A very simple, high performance, edge WebRTC SFU

SIP Stack Rust library for building SIP applications

一个使用纯Rust编写的视频监控服务，含信令服务与流媒体服务，海康、大华均测试兼容

A high-performance implementation of WebRTC.

peer to peer audio calls with iroh

Rust reader for Realtime Transport Protocol packet structure

Lightweight, multipurpose rust library for handling RTP streams.

RTP streams analysis and visualization tool

Rust library for parsing or writing RTP packets as specified RFC 3550 and other RFCs

Distributed stream analysis and visualization tool for RTP and MPEGTS protocols

Rust library for SIP calling and RTP media — register with a trunk or accept calls as a SIP server. Decoded PCM audio via crossbeam channels.

Self-hosted voice gateway — connects SIP phone calls to WebSocket audio and REST API. Single binary, stateless, Twilio-compatible WebSocket framing.

Rust parser for SMPTE 2022-1 packet headers

Rust CLI based SIP softphone

Real-time audio streaming system using RTP transport (RFC 3550) and Opus encoding (RFC 6716). This project demonstrates end-to-end systems architecture from audio capture through network transport to playback, with focus on production concerns: network resilience, observability, and audio intelligence through ML model integration.

RoQ (Pronounced as Rock) is an RTP over Quic library (An alternative to WebRTC).

Rust #no-std optimized (Apple) MIDI implementation for embedded devices.

Passive VoIP recorder that captures SIP/RTP calls from network traffic and saves them as stereo WAV files. Built with Rust for high performance.

A high-performance RTP (Real-time Transport Protocol) proxy server implemented in Rust with gRPC control interface.