rustyrtc

A robust WebRTC/RTC engine written in Rust. It includes SDP parsing, ICE, DTLS, SRTP, a full media pipeline (encode/decode/packetize/depacketize), a congestion controller, a custom signaling solution, and a GUI app for testing.

Status: actively evolving; supports secure media transport and custom signaling.

Features (High Level)

• SDP module – Parse/build offers & answers, rtpmap/fmtp helpers.
• ICE module – Connectivity checks, role selection, candidate gathering/pairing.
• DTLS module – Secure handshake and key derivation (wrapping OpenSSL).
• SRTP module – Secure Real-time Transport Protocol (AES/HMAC encryption for media).
• Congestion Controller – Bandwidth estimation and flow control.
• RTP/RTCP modules – Packet handling, headers, SR/RR reports, NACKs/PLI.
• Media Transport – Event loops for packetization/depacketization and media flow.
• Signaling – dedicated Server (signaling_server) and Client (signaling_client) implementation.
• Camera Manager – Capture frames from local devices via OpenCV.
• App/GUI module – eframe/wgpu based desktop app for testing calls.

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Quickstart

Prerequisites

• Rust (stable) — install via https://rustup.rs
• OpenSSL (dev packages) — Required for DTLS/SRTP.
• OpenH264 — Required for video encoding/decoding.
• OpenCV — Required for camera capture.
• Clang/LLVM — Required for bindgen operations.

Build & Run

1. Build the project

# clone
git clone [https://github.com/taller-1-fiuba-rust/25C2-rustyrtc](https://github.com/taller-1-fiuba-rust/25C2-rustyrtc) && cd rustyrtc

# build (release is recommended for video performance)
cargo build --release