fixture keys. |
| 2 | use super::Audio; |
| 3 | use radio_core::{music::MAX_OPUS_BYTES, protocol::SILENCE}; |
| 4 | use std::{ |
| 5 | collections::VecDeque, |
| 6 | net::{Ipv4Addr, SocketAddr}, |
| 7 | sync::Arc, |
| 8 | time::{Duration, Instant}, |
| 9 | }; |
| 10 | use str0m::{ |
| 11 | Candidate, Event, Input, Output, Rtc, |
| 12 | crypto::dtls::DtlsCert, |
| 13 | format::{Codec, FormatParams}, |
| 14 | media::{Direction, Frequency, MediaKind, Mid, Pt}, |
| 15 | net::{Protocol, Receive, Transmit}, |
| 16 | rtp::{RtpPacket, Ssrc}, |
| 17 | }; |
| 18 | |
| 19 | struct Endpoint { |
| 20 | rtc: Rtc, |
| 21 | outbound: VecDeque<Transmit>, |
| 22 | received: Vec<RtpPacket>, |
| 23 | } |
| 24 | |
| 25 | impl Endpoint { |
| 26 | fn new(port: u16, now: Instant) -> Self { |
| 27 | let mut config = Rtc::builder() |
| 28 | .set_crypto_provider(Arc::new(str0m_rust_crypto::default_provider())) |
| 29 | .set_dtls_cert(DtlsCert { |
| 30 | certificate: include_bytes!("../../tests/fixtures/certificate.der").to_vec(), |
| 31 | private_key: include_bytes!("../../tests/fixtures/key.der").to_vec(), |
| 32 | }) |
| 33 | .clear_codecs() |
| 34 | .set_rtp_mode(true); |
| 35 | // A non-default dynamic PT detects accidental hard-coded Opus values. |
| 36 | config.codec_config().add_config( |
| 37 | 103.into(), |
| 38 | None, |
| 39 | Codec::Opus, |
| 40 | Frequency::FORTY_EIGHT_KHZ, |
| 41 | Some(2), |
| 42 | FormatParams::default(), |
| 43 | ); |
| 44 | let mut this = Self { |
| 45 | rtc: config.build(now), |
| 46 | outbound: VecDeque::new(), |
| 47 | received: Vec::new(), |
| 48 | }; |
| 49 | let address = SocketAddr::from((Ipv4Addr::LOCALHOST, port)); |
| 50 | this.rtc |
| 51 | .add_local_candidate(Candidate::host(address, "udp").unwrap()); |
| 52 | this.drain(); |
| 53 | this |
| 54 | } |
| 55 | |
| 56 | fn drain(&mut self) { |
| 57 | loop { |
| 58 | match self.rtc.poll_output().unwrap() { |
| 59 | Output::Timeout(_) => break, |
| 60 | Output::Transmit(packet) => self.outbound.push_back(packet), |
| 61 | Output::Event(Event::RtpPacket(packet)) => self.received.push(packet), |
| 62 | _ => {} |
| 63 | } |
| 64 | } |
| 65 | } |
| 66 | |
| 67 | fn receive(&mut self, packet: &Transmit, now: Instant) { |
| 68 | let receive = Receive::new( |
| 69 | Protocol::Udp, |
| 70 | packet.source, |
| 71 | packet.destination, |
| 72 | &packet.contents, |
| 73 | ) |
| 74 | .unwrap(); |
| 75 | let input = Input::Receive(now, receive); |
| 76 | assert!(self.rtc.accepts(&input)); |
| 77 | self.rtc.handle_input(input).unwrap(); |
| 78 | self.drain(); |
| 79 | } |
| 80 | |
| 81 | fn timeout(&mut self, now: Instant) { |
| 82 | self.rtc.handle_input(Input::Timeout(now)).unwrap(); |
| 83 | self.drain(); |
| 84 | } |
| 85 | } |
| 86 | |
| 87 | struct Pair { |
| 88 | sender: Endpoint, |
| 89 | receiver: Endpoint, |
| 90 | audio: Audio, |
| 91 | pt: Pt, |
| 92 | ssrc: Ssrc, |
| 93 | now: Instant, |
| 94 | wire_audio: Vec<Transmit>, |
| 95 | drop_next: bool, |
| 96 | } |
| 97 | |
| 98 | impl Pair { |
| 99 | fn new() -> Self { |
| 100 | let now = Instant::now(); |
| 101 | let mut sender = Endpoint::new(41000, now); |
| 102 | let mut receiver = Endpoint::new(42000, now); |
| 103 | let mut changes = sender.rtc.sdp_api(); |
| 104 | let mid = changes.add_media(MediaKind::Audio, Direction::SendOnly, None, None, None); |
| 105 | let (offer, pending) = changes.apply().unwrap(); |
| 106 | sender.drain(); |
| 107 | let answer = receiver.rtc.sdp_api().accept_offer(offer).unwrap(); |
| 108 | receiver.drain(); |
| 109 | sender.rtc.sdp_api().accept_answer(pending, answer).unwrap(); |
| 110 | sender.drain(); |
| 111 | let pt = sender.rtc.media(mid).unwrap().remote_pts()[0]; |
| 112 | let ssrc = sender |
| 113 | .rtc |
| 114 | .direct_api() |
| 115 | .stream_tx_by_mid(mid, None) |
| 116 | .unwrap() |
| 117 | .ssrc(); |
| 118 | let mut this = Self { |
| 119 | sender, |
| 120 | receiver, |
| 121 | audio: Audio::new(mid), |
| 122 | pt, |
| 123 | ssrc, |
| 124 | now, |
| 125 | wire_audio: Vec::new(), |
| 126 | drop_next: false, |
| 127 | }; |
| 128 | for _ in 0..5000 { |
| 129 | this.progress(); |
| 130 | if this.sender.rtc.is_connected() && this.receiver.rtc.is_connected() { |
| 131 | return this; |
| 132 | } |
| 133 | } |
| 134 | panic!("public-fixture DTLS handshake timed out"); |
| 135 | } |
| 136 | |
| 137 | fn progress(&mut self) { |
| 138 | self.now += Duration::from_millis(1); |
| 139 | self.sender.timeout(self.now); |
| 140 | self.receiver.timeout(self.now); |
| 141 | for _ in 0..100 { |
| 142 | if self.sender.outbound.is_empty() && self.receiver.outbound.is_empty() { |
| 143 | return; |
| 144 | } |
| 145 | while let Some(packet) = self.sender.outbound.pop_front() { |
| 146 | let is_audio = |
| 147 | packet.contents[0] & 0xc0 == 0x80 && !(192..=223).contains(&packet.contents[1]); |
| 148 | let drop = is_audio && std::mem::take(&mut self.drop_next); |
| 149 | if !drop { |
| 150 | self.receiver.receive(&packet, self.now); |
| 151 | } |
| 152 | if is_audio { |
| 153 | self.wire_audio.push(packet); |
| 154 | } |
| 155 | } |
| 156 | while let Some(packet) = self.receiver.outbound.pop_front() { |
| 157 | self.sender.receive(&packet, self.now); |
| 158 | } |
| 159 | } |
| 160 | panic!("in-memory network did not quiesce"); |
| 161 | } |
| 162 | |
| 163 | fn send(&mut self, pts_ms: u32, bytes: &[u8]) { |
| 164 | self.audio |
| 165 | .write(&mut self.sender.rtc, pts_ms, bytes) |
| 166 | .unwrap(); |
| 167 | self.sender.drain(); |
| 168 | for _ in 0..20 { |
| 169 | self.progress(); |
| 170 | } |
| 171 | } |
| 172 | } |
| 173 | |
| 174 | #[test] |
| 175 | fn opus_payload_ownership_identity_and_clock_survive_rollover() { |
| 176 | let mut pair = Pair::new(); |
| 177 | pair.audio.next_seq = 65_534.into(); |
| 178 | // The second packet crosses the 32-bit RTP timestamp boundary. |
| 179 | let first_pts = 89_478_480_u32; |
| 180 | let packets = [ |
| 181 | vec![0xf8; MAX_OPUS_BYTES], |
| 182 | SILENCE.to_vec(), |
| 183 | vec![0xfc; 240], |
| 184 | ]; |
| 185 | for (index, expected) in packets.iter().enumerate() { |
| 186 | let mut producer_buffer = expected.clone(); |
| 187 | pair.audio |
| 188 | .write( |
| 189 | &mut pair.sender.rtc, |
| 190 | first_pts + index as u32 * 20, |
| 191 | &producer_buffer, |
| 192 | ) |
| 193 | .unwrap(); |
| 194 | // The caller may immediately refill the music buffer, even before the |
| 195 | // str0m queue is flushed. The transmitted bytes must remain unchanged. |
| 196 | producer_buffer.fill(0); |
| 197 | pair.sender.drain(); |
| 198 | for _ in 0..20 { |
| 199 | pair.progress(); |
| 200 | } |
| 201 | } |
| 202 | assert_eq!(pair.receiver.received.len(), packets.len()); |
| 203 | for (index, (received, expected)) in pair.receiver.received.iter().zip(&packets).enumerate() { |
| 204 | assert_eq!(received.payload.as_ref(), expected); |
| 205 | assert_eq!(received.header.payload_type, pair.pt); |
| 206 | assert_eq!(received.header.ssrc, pair.ssrc); |
| 207 | assert_eq!(received.header.sequence_number, (65_534 + index) as u16); |
| 208 | assert_eq!( |
| 209 | received.header.timestamp, |
| 210 | (first_pts + index as u32 * 20).wrapping_mul(48) |
| 211 | ); |
| 212 | assert!(!received.header.marker); |
| 213 | } |
| 214 | assert_eq!( |
| 215 | pair.receiver.received[0].header.ext_vals.mid, |
| 216 | Some(pair.audio.mid) |
| 217 | ); |
| 218 | assert_eq!(pair.wire_audio.len(), packets.len()); |
| 219 | // 1500-byte IPv4 path minus the 20-byte IP and 8-byte UDP headers. |
| 220 | assert!( |
| 221 | pair.wire_audio |
| 222 | .iter() |
| 223 | .all(|packet| packet.contents.len() <= 1472) |
| 224 | ); |
| 225 | assert!(pair.wire_audio[0].contents.len() > MAX_OPUS_BYTES); |
| 226 | } |
| 227 | |
| 228 | #[test] |
| 229 | fn loss_and_replay_do_not_repeat_audio_or_reuse_sequences() { |
| 230 | let mut pair = Pair::new(); |
| 231 | let start = *pair.audio.next_seq; |
| 232 | pair.send(u32::MAX - 15, SILENCE); |
| 233 | pair.drop_next = true; |
| 234 | pair.send(4, &[0xf8; 240]); |
| 235 | pair.send(24, SILENCE); |
| 236 | assert_eq!(pair.receiver.received.len(), 2); |
| 237 | assert_eq!( |
| 238 | pair.receiver.received[1].header.sequence_number, |
| 239 | (start + 2) as u16 |
| 240 | ); |
| 241 | assert_eq!( |
| 242 | pair.receiver.received[1] |
| 243 | .header |
| 244 | .timestamp |
| 245 | .wrapping_sub(pair.receiver.received[0].header.timestamp), |
| 246 | 1920 |
| 247 | ); |
| 248 | |
| 249 | // The lost packet is still unseen by SRTP: a forged authentication tag |
| 250 | // must not be accepted as fresh audio. |
| 251 | let lost = &pair.wire_audio[1]; |
| 252 | let mut corrupted = lost.contents.to_vec(); |
| 253 | *corrupted.last_mut().unwrap() ^= 1; |
| 254 | pair.receiver.receive( |
| 255 | &Transmit { |
| 256 | proto: lost.proto, |
| 257 | source: lost.source, |
| 258 | destination: lost.destination, |
| 259 | contents: corrupted.into(), |
| 260 | }, |
| 261 | pair.now, |
| 262 | ); |
| 263 | assert_eq!(pair.receiver.received.len(), 2); |
| 264 | |
| 265 | // Replaying a valid encrypted packet cannot deliver its Opus payload again. |
| 266 | pair.receiver.receive(&pair.wire_audio[0], pair.now); |
| 267 | assert_eq!(pair.receiver.received.len(), 2); |
| 268 | let next = pair.audio.next_seq; |
| 269 | assert!(pair.audio.write(&mut pair.sender.rtc, 44, &[]).is_err()); |
| 270 | assert!( |
| 271 | pair.audio |
| 272 | .write(&mut pair.sender.rtc, 44, &[0; MAX_OPUS_BYTES + 1]) |
| 273 | .is_err() |
| 274 | ); |
| 275 | assert_eq!(pair.audio.next_seq, next); |
| 276 | pair.send(44, &[0xfc; 240]); |
| 277 | assert_eq!( |
| 278 | pair.receiver.received[2].header.sequence_number, |
| 279 | (start + 3) as u16 |
| 280 | ); |
| 281 | |
| 282 | pair.sender.rtc.close().unwrap(); |
| 283 | pair.sender.drain(); |
| 284 | let next = pair.audio.next_seq; |
| 285 | assert!(pair.audio.write(&mut pair.sender.rtc, 64, SILENCE).is_err()); |
| 286 | assert_eq!(pair.audio.next_seq, next); |
| 287 | } |
| 288 | |
| 289 | #[test] |
| 290 | fn audio_requires_an_authenticated_transport() { |
| 291 | let mut endpoint = Endpoint::new(43000, Instant::now()); |
| 292 | let mut audio = Audio::new(Mid::new()); |
| 293 | let next = audio.next_seq; |
| 294 | assert!(audio.write(&mut endpoint.rtc, 0, SILENCE).is_err()); |
| 295 | assert_eq!(audio.next_seq, next); |
| 296 | assert!(endpoint.outbound.is_empty()); |
| 297 | } |