This guide shows how to integrate the Zlm4j library into a Spring Boot project to quickly set up RTSP, RTMP, HTTP, and RTP streaming services, configure ports, handle authentication and stream state events, and test the setup with ffmpeg and VLC.
This article walks through the complete process of capturing a Hikvision RTSP camera stream, converting it with FFmpeg, serving it via Nginx (including RTMP/HLS modules), and finally playing the HLS stream in a Vue 3 web page using video.js, while also providing installation scripts and performance tips.
The article surveys the evolution of live streaming toward low‑latency, interactive scenarios and details WebRTC‑based real‑time audio/video system design, covering RTP/UDP transport, FEC and ARQ loss recovery, congestion control, jitter buffering, echo cancellation, edge‑node path optimization, and a multi‑layer architecture with signaling, routing, mixing services for scalable, high‑availability PK deployments.
The article explains the three main WebRTC multi‑party architectures—Mesh, MCU, and SFU—detailing their bandwidth and processing requirements, advantages, disadvantages, and typical use cases, helping developers choose the most suitable solution for low‑latency, high‑quality audio‑video interactions.
This article explains the RTMP protocol’s handshake process, chunking and message formats, details command flows such as connect, createStream, publish and play, and describes how the Nginx‑RTMP module implements live streaming, acceleration, authentication, recording and HLS slicing.
This article explains how the RTMPGateway media server enables audio‑video communication between WeChat mini‑programs and other platforms by converting RTMP streams to RTP, handling handshakes, media encapsulation, audio transcoding, and signaling, while addressing performance and synchronization challenges.
The article outlines core live‑streaming principles, workflow, and major protocols (RTMP, HLS, HTTP‑FLV), then examines Vivo’s product‑launch challenges—massive concurrency, QPS spikes, and content safety—and presents practical solutions such as multi‑level caching, circuit‑breaker degradation, token‑bucket rate limiting to ensure stable, high‑quality streaming.
This article explains WebRTC fundamentals, including SDP, ICE, and signaling, outlines the peer‑to‑peer communication flow, and compares RTMP, direct WebRTC, and media‑server‑based WebRTC solutions for live streaming, highlighting their advantages and drawbacks.
This article traces the evolution of HTTP-based video streaming—from early full‑file downloads through progressive download, pseudo‑streaming, and finally adaptive bitrate technologies like HLS and MPEG‑DASH—explaining key concepts, server implementations, and optimization techniques.
