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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.
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 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.
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.
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.
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.
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 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.
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.