Rodin

Real-time Communication Infrastructure Engineer

Building RTC systems with Go + WebRTC

I am a backend engineer specializing in real-time communication infrastructure.

My work focuses on designing and building systems such as:

• WebRTC signaling services
• real-time call gateways
• session orchestration layers
• distributed backend services
• streaming and media systems

I have extensive experience building Go-based RTC systems integrating WebRTC, gRPC, WebSocket, and MQTT.

I am particularly interested in solving problems around:

• weak network communication
• WebRTC audio processing
• real-time system architecture
• backend infrastructure for RTC platforms

Designing backend infrastructure for real-time video/audio communication.

• Pion WebRTC integration
• ICE / STUN / TURN
• signaling architecture
• session lifecycle management

• Go microservices
• gRPC communication
• event-driven messaging
• backend orchestration

• FFmpeg integration
• RTP / RTSP / RTMP
• audio processing pipelines

• Go (primary)
• PHP
• JavaScript
• Java
• Python

• WebRTC
• RTP
• ICE
• TURN

• gRPC
• WebSocket
• MQTT
• Protobuf

• Docker
• Linux
• FFmpeg

A production-oriented real-time call gateway built with Go.

Features:

• WebRTC signaling
• session lifecycle management
• Pion WebRTC integration
• WebSocket / gRPC communication
• MQTT event propagation
• weak network testing support

A Go-based media processing service.

Supports:

• RTMP
• RTSP
• HLS
• FLV
• FFmpeg orchestration
• streaming task control

A Go-based backend admin framework.

Features:

• authentication
• admin panel
• modular backend architecture

I maintain research notes and experiments related to WebRTC and backend infrastructure.

I maintain technical research notes about WebRTC infrastructure, backend systems, and real-time communication engineering.

• WebRTC stability in complex LAN environments
  如何在复杂的局域网网络情况下，保证视频电话功能的稳定使用？

• Testing WebRTC calls in multi-layer LAN environments
  搭建多级局域网测试 WEBRTC 音视频通话情况

• Android 14 WebRTC call disconnection analysis with coturn
  Android14 音视频在配置 coturn 中继服务情况下通话自动断开问题分析

• Server-side noise suppression using WebRTC + FFmpeg + RNN
  WEBRTC + FFMPEG + RNN 服务端实现通话背景噪声消除

• Debugging Android WebRTC audio 3A using AEC_DUMP
  利用 AEC_DUMP 工具调试安卓设备 WebRTC 音频 3A 算法

• Building and integrating WebRTC audio-processing module
  WebRTC audio-processing 模块移植与编译

• Noise reduction and echo cancellation techniques for better call quality
  如何降噪、消除回声，从而获得更好的通话体验？

• Debugging a Golang production deadlock issue
  记一次线上 Golang 并发死锁问题排查

• Compiling OpenHarmony 5.0.1 WebRTC in Docker on Mac M2
  Mac M2 Docker 环境编译 OpenHarmony WebRTC

• Deploying SeamlessM4T-v2 with Docker
  Docker 部署 SeamlessM4T-v2 模型

• Email [email protected]

• Twitter https://twitter.com/rodin990

⭐ If you find my projects useful, feel free to star them.