This guide walks through setting up ZLMediaKit in Docker, installing FFmpeg, adding a Spring Boot backend with configuration and service classes to launch and manage RTMP streams, and provides front‑end HTML/JavaScript for playing the live FLV or HLS streams.
This guide walks through preparing the environment, installing ZLMediaKit and FFmpeg, configuring a Spring Boot backend with streaming services, and provides step‑by‑step usage instructions for pushing and playing live streams via RTMP, FLV, and HLS.
This tutorial walks through deploying SRS in Docker, persisting its configuration, mapping required ports, customizing the rtmp2rtc.conf file, and optionally exposing the WebRTC signaling service via Nginx with HTTPS, providing all commands and configuration snippets needed for a successful setup.
This guide shows how to list local video and audio devices using FFmpeg, push live streams to RTMP or RTSP servers, and view them with VLC, including commands for device selection, codec settings, frame rate, and bitrate adjustments.
The team built a reusable low‑latency live‑streaming platform for the European Cup that replaces HLS with RTMP over QUIC, reuses VOD bandwidth, automates stream control, and cuts playback latency by 20 seconds while dramatically lowering bandwidth costs and delivering stable 1080p 50 fps HDR video.
This guide walks through the complete offline installation of FFmpeg on a CentOS server, covering prerequisite libraries (nasm, yasm, x264), verification of gcc and pkg‑config, compilation steps, troubleshooting missing x264, and finally using FFmpeg to pull an RTSP stream and push it to an RTMP server.
This guide explains the HLS streaming protocol, its advantages and latency drawbacks, and provides step‑by‑step configurations for HTTPS‑based encryption, RTMP publishing, Nginx virtual hosts for HLS and key delivery, plus a Lua script that enforces domain and IP restrictions.
This guide walks through selecting streaming protocols, preparing a high‑performance server, installing OpenResty, compiling the nginx‑ts module, configuring Nginx for RTMP/HLS/DASH, pushing streams with FFmpeg or OBS, and applying security and stability best practices.
Ximalaya improves live‑stream start‑up by measuring first‑frame latency, optimizing push (dynamic bitrate, GOP caching) and pull (RTMP/HTTP‑FLV, HttpDNS, pre‑fetch URLs, progressive component loading, pre‑render SurfaceView, low‑buffer water‑level) across detailed pipeline stages, achieving over 90 % audio and 85 % video instant‑open rates while planning H.265 adoption.
This article explains common audio‑video streaming protocols such as RTMP and HLS, compares their use cases, then dives into WebRTC fundamentals, device detection, media capture, recording, connection setup, codec considerations, and how to display remote streams, providing a comprehensive guide for building real‑time web communication applications.
This article provides a comprehensive overview of the iOS live‑streaming workflow, detailing the six stages—capture, processing, encoding, packaging, network transmission, and playback—along with sample code for video/audio capture, encoding settings, and RTMP transmission.
The article outlines Vivo's live streaming platform evolution, detailing its business scenarios, underlying architecture, key technical challenges such as beautification, messaging, latency, and cost, and the concrete solutions implemented across entertainment, interactive, and corporate event live streams.
This guide walks through the fundamentals of the RTMP protocol, introduces FFmpeg, shows how to download and configure Nginx‑RTMP on Windows, create the necessary directories, start the server, craft a batch script to push a local video stream, and verify playback with VLC.
This article chronicles the development of a new web live streaming solution, detailing the shift from Flash to HTML5 video, the integration of AI voice detection using hark, audio encoding with lamejs, WebRTC-based real‑time communication, and a systematic modular refactor that transformed the project into a reusable frontend framework.
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.
The article explains RTMP’s multiplexed, packetized streaming over TCP, detailing its chunk structure, message types, handshake, and connection workflow, then demonstrates livego’s publishing and pulling processes, discusses typical latency sources, and offers mitigation strategies and reference resources for developers.
This guide walks beginners through setting up a basic live‑streaming pipeline using the open‑source LiveGo server, OBS for RTMP publishing, and flv.js for browser playback, while explaining push‑pull architecture, protocol choices, Docker deployment, and a Redis‑based chat system for live rooms.
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 presents a comprehensive overview of Ctrip's live streaming solution, covering the evolution of live streaming, core streaming principles, encoding and protocol choices such as RTMP, HLS, WebRTC, the front‑end framework built with Android, ReactNative and native modules, encountered integration challenges, and the final architectural summary.
The guide shows how to assemble a low‑latency real‑time video transmission system by connecting a VisionSeed AI camera to a Raspberry Pi 4B, using FFmpeg for OpenMAX‑accelerated H.264 encoding, streaming via an SRS RTMP server over a custom Wi‑Fi hotspot, and tuning playback with FFplay.
iQIYI’s Qixiu platform supports both single‑host RTMP‑based ordinary live streams and multi‑host UDP‑RTP link‑mic sessions by using an enhanced WebRTC (woogeen) stack with a client SDK, load‑aware MCU nodes, and backend services that handle signaling, mixing, transcoding, adaptive bitrate, echo cancellation, and fault‑tolerant deployment, while outlining future improvements such as real‑time MCU switching and mobile network handover.
This article explains the AI live‑streaming architecture used by Liulishuo, covering the basic workflow of push, server, and pull, video/audio encoding and container formats, streaming protocols such as RTMP, HLS and HTTP‑FLV, SEI integration, and provides a step‑by‑step demo using nginx‑rtmp and FFmpeg.
This article explains the complete live‑streaming workflow—from video/audio capture on the broadcaster side, through encoding, transmission, server processing, and playback—while comparing third‑party services with self‑built solutions and providing concrete iOS code examples using AVFoundation, LFLiveKit and GPUImage.
This guide shows how to containerize an Nginx‑RTMP live‑streaming service with Docker, add customizable watermarks, and split the stream to two cloud providers—one with watermark and one without—using ffmpeg, SSL, and HLS playback.
The 58 Group technical salon presented detailed practices on Tencent Cloud video services, real‑time audio/video communication, and mobile live‑streaming SDK implementation, covering architecture, protocols, encoding, troubleshooting, and future directions for improving streaming quality and user experience.
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 explains streaming media fundamentals and compares major live‑streaming protocols—including RTMP, HTTP‑FLV, HLS, HDS, and MPEG‑DASH—detailing their architectures, workflow diagrams, MPD structure, segment handling, and practical steps for building a DASH demo with MP4Box and dash.js.
Live streaming platforms increasingly rely on CDN technology to handle sudden traffic spikes, improve latency and stability, and support various streaming protocols such as RTMP, HLS, and FLV, with a detailed workflow that shows how broadcasters, schedulers, and viewers interact through edge nodes.
This article explains why live streaming platforms rely on Content Delivery Networks, describes the CDN architecture and workflow, and compares the main streaming protocols—RTMP, HLS, and FLV—highlighting their characteristics and impact on latency and stability.
This article describes a comprehensive solution for enabling real‑time audio and video communication between existing WebRTC endpoints and WeChat Mini Programs by introducing a WebRTC Gateway and Streaming Server, detailing architecture, signaling flows, media conversion, performance optimizations, and session reliability mechanisms.
This article introduces the fundamentals of web live streaming, covering typical workflows, video compression concepts, key quality metrics, comparisons of RTMP, RTP, HLS, flv.js and WebRTC, and practical implementation details for mini‑programs with code examples.
This article explains the fundamentals of live streaming, describing the push‑streaming and pull‑streaming processes, key video protocols such as RTMP, WebRTC and UDP, and provides a step‑by‑step guide for configuring OBS to publish and receive streams.
This article walks through the complete web live‑streaming workflow—covering capture sources, video/audio processing, H.264 encoding, RTMP push, transcoding, CDN distribution, HLS playback, pseudo‑fullscreen tricks, adaptive sizing, playback detection, and a custom SDK—providing practical guidance for building robust live video experiences.
Tencent Video Cloud’s Mini‑Program audio/video, built on the LiteAV SDK with RTMP/HTTP‑FLV tags, contrasts WebRTC’s browser‑native RTP/RTCP approach, but a bridge using an RTMP server and WebRTC‑Proxy translates streams, enabling interoperable live‑push/pull, room‑based multi‑person calls and rapid integration via a packaged webrtc‑room component.
This article introduces essential live‑streaming terminology and outlines a step‑by‑step testing process—including test case design, execution, and verification—to ensure that a live streaming application launches and operates reliably.
This guide introduces St-load, a Linux‑based streaming load testing tool, and provides step‑by‑step instructions for installing, compiling, and using its RTMP, HTTP, and HLS testing features, including command‑line examples and parameter explanations for both push and pull scenarios.
This article provides a detailed walkthrough of building an iOS live‑streaming application, covering the end‑to‑end workflow from video/audio capture, encoding, and transmission to playback, while explaining key protocols, container formats, GPUImage processing, and code examples for developers.
This article examines the rapid growth of personal live streaming, outlines major market players, compares cloud provider offerings, and dives into essential technologies such as RTMP variants, source‑side processing, low‑power encoding, fast startup, and methods to reduce stutter for a seamless viewer experience.
This guide walks through the fundamentals of HLS and RTMP protocols, compares their trade‑offs, and provides step‑by‑step instructions for setting up an Nginx‑RTMP server, configuring HLS, recording with WebRTC, and playing live streams in HTML5 video tags.
This article examines Periscope’s live‑streaming architecture, detailing its use of RTMP and HLS protocols, CDN deployment, video/audio encoding choices, QoS measurements, latency, playback smoothness, and mobile power consumption, offering insights for engineers interested in large‑scale streaming systems.
This article walks through mobile video live streaming fundamentals, covering H5 playback, WebRTC recording, HLS streaming, RTMP server setup, iOS capture, user interaction features, and practical code examples to help front‑end engineers build robust live video solutions.
This article explains how modern live‑streaming works—from client video capture and cloud‑based SaaS platforms to the RTMP and HLS protocols that power real‑time delivery, and how developers can leverage open SDKs from providers like Baidu and Huawei.
