shadow_protocols/
webrtc.rs

1//! WebRTC Protocol Mimicry
2//! 
3//! This module mimics WebRTC video conferencing traffic to disguise P2P communication.
4//! It generates RTP packets with realistic timing, size distributions, and codec characteristics.
5
6use shadow_core::{Packet, PacketType, PeerInfo, Transport, TransportStats, CoverProtocol};
7use shadow_core::error::{Result, ShadowError};
8use bytes::Bytes;
9use async_trait::async_trait;
10use tokio::sync::mpsc;
11use std::time::{Duration, Instant};
12use rand::Rng;
13
14/// RTP packet structure (simplified)
15#[derive(Debug, Clone)]
16pub struct RTPPacket {
17    /// RTP version (always 2)
18    pub version: u8,
19    /// Padding flag
20    pub padding: bool,
21    /// Extension flag
22    pub extension: bool,
23    /// CSRC count
24    pub csrc_count: u8,
25    /// Marker bit
26    pub marker: bool,
27    /// Payload type (96-127 for dynamic)
28    pub payload_type: u8,
29    /// Sequence number
30    pub sequence: u16,
31    /// Timestamp (90kHz clock for video)
32    pub timestamp: u32,
33    /// SSRC identifier
34    pub ssrc: u32,
35    /// Payload data
36    pub payload: Vec<u8>,
37}
38
39impl RTPPacket {
40    /// Create a new RTP packet
41    pub fn new(sequence: u16, timestamp: u32, ssrc: u32, payload: Vec<u8>) -> Self {
42        Self {
43            version: 2,
44            padding: false,
45            extension: false,
46            csrc_count: 0,
47            marker: false,
48            payload_type: 96, // VP8/VP9
49            sequence,
50            timestamp,
51            ssrc,
52            payload,
53        }
54    }
55
56    /// Serialize to bytes (RTP header format)
57    pub fn to_bytes(&self) -> Vec<u8> {
58        let mut bytes = Vec::with_capacity(12 + self.payload.len());
59        
60        // Byte 0: V(2), P(1), X(1), CC(4)
61        bytes.push(
62            (self.version << 6) |
63            ((self.padding as u8) << 5) |
64            ((self.extension as u8) << 4) |
65            self.csrc_count
66        );
67        
68        // Byte 1: M(1), PT(7)
69        bytes.push(
70            ((self.marker as u8) << 7) |
71            self.payload_type
72        );
73        
74        // Bytes 2-3: Sequence number
75        bytes.extend_from_slice(&self.sequence.to_be_bytes());
76        
77        // Bytes 4-7: Timestamp
78        bytes.extend_from_slice(&self.timestamp.to_be_bytes());
79        
80        // Bytes 8-11: SSRC
81        bytes.extend_from_slice(&self.ssrc.to_be_bytes());
82        
83        // Payload
84        bytes.extend_from_slice(&self.payload);
85        
86        bytes
87    }
88
89    /// Parse from bytes
90    pub fn from_bytes(data: &[u8]) -> Result<Self> {
91        if data.len() < 12 {
92            return Err(ShadowError::InvalidPacket("RTP packet too short".into()));
93        }
94
95        let version = (data[0] >> 6) & 0x03;
96        let padding = (data[0] & 0x20) != 0;
97        let extension = (data[0] & 0x10) != 0;
98        let csrc_count = data[0] & 0x0F;
99        
100        let marker = (data[1] & 0x80) != 0;
101        let payload_type = data[1] & 0x7F;
102        
103        let sequence = u16::from_be_bytes([data[2], data[3]]);
104        let timestamp = u32::from_be_bytes([data[4], data[5], data[6], data[7]]);
105        let ssrc = u32::from_be_bytes([data[8], data[9], data[10], data[11]]);
106        
107        let payload = data[12..].to_vec();
108
109        Ok(Self {
110            version,
111            padding,
112            extension,
113            csrc_count,
114            marker,
115            payload_type,
116            sequence,
117            timestamp,
118            ssrc,
119            payload,
120        })
121    }
122}
123
124/// WebRTC transport that mimics video conferencing traffic
125pub struct WebRTCTransport {
126    /// Local SSRC
127    ssrc: u32,
128    /// Sequence number counter
129    sequence: u16,
130    /// Timestamp counter (90kHz clock)
131    timestamp: u32,
132    /// Start time for timing calculations
133    start_time: Instant,
134    /// Packet sender channel
135    tx: mpsc::UnboundedSender<Packet>,
136    /// Packet receiver channel
137    rx: mpsc::UnboundedReceiver<Packet>,
138    /// Statistics
139    stats: TransportStats,
140    /// Enable dummy traffic
141    dummy_traffic: bool,
142    /// Target bitrate (bytes per second)
143    target_bitrate: u64,
144}
145
146impl WebRTCTransport {
147    /// Create new WebRTC transport
148    pub fn new() -> Self {
149        let (tx, rx) = mpsc::unbounded_channel();
150        let mut rng = rand::thread_rng();
151        
152        Self {
153            ssrc: rng.gen(),
154            sequence: 0,
155            timestamp: 0,
156            start_time: Instant::now(),
157            tx,
158            rx,
159            stats: TransportStats::default(),
160            dummy_traffic: true,
161            target_bitrate: 500_000, // 500 KB/s default
162        }
163    }
164
165    /// Create RTP packet with shadow network data embedded
166    fn create_rtp_packet(&mut self, data: &[u8]) -> RTPPacket {
167        // Simulate video frame packetization
168        let max_payload = 1200; // Typical MTU consideration
169        let chunks: Vec<&[u8]> = data.chunks(max_payload).collect();
170        
171        // For now, just use first chunk (in production, handle fragmentation)
172        let payload = if !chunks.is_empty() {
173            chunks[0].to_vec()
174        } else {
175            vec![]
176        };
177
178        self.sequence = self.sequence.wrapping_add(1);
179        
180        // Update timestamp (90kHz clock, ~30fps = 3000 ticks per frame)
181        self.timestamp = self.timestamp.wrapping_add(3000);
182        
183        RTPPacket::new(self.sequence, self.timestamp, self.ssrc, payload)
184    }
185
186    /// Generate realistic packet sizes (following VP8/VP9 distribution)
187    fn generate_packet_size(&self) -> usize {
188        let mut rng = rand::thread_rng();
189        
190        // VP8 packet sizes follow a distribution:
191        // - Keyframes: 5000-15000 bytes
192        // - Delta frames: 500-3000 bytes
193        // - Small packets: 100-500 bytes
194        
195        let r: f64 = rng.gen();
196        if r < 0.03 {
197            // Keyframe (3% of packets)
198            rng.gen_range(5000..15000)
199        } else if r < 0.7 {
200            // Normal delta frame (67% of packets)
201            rng.gen_range(500..3000)
202        } else {
203            // Small packet (30% of packets)
204            rng.gen_range(100..500)
205        }
206    }
207
208    /// Add realistic jitter (following network characteristics)
209    async fn add_jitter(&self) {
210        // Generate jitter value (not the RNG itself) before async block
211        let jitter_ms = {
212            let mut rng = rand::thread_rng();
213            // Gaussian jitter: mean=20ms, stddev=10ms
214            (rng.gen::<f64>() * 10.0 + 20.0).max(0.0) as u64
215        };
216        
217        tokio::time::sleep(Duration::from_millis(jitter_ms)).await;
218    }
219}
220
221impl Default for WebRTCTransport {
222    fn default() -> Self {
223        Self::new()
224    }
225}
226
227#[async_trait]
228impl Transport for WebRTCTransport {
229    fn protocol(&self) -> CoverProtocol {
230        CoverProtocol::WebRTC
231    }
232
233    async fn send(&mut self, packet: Packet, _peer: &PeerInfo) -> Result<()> {
234        // Convert shadow packet to RTP packet
235        let rtp = self.create_rtp_packet(&packet.payload);
236        let rtp_bytes = rtp.to_bytes();
237        
238        // Add realistic jitter
239        self.add_jitter().await;
240        
241        // Update statistics
242        self.stats.packets_sent += 1;
243        self.stats.bytes_sent += rtp_bytes.len() as u64;
244        
245        // In real implementation, send over UDP socket
246        // For now, just store in channel for demonstration
247        self.tx.send(packet).map_err(|e| {
248            ShadowError::Transport(format!("Failed to send packet: {}", e))
249        })?;
250        
251        Ok(())
252    }
253
254    async fn recv(&mut self) -> Result<Packet> {
255        // Receive packet from channel
256        self.rx.recv().await.ok_or_else(|| {
257            ShadowError::Transport("Channel closed".into())
258        })
259    }
260
261    fn stats(&self) -> TransportStats {
262        self.stats.clone()
263    }
264
265    async fn start_background_traffic(&mut self) -> Result<()> {
266        self.dummy_traffic = true;
267        
268        // Spawn background task for dummy traffic
269        // In production, this would generate continuous RTP packets
270        
271        Ok(())
272    }
273
274    async fn stop_background_traffic(&mut self) -> Result<()> {
275        self.dummy_traffic = false;
276        Ok(())
277    }
278
279    async fn set_bandwidth_limit(&mut self, bytes_per_sec: u64) -> Result<()> {
280        self.target_bitrate = bytes_per_sec;
281        Ok(())
282    }
283}
284
285#[cfg(test)]
286mod tests {
287    use super::*;
288
289    #[test]
290    fn test_rtp_packet_serialization() {
291        let packet = RTPPacket::new(12345, 67890, 11111, vec![1, 2, 3, 4, 5]);
292        let bytes = packet.to_bytes();
293        let parsed = RTPPacket::from_bytes(&bytes).unwrap();
294        
295        assert_eq!(parsed.sequence, 12345);
296        assert_eq!(parsed.timestamp, 67890);
297        assert_eq!(parsed.ssrc, 11111);
298        assert_eq!(parsed.payload, vec![1, 2, 3, 4, 5]);
299    }
300
301    #[tokio::test]
302    async fn test_webrtc_transport() {
303        let mut transport = WebRTCTransport::new();
304        
305        assert_eq!(transport.protocol(), CoverProtocol::WebRTC);
306        
307        let packet = Packet::new(
308            PacketType::Data,
309            None,
310            None,
311            Bytes::from(vec![1, 2, 3, 4]),
312        );
313        
314        let peer = PeerInfo::new(
315            shadow_core::PeerId::random(),
316            vec!["127.0.0.1:9000".to_string()],
317            [0u8; 32],
318            [0u8; 32],
319        );
320        
321        transport.send(packet, &peer).await.unwrap();
322        
323        let stats = transport.stats();
324        assert_eq!(stats.packets_sent, 1);
325    }
326}
shadow_protocols/webrtc.rs

shadow_protocols/
webrtc.rs
