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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
Last synced: 3 days ago
JSON representation
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DASH学习资料快速链接
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Webrtc学习资料快速链接
- 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
- 谁是最好的WebRTC SFU？
- WebRTC Media Server--medooze
- WebRTC Media Server--pions
- SIP系列讲座-NAT解决方法探讨-STUN-TURN-ICE
- webrtc-load-testing
- 姜健：VP9 可適性視訊編碼 (SVC) 新特性
- How Discord Handles Two and Half Million Concurrent Voice Users using WebRTC
- Webrtc Data channel --- QUIC
- JS端的API文件
- 支持webrtc人脸实时检测
- medooze API For node.js
- WebRTC Media Server--janus
- webrtc-build-scripts(ios && android build script)
- 跨国实时网络调度系统设计(即构科技)
- webrtc官网
- webrtc spec
- 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
- Improving Scale and Media Quality with Cascading SFUs
- Optimizing video quality using Simulcast (Oscar Divorra)
- Considerations for deploying a geographically distributed video conferencing system
- WebRTC Media Server--open-webrtc-toolkit
- 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
- 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
- Spatial audio
- Scaling WebRTC for Large Rooms
- Meet vs. Duo – 2 faces of Google’s WebRTC
- 姜健：VP9 可適性視訊編碼 (SVC) 新特性
- 在Google Chrome WebRTC中分层蛋糕式的VP9 SVC
- Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
- Simulcast and Janus: what’s new? (and where’s my SSRC?)
- 如何构建分布式SFU/MCU媒体服务器？
- 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
- 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
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SDP
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ICE
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FEC
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Qos优化 - JitterBuffer
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Qos优化 - 拥塞控制和BWE算法
- NADA: A Unified Congestion Control Scheme for Real-Time Media draft-ietf-rmcat-nada-13
- 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
- EricssonResearch/scream
- PCC: Performance-oriented Congestion Control
- Bandwidth Estimation in WebRTC (and the new Sender Side BWE)
- 小议WebRTC拥塞控制算法：GCC介绍
- 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
- 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
- PCC Vivace: Online-Learning Congestion Control
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Qos优化 - 测试方法
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Nginx学习资料快速链接
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Quic && KCP && KTP && PCC && SRT 学习资料快速链接
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HLS协议规范
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HLS学习资料快速链接
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通用工具
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音视频峰会
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行业报告
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Webrtc rfc
- 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
- 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
- rfc5245（ICE)
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MediaServer
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编解码
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SRT学习资料链接
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Webrtc工具集
Programming Languages
Categories
Sub Categories
Keywords
nodejs
2
video
2
streaming
2
webrtc
2
mp4
2
dash
2
hls
2
mss
1
livestream
1
hds
1
drm
1
hevc
1
codec
1
cmaf
1
avc
1
av1
1
audio
1
ac-4
1
ac-3
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
age-estimation
1
specification
1
vod
1
video-streaming
1
stream
1
nginx
1
aac
1