Ecosyste.ms: Awesome
An open API service indexing awesome lists of open source software.
awesome-live-stream
Webrtc && Nginx && DASH && Quic 学习资料收集，持续更新中
https://github.com/liwf616/awesome-live-stream
- fmp4实现开源方式
- fmp4 nginx实现-nginx-vod-module
- dash相关介绍
- hls vs dash
- fmp4开源-shaka-packager
- nginx rtmp -> dash
- nginx ts->dash
- mp4协议介绍。学好 MP4，让直播更给力
- 媒体文件格式分析之FMP4
- Device and Cross Browser Support For DASH
- mpeg-dash-vp9-vod-live
- webrtc官网
- webrtc spec
- JS端的API文件
- Native端的API文件
- webrtchacks
- 完整WebRTC技术及应用概要
- WebRTC权威指南.pdf(第三版,建议大家购买正版书籍)
- WebRTC语音引擎中NetEQ技术的研究_吴江锐.pdf
- Comparative Study of WebRTC Open Source SFUs for Video Conferencing(开源webrtc的sfu效果对比)
- andre2018_slides_Comparative_Study_of_SFUs
- Improving Scale and Media Quality with Cascading SFUs
- Optimizing video quality using Simulcast (Oscar Divorra)
- Considerations for deploying a geographically distributed video conferencing system
- 支持webrtc人脸实时检测
- 谁是最好的WebRTC SFU？
- WebRTC Media Server--medooze
- medooze API For node.js
- WebRTC Media Server--pions
- WebRTC Media Server--janus
- WebRTC Media Server--open-webrtc-toolkit
- SIP系列讲座-NAT解决方法探讨-STUN-TURN-ICE
- 跨国实时网络调度系统设计(即构科技)
- 在Google Chrome WebRTC中分层蛋糕式的VP9 SVC
- webrtc-build-scripts(ios && android build script)
- Webrtc Data channel --- QUIC
- 聊聊WebRTC网关服务器1：如何选择服务端端口方案?
- 聊聊WebRTC网关服务器2：如何选择PeerConnection方案?
- 聊聊WebRTC网关服务器3：如何优化Server的线程方案？
- 聊聊WebRTC网关服务器4：QoS方案分析
- WebRTC拥塞控制策略
- Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
- Simulcast and Janus: what’s new? (and where’s my SSRC?)
- webrtc-load-testing
- Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences
- 如何构建分布式SFU/MCU媒体服务器？
- 姜健：VP9 可適性視訊編碼 (SVC) 新特性
- WebRTC演示分屏实现思路
- how-many-users-webrtc-call
- 移动互联网的音视频传输挑战（声网）
- Dominant speaker identification for multipoint videoconferencing
- Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences
- webrtcH4cKS: ~ How Zoom’s web client avoids using WebRTC (DataChannel Update)
- FreeSWITCH视频会议“标准”解决方案
- Dominant Speaker Identification for Multipoint Videoconferencing
- Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences
- 腾讯云快直播——超低延迟直播技术方案及应用
- 阿里云 GRTN QoS 体系 — 构建实时音视频产品最佳体验
- How Discord Handles Two and Half Million Concurrent Voice Users using WebRTC
- Spatial audio
- Scaling WebRTC for Large Rooms
- Meet vs. Duo – 2 faces of Google’s WebRTC
- WebRTC 流媒体服务器（三）- Mediasoup
- **Anatomy of a WebRTC SDP**
- Unified Plan 过度指南
- RFC3016 LATM
- WebRTC SDP 详解和剖析
- webrtc P2P 连接过程
- webrtc quic transport
- FEC之我见二
- webrtc学习之fec模块(ULPFEC Fec && Flex Fec)
- AV1的实时模式
- ENHANCING THE QOS OF A VOIP CALL USING AN ADAPTIVE JITTER BUFFER PLAYOUT ALGORITHM WITH VARIABLE WINDOW SIZE
- WebRTC Native 源码导读（十五）：RTP H.264 封装与解封装
- 张轲：腾讯云H5语音通信QoE优化
- 小议WebRTC拥塞控制算法：GCC介绍
- EricssonResearch/scream
- Bandwidth Estimation in WebRTC (and the new Sender Side BWE)
- WebRTC-GCC两种实现方案对比
- WebRTC的拥塞控制和带宽策略
- WebRTC帧率调整策略
- NADA: A Unified Congestion Control Scheme for Real-Time Media draft-ietf-rmcat-nada-13
- Congestion Control for Real-time Communications: a comparison between NADA and GCC
- 一文解释清楚GOOGLE BBR拥塞控制算法原理
- BBR及其在实时音视频领域的应用
- PCC Vivace: Online-Learning Congestion Control
- PCC: Performance-oriented Congestion Control
- WebRTC基于TransportCC和Trendline Filter的发送端码率估计(Sendside-BWE)
- Analysis and Design of the Google Congestion Control for Web Real-time Communication (WebRTC)
- Evaluating Congestion Control for Interactive Real-time Media
- WebRTC中PacedSender工作原理和代码分析
- Webrtc Nack重传指数退避算法
- Evaluating COPA congestion control for improved video performance
- WebRTC vs. Zoom 之外：WebRTC 的弱网模拟测试
- Nginx 对udp多packet的支持
- 深入理解Nginx模块开发和架构解析
- Nginx开发从入门到精通
- nginx-rtmp-module
- BLSS(NGINX-based Live Media Streaming Server)
- Nginx限速模块初探
- 动态追踪技术漫谈
- lua-nginx-module
- openresty-systemtap-toolkit
- sample-bt (CPU Flame Graphs)
- ngx-sample-lua-bt (CPU Flame Graphs)
- awesome-nginx
- annotated_nginx
- kcp-go
- Nginx支持quic的最新消息
- Golang版本quic<==>quic-go
- QUIC 开源项目汇总
- 快手多媒体传输算法优化实践
- B站QUIC实践之路
- RTP over QUIC draft-rtpfolks-quic-rtp-over-quic-01
- Savoury implementation of the QUIC transport protocol and HTTP/3
- nginx-quic开源实现
- mvfst - An implementation of the QUIC transport protocol.
- Experiment with HTTP/3 using NGINX and quiche
- msquic
- 阿里XQUIC：标准QUIC实现自研之路
- QUIC协议在BIGO的实践与优化
- ffmpeg enable-libsrt问题解决
- srt-live-server
- 拆解SRT：新UDP视频传输协议
- SRT-GO
- HTTP Live Streaming(rfc8216)
- hls之m3u8、ts流格式详解
- HLS fmp4 h264点播播放地址
- HLS fmp4 h265点播播放地址
- 第一章：TS预备知识
- 第二章：从TS到PAT和PMT
- 第三章：深入学习PSI
- 第四章：深入学习SI
- HLS vs DASH vs HDS vs MSS
- HLS 最新的协议（cbcs）
- mp4 规范(亚马逊的fmp4实现参照这个方案)
- HLS对4K技术的支持
- HLS对HDR技术的支持
- DASH 播放器
- HLS fmp4播放器
- mp4box 工具
- hls.js 播放器
- mp4box 工具
- qt实现的mp4分析工具
- 弱网模拟的工具-network-emulator-toolkit
- 弱网模拟的工具-clumsy
- webrtc munge-sdp
- obs.ninja
- obsninja
- WWDC16: HLS Supports Fragmented MP4
- WWDC17 – HEVC with HLS
- 2017腾讯LIVE开发者大会
- 2018腾讯LIVE开发者大会
- 2017杭州云栖大会100位大咖视频+讲义全分享
- FOSDEM 2019 - Real Time Communications (RTC) devroom
- 2019杭州云栖大会回顾
- URTC万人直播互动实践之路
- Bitmovin: 视频开发者报告 2018
- 2019年低延迟直播技术展望
- On the Road to WebRTC 1.0, Including VP8
- 低延时HLS直播（苹果公司）
- 【杭州云栖】AliQUIC：场景化高性能传输网络实践
- WebRTC project updates 2019年11月15日
- rfc5245（ICE)
- RTP Payload for Redundant Audio Data
- rfc3550（RTP: A Transport Protocol for Real-Time Applications）
- Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)(NACK/PLI/SLI/RPSI/TSTR/TSTN/VBCM)rfc4585
- Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)(TMMBR/TMMBN)
- RTP Extensions for Transport-wide Congestion Control draft-holmer-rmcat-transport-wide-cc-extensions-01(TCC format)
- RTP Payload Format for H.264 Video
- rfc4566（SDP: Session Description Protocol）
- Annotated Example SDP for WebRTC draft-ietf-rtcweb-sdp-09
- rfc3711 (The Secure Real-time Transport Protocol (SRTP))
- rfc5285 (A General Mechanism for RTP Header Extensions）
- Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)
- WebRTC MediaStream Identification in the Session Description Protocol draft-ietf-mmusic-msid-16
- Using Simulcast in SDP and RTP Sessions(draft-ietf-mmusic-sdp-simulcast-11)
- Selective Forwarding Middlebox
- Scalable Video Coding (SVC) Extension for WebRTC
- RTP Payload Format for Flexible Forward Error Correction (FEC)
- ICE Renomination: Dynamically selecting ICE candidate pairs draft-thatcher-ice-renomination-01
- Frame Marking RTP Header Extension draft-ietf-avtext-framemarking-10
- TCP Candidates with Interactive Connectivity Establishment (ICE)
- Datagram Transport Layer Security Version 1.2
- RTP Payload Format for MPEG-4 Audio/Visual Streams
- RTP Control Protocol Extended Reports (RTCP XR)
- Reed-Solomon Forward Error Correction (FEC) Schemes
- RTP Stream Identifier Source Description (SDES) draft-ietf-avtext-rid-09
- RTP Topologies
- Sending Multiple RTP Streams in a Single RTP Session
- internals-parameters
Programming Languages
Keywords
webrtc
3
hls
3
nginx
3
mp4
2
streaming
2
network-programming
2
nodejs
2
protocol
2
livestream
2
quic
2
rust
2
video
2
dash
2
ngx-http
1
ngx-stream
1
ngx-thread-pool
1
arq
1
cryptography
1
kcp
1
reedsolomon
1
udp
1
c
1
cpp
1
ngx-event
1
ngx-connection
1
ngx-lua
1
nginx-server
1
nginx-http
1
lua-resty
1
lua-nginx
1
awesome-list
1
awesome
1
rtmp-server
1
rtmp
1
httpflv-server
1
httpflv
1
h265
1
specification
1
http3
1
mediaserver
1
tfjs
1
tensorflowjs
1
tensorflow
1
js
1
javascript
1
gender-recognition
1
face-recognition
1
face-landmarks
1
face-detection
1
emotion-recognition
1