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rewrite README with full protocol details, wire format specs, security mechanisms

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- Wire format byte-level specs for webrtc/ipcam/rtpish/shim
- Full-frame AES-GCM encryption mechanism explained with before/after diagrams
- Anti-DPI timing model: decoy delay, audio ticker, STUN consent
- Decoy type matrix with trigger modes
- 7-dimension DPI resistance analysis table with scores
- CLI reference split into core/traffic/webrtc/ipcam/debug sections
- Metrics reference table
- SpeedTest demo with CRC32 verification results
- Chinese + English bilingual documentation
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README.md
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@@ -1,26 +1,80 @@
# CamoStream
Network traffic obfuscation tool that disguises real UDP/TCP traffic as legitimate video streaming protocols. Supports multiple wire formats for different camouflage scenarios.
Network traffic obfuscation tool that disguises real UDP/TCP traffic as legitimate video streaming protocols. Zero external dependencies, single binary, multiple camouflage strategies.
> For authorized internal security testing only.
## Features
- **4 种伪装模式**: WebRTC 视频通话 / 监控摄像头 / 通用 RTP / 原始隧道
- **全帧加密**: AES-GCM 整体加密 shim 头 + 载荷，消除协议指纹
- **多维诱饵注入**: shim-decoy / RTCP SR+SDES / RTCP RR / RTP keepalive / STUN Binding
- **协议仿真**: DTLS 1.2 握手、SRTP auth tag、H.264 FU-A 分片、GOP I/P 帧
- **码率整形**: Token Bucket 令牌桶限速
- **PCAP 调试**: 带大小上限的抓包输出
- **expvar 指标**: 实时流量/诱饵/会话统计
## Wire Modes
| Mode | Disguise As | Protocol Stack |
|------|-------------|----------------|
| `rtpish` | Generic RTP video | RTP(12B) + Shim + Payload |
| `webrtc` | WebRTC video call | SRTP(24B) + Extensions + Auth Tag + Opus Audio + Compound RTCP + STUN |
| `ipcam` | Surveillance camera | H.264 FU-A over RTP + SPS/PPS + GOP I/P frames |
| `shim` | Raw tunnel (no disguise) | ShimHeader + Payload |
### `webrtc` — WebRTC 视频通话伪装（推荐）
将流量伪装为一个完整的 WebRTC 1:1 视频通话会话。
**协议栈仿真**:
- DTLS 1.2 握手（ClientHello / ServerHello / ChangeCipherSpec / Finished）
- SRTP 24 字节头：V=2, X=1, 含 `0xBEDE` 一字节扩展
- abs-send-time (id=3, 3 字节, NTP 6.18 定点)
- transport-cc (id=5, 2 字节, 跨流共享递增计数器)
- 10 字节 SRTP HMAC-SHA1-80 认证标签
- Opus 音频流 (PT=111, 独立 SSRC, 50 pps / 20ms 间隔)
- 视频流 (PT=96, 动态载荷类型)
- Compound RTCP: SR(28B) + SDES(CNAME) + SRTCP index + auth tag (RFC 3550)
- STUN Binding Request/Response 含 FINGERPRINT (CRC32 XOR 0x5354554E)
- ICE consent freshness 每 5 秒
**线上包格式**:
```
[SRTP Header 24B][Nonce 12B][AES-GCM(ShimHeader 20B + Payload)][Auth Tag 10B]
```
### `ipcam` — 监控摄像头伪装
模拟 Hikvision / Dahua 风格的 H.264 IP 摄像头 RTP 视频流。
**协议栈仿真**:
- H.264 over RTP (RFC 6184)
- FU-A 分片 (NAL type 28): IDR indicator 0x7C / non-IDR 0x5C
- STAP-A (NAL type 24): SPS (High profile, Level 4.0) + PPS 周期发送
- GOP 状态机: I 帧间隔可配 (默认 50 帧 = 2 秒 @25fps)
- I 帧 burst ~120KB / P 帧 ~15KB，VBR 波动 ±30%
- 90kHz 时钟, 时间戳增量 = 90000 / fps
**线上包格式**:
```
[RTP Header 12B][FU-A Indicator 1B][FU Header 1B][Nonce 12B][AES-GCM(Shim + Payload)]
```
### `rtpish` — 通用 RTP 伪装
基础 RTP 封装，适用于一般场景。
```
[RTP Header 12B][Nonce 12B][AES-GCM(ShimHeader 20B + Payload)]
```
### `shim` — 原始隧道（无伪装）
仅 shim 头 + 载荷，无 RTP 包装。用于 TCP 模式或调试。
## Architecture
```
[App] --UDP--> [CamoStream Client :37001]
|
| encrypted + disguised tunnel
v
[CamoStream Server :39001] --UDP--> [Real Backend :18081]
Encrypted + Disguised Tunnel
┌──────────────────────┐
[User App] ──UDP──▶ │ CamoStream Client │ ════════════════▶ │ CamoStream Server │ ──UDP──▶ [Backend]
:any │ :37001 │ looks like │ :39001 │ :18081
│ encode + encrypt │ video stream │ decrypt + decode │
└──────────────────────┘ └──────────────────────┘
```
## Build
@@ -29,14 +83,14 @@ Network traffic obfuscation tool that disguises real UDP/TCP traffic as legitima
go build -o camostream .
```
Requires Go 1.24+, zero external dependencies (stdlib only).
Go 1.24+, 零外部依赖（纯标准库）。
## Quick Start
### WebRTC Mode (Recommended)
### WebRTC Mode
```bash
# Server side
# Server
./camostream -role=server -mode=udp -wire=webrtc \
-listen=:39001 -forward=127.0.0.1:18081 \
-bitrate-mbps=20 -fps=30 \
@@ -44,7 +98,7 @@ Requires Go 1.24+, zero external dependencies (stdlib only).
-decoy-rps=5 -rtcp-sr-rps=1 -stun-rps=0.2 \
-metrics=:9100 -log=info
# Client side
# Client
./camostream -role=client -mode=udp -wire=webrtc \
-listen=:37001 -server=<server-ip>:39001 \
-bitrate-mbps=20 -fps=30 \
@@ -53,7 +107,7 @@ Requires Go 1.24+, zero external dependencies (stdlib only).
-metrics=:9101 -log=info
```
### IPCAM Mode (Surveillance Camera)
### IPCAM Mode
```bash
# Server
@@ -74,136 +128,213 @@ Requires Go 1.24+, zero external dependencies (stdlib only).
### TCP Mode
```bash
# Server
./camostream -role=server -mode=tcp -listen=:39001 -forward=127.0.0.1:4141 \
-bitrate-mbps=20 -decoy-rps=10 -aes=0123456789abcdef0123456789abcdef
# Client
./camostream -role=client -mode=tcp -listen=:37001 -server=127.0.0.1:39001 \
-bitrate-mbps=20 -decoy-rps=10 -aes=0123456789abcdef0123456789abcdef
```
## SpeedTest Demo
## SpeedTest & Data Integrity Verification
Built-in CRC32 integrity verification and throughput measurement:
内置 CRC32 完整性校验 + 吞吐量测量的 demo:
```bash
# Start tunnel (webrtc mode)
# 启动隧道
./camostream -role=server -mode=udp -wire=webrtc -listen=:39001 -forward=127.0.0.1:18081 \
-bitrate-mbps=50 -aes=0123456789abcdef0123456789abcdef -dtls=false -log=warn &
./camostream -role=client -mode=udp -wire=webrtc -listen=:37001 -server=127.0.0.1:39001 \
-bitrate-mbps=50 -aes=0123456789abcdef0123456789abcdef -dtls=false -log=warn &
# Start receiver
# 接收端
go run demo/speedtest.go -mode=server -recv=:18081 &
# Run speedtest (200 pps, 1000 byte packets, 10 seconds)
# 发送端 (200 pps, 1000 字节, 10 秒)
go run demo/speedtest.go -mode=client -send=127.0.0.1:37001 -size=1000 -pps=200 -duration=10
```
Or run all wire modes:
一键测试所有 wire 模式: `bash demo/run_speedtest.sh`
```bash
bash demo/run_speedtest.sh
```
### SpeedTest Results (Local, WebRTC Mode)
**SpeedTest 结果 (WebRTC + AES-GCM, 本地回环)**:
```
Sent: 1997 packets, 1.60 Mbps
Received: 818 packets through tunnel
CRC32 OK: 818 FAIL: 0 (100% integrity)
OOO: 0 DUP: 0
Received: 818 packets (through encrypted tunnel)
CRC32 OK: 818 FAIL: 0 Integrity: 100%
Out-of-Order: 0 Duplicates: 0
```
## Security Features
每个包携带: `[4B seq][4B CRC32][8B timestamp][payload]`，接收端逐包校验 CRC32。
### Encryption
- **AES-GCM** encrypts the entire shim header + payload together
- Magic bytes (`0x5C10ADED`) never appear on the wire when encryption is enabled
- Without AES: magic is XOR-masked with session-derived key to prevent static fingerprinting
## Security Mechanisms
### WebRTC Camouflage
- 24-byte SRTP headers with `0xBEDE` extensions (abs-send-time, transport-cc)
- 10-byte SRTP authentication tag on every packet
- Opus audio stream at 50 pps (PT=111) with separate SSRC
- Compound RTCP (SR + SDES with CNAME) per RFC 3550
- STUN Binding Request/Response with FINGERPRINT attribute
- STUN consent freshness every 5 seconds
- DTLS 1.2 handshake simulation at session start (optional)
### Encryption: Full-Frame AES-GCM
### Decoy System
- **Shim decoys**: encrypted fake frames injected at configurable RPS
- **RTCP SR/RR**: realistic sender/receiver reports
- **RTP keepalive**: comfort noise (PT=13) packets
- **STUN Binding**: ICE connectivity checks with proper responses
- Decoys sent with 2-8ms random delay to avoid burst timing fingerprint
```
传统方式 (已弃用): [RTP][Magic 0x5C10ADED][Shim Header][AES-GCM(Payload)]
↑ DPI 一条规则即可指纹识别
## DPI Resistance
当前方式: [RTP][Nonce 12B][AES-GCM(Magic + Shim Header + Payload)]
↑ 整体加密, 无可识别特征
```
Tested with automated 7-dimension analysis:
- 加密范围覆盖 shim 头（含 magic/version/flags/session）+ 用户载荷
- 未启用 AES 时: magic 与 session ID 派生的掩码 XOR，防止静态指纹
| Dimension | Score | Description |
|-----------|-------|-------------|
| Protocol Conformance | 100/100 | All packets classify as RTP/RTCP/STUN |
| Packet Size Distribution | 69/100 | Bimodal (audio small + video large) |
| Timing Analysis | 69/100 | Consistent with video call FPS |
| Entropy | 53/100 | High entropy from AES-GCM |
| RTP Consistency | 100/100 | Perfect sequence/timestamp progression |
| Decoy Coverage | 40/100 | Multiple decoy types present |
| **Overall** | **73.8/100** | **Grade C - Passes basic DPI** |
### Anti-DPI Timing
```
旧模式: [Real Frame][Decoy][RTCP][STUN] ← 微秒级突发, 可被统计检测
t=0 t=0 t=0 t=0
新模式: [Real Frame]...[Decoy]........[Audio]........[RTCP]
t=0 t=+4ms t=+20ms t=+1s
```
- 诱饵帧延迟 2-8ms 随机间隔发送
- 音频 ticker 每 20ms 发一个 Opus 包 (±2ms 抖动)
- STUN consent 每 5s (±1s) 发送
- RTCP 按 RPS 速率均匀分布
### Decoy Types
| 类型 | 格式 | 触发方式 | 用途 |
|------|------|----------|------|
| Shim Decoy | 与真实帧相同格式 | RPS 或百分比 | 混淆真实帧识别 |
| RTCP SR+SDES | RFC 3550 compound | RPS 调度 | 模拟媒体会话报告 |
| RTCP RR | Receiver Report | RPS 调度 | 模拟接收端反馈 |
| RTP Keepalive | PT=13 (CN) | RPS 调度 | 模拟静音检测 |
| STUN Binding | 含 FINGERPRINT | RPS 调度 | 模拟 ICE 连通性 |
## DPI Resistance Analysis
使用 `tests/scripts/analyze_pcap.py` 进行 7 维度自动化分析:
| 维度 | 权重 | 评分 | 说明 |
|------|------|------|------|
| 协议一致性 | 25% | 100/100 | 100% 包被 Wireshark 识别为 RTP/RTCP/STUN |
| 包大小分布 | 20% | 69/100 | 双峰分布 (音频 ~160B + 视频 ~880B), CV=0.93 |
| 时序分析 | 15% | 69/100 | 均值 IAT 13ms, 中位数 18ms, 符合 30fps 视频 |
| 载荷熵值 | 20% | 53/100 | AES-GCM 加密载荷高熵 |
| RTP 序列一致性 | 10% | 100/100 | 3 个 SSRC 流, 序列号 100% 递增 |
| 诱饵覆盖率 | 10% | 40/100 | RTCP + STUN 双类型覆盖 |
| **总分** | **100%** | **73.8/100** | **通过基础 DPI 检测** |
运行分析:
```bash
python3 tests/scripts/analyze_pcap.py <pcap-file> --mode udp --output report.json
```
## CLI Reference
```
-wire shim|rtpish|webrtc|ipcam Wire format (UDP only)
-role server|client|selftest Role
-mode udp|tcp Transport mode
-listen :port Listen address
-server host:port Server address (client mode)
-forward host:port Forward target (server mode)
-aes hex-key AES-GCM 128/192/256 bit key
-bitrate-mbps N Bitrate cap in Mbps
-fps N Frames per second
-decoy-rps N Shim decoy frames per second
-rtcp-sr-rps N RTCP SR decoys per second
-rtcp-rr-rps N RTCP RR decoys per second
-stun-rps N STUN decoys per second
-dtls bool DTLS handshake (webrtc, default true)
-audio-rps N Audio packets/sec (webrtc, default 50)
-ipcam-fps N Camera FPS (ipcam, default 25)
-ipcam-gop N GOP size (ipcam, default 50)
-pcap path PCAP output file
-pcap-max-mb N Max PCAP size in MB
-metrics :port Metrics HTTP endpoint
-log debug|info|warn|error Log level
```
### 核心参数
| 参数 | 默认值 | 说明 |
|------|--------|------|
| `-wire` | `rtpish` | 伪装模式: `shim` / `rtpish` / `webrtc` / `ipcam` |
| `-role` | `server` | 角色: `server` / `client` / `selftest` |
| `-mode` | `udp` | 传输: `udp` / `tcp` |
| `-listen` | `:9001` | 监听地址 |
| `-server` | `127.0.0.1:9001` | 服务端地址 (client 模式) |
| `-forward` | `127.0.0.1:18081` | 转发目标 (server 模式) |
| `-aes` | (空) | AES-GCM 密钥, 16/24/32 字节 hex 编码 |
### 流量控制
| 参数 | 默认值 | 说明 |
|------|--------|------|
| `-bitrate-mbps` | `20` | 码率上限 (Mbps) |
| `-fps` | `60` | 帧率 |
| `-jitter` | `30` | 抖动百分比 (0-100) |
| `-decoy-rps` | `0` | Shim 诱饵帧/秒 |
| `-rtcp-sr-rps` | `0` | RTCP SR 报告/秒 |
| `-rtcp-rr-rps` | `0` | RTCP RR 报告/秒 |
| `-rtpkeep-rps` | `0` | RTP keepalive/秒 |
| `-stun-rps` | `0` | STUN Binding/秒 |
### WebRTC 专用
| 参数 | 默认值 | 说明 |
|------|--------|------|
| `-dtls` | `true` | 会话启动时模拟 DTLS 握手 |
| `-audio-rps` | `50` | Opus 音频包/秒 (20ms 间隔) |
| `-stun-interval` | `5` | STUN consent freshness 间隔 (秒) |
### IPCAM 专用
| 参数 | 默认值 | 说明 |
|------|--------|------|
| `-ipcam-fps` | `25` | 摄像头帧率 |
| `-ipcam-gop` | `50` | GOP 大小 (帧数, 50 = 2 秒 @25fps) |
### 调试
| 参数 | 默认值 | 说明 |
|------|--------|------|
| `-pcap` | (空) | PCAP 输出路径 |
| `-pcap-max-mb` | `50` | PCAP 大小上限 (MB) |
| `-metrics` | `:9100` | Metrics HTTP 端口 (`/debug/vars`) |
| `-log` | `info` | 日志级别: `debug` / `info` / `warn` / `error` |
| `-showdrop` | `false` | 日志中显示诱饵丢弃信息 |
## Metrics
Available at `http://host:port/debug/vars`:
`http://host:port/debug/vars` 返回 JSON:
```
bytes_up, bytes_down, frames_up, frames_down,
decoy_dropped, sessions_active, shim_decoy_sent,
rtcp_sr_sent, rtcp_rr_sent, rtp_keepalive_sent,
stun_sent, dtls_handshake_sent, audio_packets_sent
| 指标 | 说明 |
|------|------|
| `bytes_up` / `bytes_down` | 上/下行字节数 |
| `frames_up` / `frames_down` | 上/下行帧数 |
| `decoy_dropped` | 丢弃的诱饵帧数 |
| `shim_decoy_sent` | 发送的 shim 诱饵数 |
| `rtcp_sr_sent` / `rtcp_rr_sent` | RTCP 报告发送数 |
| `rtp_keepalive_sent` | RTP keepalive 发送数 |
| `stun_sent` | STUN 请求发送数 |
| `dtls_handshake_sent` | DTLS 握手完成数 |
| `audio_packets_sent` | 音频包发送数 |
| `sessions_active` | 活跃会话数 |
## Testing
```bash
# 本地 E2E 测试 (无 Docker)
bash tests/scripts/test_e2e.sh
# PCAP 深度分析
python3 tests/scripts/analyze_pcap.py <pcap> --mode udp
# Docker Compose 集成测试
cd tests && docker compose up --build --abort-on-container-exit
# 全套测试
bash tests/run_all_tests.sh
```
## Project Structure
```
├── main.go Core framework, UDP/TCP client/server, CLI
├── crypto.go sealFrame/openFrame (full shim+payload encryption)
├── dtls.go DTLS 1.2 handshake simulation
├── wire.go Unified encode/decode path for all wire formats
├── wire_webrtc.go WebRTC: SRTP, audio ticker, compound RTCP, STUN
├── wire_ipcam.go IPCAM: H.264 FU-A, GOP state, SPS/PPS
├── demo/ SpeedTest demo with CRC32 verification
├── sim/ UDP echo server for testing
└── tests/ E2E tests, PCAP analysis, Docker environment
camostream/
├── main.go 核心框架: UDP/TCP client/server, CLI, metrics, PCAP
├── crypto.go 加密: sealFrame/openFrame, AES-GCM 全帧加密, magic XOR 掩码
├── dtls.go DTLS 1.2 仿真: ClientHello/ServerHello/CCS/Finished
├── wire.go 统一编解码: encodeUDPFrame/decodeUDPFrame, 所有 wire 模式入口
├── wire_webrtc.go WebRTC: SRTP 头, 音频 ticker, compound RTCP, STUN consent
├── wire_ipcam.go IPCAM: H.264 FU-A/STAP-A, GOP 状态机, 帧大小仿真
├── demo/
│ ├── speedtest.go UDP 吞吐测试 + CRC32 完整性校验
│ └── run_speedtest.sh 一键测试所有 wire 模式
├── sim/
│ └── udp_server.go 简易 UDP echo 服务器
└── tests/
├── docker-compose.yml Docker 多容器测试环境
├── Dockerfile 多阶段构建 (golang + alpine + tshark)
├── scripts/
│ ├── test_e2e.sh 本地 E2E 测试 (TCP/UDP/加密/诱饵/PCAP/Metrics)
│ ├── analyze_pcap.py 7 维度 DPI 抵抗评分 (协议/大小/时序/熵/RTP/诱饵)
│ └── test_traffic_stealth.sh tshark 协议层次分析
└── backend/
└── server.py HTTP echo 后端
```
## Disclaimer
This tool is designed for authorized internal security testing only. Traffic obfuscation capabilities must be used in compliance with applicable laws and organizational policies.
This tool is designed for authorized internal security testing only. Features involving traffic disguise and obfuscation must be used in strict compliance with applicable laws and organizational policies.