Official account of Douyu Streaming Development Department, sharing audio and video technology best practices.
This article examines the shortcomings of loss‑based congestion control, explains how shallow and deep buffering affect throughput, and provides a comprehensive, step‑by‑step breakdown of the BBR algorithm—including its network path model, terminology, design overview, control parameters, state machine, and detailed implementation code.
This article describes the design and implementation of a cloud‑native automated web‑page recording system for Douyu advertising, covering the problem background, technology stack, end‑to‑end workflow, Chrome extension tricks, containerization with Docker, database handling, and deployment on Kubernetes, enabling scalable, hands‑free video capture and delivery.
This guide explains how to obtain the WebRTC m79 source, compile it for Windows, Android, and iOS, walk through the basic signaling and peer‑connection workflow, and implement advanced video‑capture and audio‑volume features with custom C++ extensions, while unifying the codebase across platforms.
This article explains the end‑to‑end workflow of Douyu’s fast live streaming system, detailing how FLV tags are converted to RTP packets and back, covering WebRTC’s SDP/ICE/DTLS handshake, FLV and RTP header structures, payload formats for audio (OPUS) and video (H.264), and the server‑side processing pipeline.
This article explains the concept, components, and algorithms of WebRTC's adaptive jitter buffer, detailing how it calculates network, decode, and render delays to ensure smooth video playback while balancing latency and packet loss.
This article explains NetEQ, the adaptive audio jitter buffer and packet‑loss concealment technology used in WebRTC, detailing its architecture, core modules, jitter‑estimation algorithms, decision logic, and speed‑change processing, and shows how a custom server‑side implementation can improve stability and listening experience.
Douyu’s senior streaming engineer Zhou Sha details the company’s self‑developed P2P solution, covering its background, architecture, key technologies such as sub‑streaming, WebRTC, data slicing, SDK design, and the strategies used to boost sharing rates and future roadmap.
This article explains the security risks of early Internet protocols, the evolution of SSL/TLS and DTLS, fundamental cryptographic concepts such as symmetric and asymmetric encryption, hashing, MACs, digital signatures, AES modes and padding, and how these technologies are applied in TLS handshakes, DTLS, and WebRTC.
This article provides a systematic overview of SDP (Session Description Protocol) and its detailed usage in WebRTC, covering protocol structure, media and session level fields, Plan B vs Unified Plan styles, and the extensive a‑line extensions such as codec, ICE, DTLS, and simulcast parameters.
This article explains NetEQ, the adaptive audio jitter buffer and packet‑loss concealment engine used in WebRTC, detailing its architecture, core modules, jitter estimation, buffer management, and speed‑up/slow‑down algorithms that together ensure robust, low‑latency voice quality even under high packet loss.
