This article provides a comprehensive technical overview of SIP signaling, SDP negotiation, RTP/RTCP media transport, and the Binary Floor Control Protocol (BFCP), including their interactions, message flows, deployment models, and practical screen‑sharing implementations for video‑conference systems.
This guide shows how to implement a complete WebRTC solution using only PHP with FFI, covering dependency installation, Composer setup, and detailed RTCPeerConnection configuration including STUN/TURN servers and TLS certificates.
The article explains WebRTC audio‑video synchronization by detailing the ITU‑R BT.1359‑1 standard, describing a synchronizer that computes target delays from desired and packet‑pair delays, applying bounded adjustments via sliding‑average filtering, and outlining a hardware method for measuring resulting latency.
This article explains the structure and generation of RTP timestamps for audio and video, the role of NTP timestamps as a common time base, how RTP and NTP are correlated through Sender Reports and linear regression, and the calculations used to achieve accurate audio‑video synchronization and target delay management in WebRTC.
This article details the design of an AI interview robot for 58.com, covering its backend architecture, dialogue engine, resource‑management strategies, performance‑testing methodology, and the optimizations that boosted concurrent interview capacity by twenty times while improving user experience.
This article details the backend architecture, dialogue engine design, resource estimation, and performance optimization techniques of an AI interview robot used in 58.com’s recruitment platform, illustrating how multi‑branch dialogue flows, RTP communication, session management, and monitoring enable scalable, stable, and efficient online interview services.
This article explains how RTC and WebRTC use P2P communication, the challenges posed by NAT, and how ICE, STUN, TURN, SDP and RTP work together to enable reliable real‑time audio‑video transmission over the internet.
This article provides a comprehensive walkthrough of WebRTC's video sender NACK implementation, covering the underlying ACK/NACK concepts, RFC4585 retransmission types, and the three-step process of storing RTP packets, handling RTCP NACK messages, and retransmitting lost media with priority handling.
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 why real‑time audio/video communication prefers UDP over TCP, compares TCP and UDP characteristics, introduces the RTP and RTCP application‑layer protocols that enable reliable media transport, and provides a quick Wireshark guide for capturing and analyzing RTP/RTCP packets.
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.
This article explains how an intelligent voice robot leverages TTS and SIP to convert server alerts into spoken notifications, detailing the end‑to‑end workflow, DTMF transmission methods, SIP detection techniques, SDP media negotiation, and RTP‑based DTMF parsing to enable reliable key‑press handling.
The article reviews the design and implementation of 58.com’s intelligent voice robot backend, detailing its four‑layer architecture, SIP/SDP/RTP protocols, multi‑round voice interaction flow, algorithm service modularization, SIP scheduling optimizations, and Java thread‑pool system tuning for high‑concurrency scenarios.
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.
Shang Shidong outlines Tencent Meeting’s shift from analog PSTN to IP‑based VoIP, using H.323, SIP, RTP/UDP and the Opus codec, while AI‑driven super‑resolution, deep‑learning packet‑loss concealment, advanced noise reduction, and speech‑music classification boost audio quality, complemented by reference‑free MOS assessment and future 5G‑enabled cloud, IoT and WebRTC integration.
This article introduces the GB28181 national video surveillance standard, explains its SIP‑based signaling architecture, details the SIP methods and message flows, describes the system components and hierarchical deployment, and outlines the registration and real‑time streaming processes required for implementation.
This article explains how WebRTC-based real-time audio/video uses P2P and server‑relay modes, details the discovery and tracing of rotation metadata issues, describes SDP and RTP header extensions for video orientation, and shows how to map this information to ffmpeg rotation parameters.
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.
The article explains why TCP‑based protocols cannot meet low‑latency requirements for live‑streaming conferences and introduces RTP, RTCP, jitter, round‑trip time, and three packet‑loss mitigation strategies—retransmission, forward error correction, and cross‑transport—along with a brief overview of DCCP for congestion control.
