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https://github.com/liwf616/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
List: awesome-live-stream
dash ffmpeg fmp4 hls hls-fmp4 mp4box nginx quic rtmp webrtc
Last synced: about 1 month ago
JSON representation
Webrtc && Nginx && DASH && Quic 学习资料收集，持续更新中
- Host: GitHub
- URL: https://github.com/liwf616/awesome-live-stream
- Owner: liwf616
- Created: 2018-06-06T08:39:01.000Z (over 6 years ago)
- Default Branch: master
- Last Pushed: 2022-06-16T13:57:50.000Z (over 2 years ago)
- Last Synced: 2024-05-19T22:00:56.952Z (7 months ago)
- Topics: dash, ffmpeg, fmp4, hls, hls-fmp4, mp4box, nginx, quic, rtmp, webrtc
- Homepage:
- Size: 21.9 MB
- Stars: 437
- Watchers: 44
- Forks: 138
- Open Issues: 1
-
Metadata Files:
- Readme: README.md
Awesome Lists containing this project
- ultimate-awesome - awesome-live-stream - Webrtc && Nginx && DASH && Quic 学习资料收集，持续更新中. (Other Lists / Monkey C Lists)
README
## DASH学习资料快速链接
* [ISO_IEC_23009-1_2014](https://github.com/liwf616/awesome-live-stream/blob/master/Ebook/ISO_IEC_23009-1_2014.pdf)
* [fmp4实现开源方式](https://github.com/axiomatic-systems/Bento4)
* [fmp4 nginx实现-nginx-vod-module](https://github.com/kaltura/nginx-vod-module)
* [dash相关介绍](https://bitmovin.com/dynamic-adaptive-streaming-http-mpeg-dash)
* [hls vs dash](https://www.vidbeo.com/blog/hls-vs-dash)
* [fmp4开源-shaka-packager](https://github.com/google/shaka-packager/commit/4891d9a6bf96f3655a7df4b908f96cc036a8c51b)
* [nginx rtmp -> dash](https://github.com/arut/nginx-rtmp-module)
* [nginx ts->dash](https://github.com/arut/nginx-ts-module)
* [mp4协议介绍。学好 MP4，让直播更给力](https://www.villainhr.com/page/2017/08/21/%E5%AD%A6%E5%A5%BD%20MP4%EF%BC%8C%E8%AE%A9%E7%9B%B4%E6%92%AD%E6%9B%B4%E7%BB%99%E5%8A%9B)
* [媒体文件格式分析之FMP4](https://github.com/liwf616/awesome-dash/wiki)
* [Device and Cross Browser Support For DASH](https://bitmovin.com/docs/player/articles/device-and-cross-browser-support)
* [mpeg-dash-vp9-vod-live](https://bitmovin.com/mpeg-dash-vp9-vod-live/)
## Webrtc学习资料快速链接
* [webrtc官网](https://webrtc.org/)
* [webrtc spec](https://www.w3.org/TR/webrtc/)
* [JS端的API文件](http://w3c.github.io/webrtc-pc/)
* [Native端的API文件](https://webrtc.org/native-code/native-apis/)
* [webrtchacks](https://webrtchacks.com/)
* [完整WebRTC技术及应用概要](https://mp.weixin.qq.com/s/EC8Yd74HEoIO2QxJe8-iNQ)
* [WebRTC权威指南.pdf(第三版,建议大家购买正版书籍)](https://github.com/mobinsheng/books/blob/master/1.%20WebRTC%E6%9D%83%E5%A8%81%E6%8C%87%E5%8D%97%EF%BC%88%E7%AC%AC%E4%B8%89%E7%89%88%E4%B8%AD%E6%96%87%E7%89%88%EF%BC%89.pdf)
* [WebRTC语音引擎中NetEQ技术的研究_吴江锐.pdf](https://github.com/mobinsheng/books/blob/master/WebRTC%E8%AF%AD%E9%9F%B3%E5%BC%95%E6%93%8E%E4%B8%ADNetEQ%E6%8A%80%E6%9C%AF%E7%9A%84%E7%A0%94%E7%A9%B6_%E5%90%B4%E6%B1%9F%E9%94%90.pdf)
* [Comparative Study of WebRTC Open Source SFUs for Video Conferencing(开源webrtc的sfu效果对比)](https://www.cosmosoftware.io/publications/andre2018_Comparative_Study_of_SFUs.pdf)
* [andre2018_slides_Comparative_Study_of_SFUs](https://www.cosmosoftware.io/publications/andre2018_slides_Comparative_Study_of_SFUs.pdf)
* [Improving Scale and Media Quality with Cascading SFUs](https://webrtchacks.com/sfu-cascading/)
* [Optimizing video quality using Simulcast (Oscar Divorra)](https://webrtchacks.com/sfu-simulcast/)
* [Considerations for deploying a geographically distributed video conferencing system](https://jitsi.org/wp-content/uploads/2018/11/ccwc2018-geo.pdf)
* [支持webrtc人脸实时检测](https://github.com/justadudewhohacks/face-api.js)
* [谁是最好的WebRTC SFU？](https://mp.weixin.qq.com/s/H_kBcWrzvqFlvSJyPXCeQw)
* [WebRTC Media Server--medooze](https://github.com/medooze)
* [medooze API For node.js](https://medooze.github.io/media-server-node/)
* [WebRTC Media Server--pions](https://github.com/pions/webrtc)
* [WebRTC Media Server--janus](https://github.com/meetecho/janus-gateway)
* [WebRTC Media Server--open-webrtc-toolkit](https://github.com/open-webrtc-toolkit)
* [SIP系列讲座-NAT解决方法探讨-STUN-TURN-ICE](https://mp.weixin.qq.com/s?__biz=MzA4NjU0NTIwNQ==&mid=2656444027&idx=1&sn=3a5236c3bdff4e411db0f3a3a0d8cded&chksm=8465b821b3123137e21d15d510757b9a344294ab9e53a17975bcca0e351f917daa4793723a4c&scene=21#wechat_redirect)
* [跨国实时网络调度系统设计(即构科技)](https://www.zego.im/article/2018/10/29/%E5%86%BC%E7%89%9B%EF%BC%9A%E5%8D%B3%E6%9E%84%E5%AE%9E%E6%97%B6%E7%BD%91%E7%BB%9C%E8%B0%83%E5%BA%A6%E7%B3%BB%E7%BB%9F%E5%A6%82%E4%BD%95%E5%BA%94%E5%AF%B9%E8%B7%A8%E5%9B%BD%E5%9C%BA%E6%99%AF%E6%8C%91/)
* [在Google Chrome WebRTC中分层蛋糕式的VP9 SVC](https://www.zego.im/article/2018/02/26/%E5%9C%A8google-chrome-webrtc%E4%B8%AD%E5%88%86%E5%B1%82%E8%9B%8B%E7%B3%95%E5%BC%8F%E7%9A%84vp9-svc/)
* [webrtc-build-scripts(ios && android build script)](https://github.com/pristineio/webrtc-build-scripts)
* [Webrtc Data channel --- QUIC](https://w3c.github.io/webrtc-quic/)
* [聊聊WebRTC网关服务器1：如何选择服务端端口方案?](http://yunxin.163.com/blog/webrtc-1/)
* [聊聊WebRTC网关服务器2：如何选择PeerConnection方案?](http://yunxin.163.com/blog/webrtc-2/)
* [聊聊WebRTC网关服务器3：如何优化Server的线程方案？](https://zhuanlan.zhihu.com/p/37538078)
* [聊聊WebRTC网关服务器4：QoS方案分析](https://zhuanlan.zhihu.com/p/37589412)
* [WebRTC拥塞控制策略](https://www.freehacker.cn/media/webrtc-gcc/)
* [Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.](https://webrtc-review.googlesource.com/c/src/+/64100)
* [Simulcast and Janus: what’s new? (and where’s my SSRC?)](https://www.meetecho.com/blog/simulcast-janus-ssrc/)
* [webrtc-load-testing](https://www.cosmosoftware.io/products/webrtc-load-testing)
* [Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences](https://jitsi.org/wp-content/uploads/2016/12/nossdav2015lastn.pdf)
* [如何构建分布式SFU/MCU媒体服务器？](https://mp.weixin.qq.com/s/VelFZ4QYtu6XKhBpx685lw)
* [姜健：VP9 可適性視訊編碼 (SVC) 新特性](https://www.jishuwen.com/d/2VKl/zh-tw)
* [WebRTC演示分屏实现思路](https://ouchunrun.github.io/2018/10/25/WebRTC%E6%BC%94%E7%A4%BA%E5%88%86%E5%B1%8F%E5%AE%9E%E7%8E%B0%E6%80%9D%E8%B7%AF/)
* [how-many-users-webrtc-call](https://bloggeek.me/how-many-users-webrtc-call/)
* [移动互联网的音视频传输挑战（声网）](https://myslide.cn/slides/1407?vertical=1)
* [Dominant speaker identification for multipoint videoconferencing](https://webee.technion.ac.il/Sites/People/IsraelCohen/Publications/CSL_June2013.pdf)
* [Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences](https://jitsi.org/wp-content/uploads/2016/12/nossdav2015lastn.pdf)
* [webrtcH4cKS: ~ How Zoom’s web client avoids using WebRTC (DataChannel Update)](https://webrtchacks.com/zoom-avoids-using-webrtc/#datachannels)
* [FreeSWITCH视频会议“标准”解决方案](https://mp.weixin.qq.com/s/LOCxUNBA1j94qJPqY1RKrA)
* [Dominant Speaker Identification for Multipoint Videoconferencing](https://israelcohen.com/wp-content/uploads/2018/05/IEEEI2012_Volfin.pdf)
* [Last N: Relevance-Based Selectivity for Forwarding Video in Multimedia Conferences](https://jitsi.org/wp-content/uploads/2016/12/nossdav2015lastn.pdf)
* [腾讯云快直播——超低延迟直播技术方案及应用](https://mp.weixin.qq.com/s/4gy5RDYrzcs1KjukqdWP_A)
* [阿里云 GRTN QoS 体系 — 构建实时音视频产品最佳体验](https://mp.weixin.qq.com/s/ElxkvOAZpp_sDCsNaJ9FmQ)
* [How Discord Handles Two and Half Million Concurrent Voice Users using WebRTC](https://blog.discord.com/how-discord-handles-two-and-half-million-concurrent-voice-users-using-webrtc-ce01c3187429)
* [Spatial audio](https://webrtccourse.com/course/webrtc-codelab/module/fiddle-of-the-month/lesson/spatial-audio/)
* [Scaling WebRTC for Large Rooms](https://www.slideshare.net/ggarber/scaling-webrtc-for-large-rooms)
* [Meet vs. Duo – 2 faces of Google’s WebRTC](https://webrtchacks.com/meet-vs-duo-2-faces-of-googles-webrtc/)
## MediaServer
* [WebRTC 流媒体服务器（三）- Mediasoup](https://miaopei.github.io/2019/10/21/WebRTC/mediaserver-02/)
## SDP
* [SDP Info](https://github.com/liwf616/awesome-live-stream/wiki/sdpinfo)
* [**Anatomy of a WebRTC SDP**](https://webrtchacks.com/sdp-anatomy/)
* [Unified Plan 过度指南](https://ouchunrun.github.io/2018/10/23/%E2%80%9CUnified%20Plan%E2%80%9D%20%E8%BF%87%E6%B8%A1%E6%8C%87%E5%8D%97/)
* [RFC3016 LATM](http://elkpi.com/topics/2017/06/rfc3016-latm.html)
* [WebRTC SDP 详解和剖析](https://mp.weixin.qq.com/s/L4BABJQKOCJYp_63gyEe4A)
## ICE
* [webrtc P2P 连接过程](https://blog.piasy.com/2017/08/30/WebRTC-P2P-part1/index.html)
* [webrtc quic transport](https://webrtc.org.cn/20190411-protocol-webrtc-nattraversal/)
## FEC
* [FEC之我见二](https://blog.csdn.net/zjqlovell/article/details/50978756)
* [webrtc学习之fec模块(ULPFEC Fec && Flex Fec)](https://xjsxjtu.github.io/2017-07-16/LearningWebRTC-fec/)
## 编解码
* [AV1的实时模式](https://mp.weixin.qq.com/s/JVkADQaFmTOD7_g90XFoMg)
## Qos优化 - JitterBuffer
* [ENHANCING THE QOS OF A VOIP CALL USING AN ADAPTIVE JITTER BUFFER PLAYOUT ALGORITHM WITH VARIABLE WINDOW SIZE](https://pdfs.semanticscholar.org/4665/8c1712933ca5768b6fe761b16e9ad2d4c4b9.pdf?_ga=2.239747271.550109747.1567499718-210743274.1564378121)
* [WebRTC Native 源码导读（十五）：RTP H.264 封装与解封装](https://blog.piasy.com/2019/01/01/WebRTC-RTP-Mux-Demux/index.html)
* [张轲：腾讯云H5语音通信QoE优化](https://cloud.tencent.com/developer/article/1109069)
## Qos优化 - 拥塞控制和BWE算法
* [小议WebRTC拥塞控制算法：GCC介绍](http://yunxin.163.com/blog/video18-0905/)
* [EricssonResearch/scream](https://github.com/EricssonResearch/scream)
* [Bandwidth Estimation in WebRTC (and the new Sender Side BWE) ](http://www.rtcbits.com/2017/01/bandwidth-estimation-in-webrtc-and-new.html)
* [WebRTC-GCC两种实现方案对比](https://www.freehacker.cn/media/tcc-vs-gcc/)
* [WebRTC的拥塞控制和带宽策略](https://mp.weixin.qq.com/s/Ej63-FTe5-2pkxyXoXBUTw)
* [WebRTC帧率调整策略](https://www.freehacker.cn/media/webrtc-frame/)
* [Congestion Control and Packet Scheduling for
Multipath Real Time Video Streaming](https://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=8701688)
* [NADA: A Unified Congestion Control Scheme for Real-Time Media draft-ietf-rmcat-nada-13](https://tools.ietf.org/html/draft-ietf-rmcat-nada-13)
* [Congestion Control for Real-time Communications: a comparison between NADA and GCC](https://c3lab.poliba.it/images/3/39/Med-conf-gcc.pdf)
* [一文解释清楚GOOGLE BBR拥塞控制算法原理](https://www.taohui.pub/2019/08/07/%E4%B8%80%E6%96%87%E8%A7%A3%E9%87%8A%E6%B8%85%E6%A5%9Agoogle-bbr%E6%8B%A5%E5%A1%9E%E6%8E%A7%E5%88%B6%E7%AE%97%E6%B3%95%E5%8E%9F%E7%90%86/)
* [BBR及其在实时音视频领域的应用](https://mp.weixin.qq.com/s/8Hy5SBWXzhZ2X4YnjFflJw)
* [PCC Vivace: Online-Learning Congestion Control](https://www.usenix.org/conference/nsdi18/presentation/dong)
* [PCC: Performance-oriented Congestion Control](https://modong.github.io/pcc-page/)
* [WebRTC基于TransportCC和Trendline Filter的发送端码率估计(Sendside-BWE)](https://www.jianshu.com/p/ab32a8a3552f)
* [Analysis and Design of the Google Congestion Control for Web Real-time Communication (WebRTC)](https://c3lab.poliba.it/images/6/65/Gcc-analysis.pdf)
* [Evaluating Congestion Control for Interactive Real-time Media](https://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-criteria/)
* [WebRTC中PacedSender工作原理和代码分析](https://www.jianshu.com/p/3fde9b8d77f6)
* [Webrtc Nack重传指数退避算法](https://gitlab.com/webrtc-mirror/webrtc/commit/3eae7e4e3cdf1a7459905e734f2902876bdaa9cd)
* [Evaluating COPA congestion control for improved video performance](https://engineering.fb.com/video-engineering/copa/)
## Qos优化 - 测试方法
* [WebRTC vs. Zoom 之外：WebRTC 的弱网模拟测试](https://webrtc.org.cn/network-test-for-webrtc/)
## Nginx学习资料快速链接
* [Nginx 对udp多packet的支持](http://hg.nginx.org/nginx/rev/d27aa9060c95)
* [深入理解Nginx模块开发和架构解析](https://github.com/cjl3080434008/2014/blob/master/read_book/nginx/%E6%B7%B1%E5%85%A5%E7%90%86%E8%A7%A3Nginx%E6%A8%A1%E5%9D%97%E5%BC%80%E5%8F%91%E5%8F%8A%E6%9E%B6%E6%9E%84%E8%A7%A3%E6%9E%90.pdf)
* [Nginx开发从入门到精通](http://tengine.taobao.org/book/)
* [nginx-rtmp-module](https://github.com/arut/nginx-rtmp-module)
* [BLSS(NGINX-based Live Media Streaming Server)](https://github.com/gnolizuh/BLSS)
* [Nginx限速模块初探](https://www.cnblogs.com/CarpenterLee/p/8084533.html)
* [动态追踪技术漫谈](https://openresty.org/posts/dynamic-tracing/)
* [lua-nginx-module](https://github.com/openresty/lua-nginx-module)
* [openresty-systemtap-toolkit ](https://github.com/openresty/openresty-systemtap-toolkit)
* [sample-bt (CPU Flame Graphs)](https://github.com/openresty/openresty-systemtap-toolkit#sample-bt)
* [ngx-sample-lua-bt (CPU Flame Graphs)](https://github.com/openresty/openresty-systemtap-toolkit#ngx-sample-lua-bt)
* [awesome-nginx](https://github.com/agile6v/awesome-nginx)
* [annotated_nginx](https://github.com/chronolaw/annotated_nginx)
## Quic && KCP && KTP && PCC && SRT 学习资料快速链接
* [kcp-go](https://github.com/xtaci/kcp-go)
* [Nginx支持quic的最新消息](https://www.nginx.com/blog/introducing-technology-preview-nginx-support-for-quic-http-3/?from=timeline)
* [Golang版本quic<==>quic-go](https://github.com/lucas-clemente/quic-go)
* [QUIC 开源项目汇总](https://github.com/quicwg/base-drafts/wiki/Implementations)
* [快手多媒体传输算法优化实践](https://mp.weixin.qq.com/s/iyX6bEBTQxd2V9OXNnvUUA)
* [B站QUIC实践之路](https://mp.weixin.qq.com/s/DrGm-OkSpJbzPWbFmSBT8g)
* [RTP over QUIC draft-rtpfolks-quic-rtp-over-quic-01](https://tools.ietf.org/html/draft-rtpfolks-quic-rtp-over-quic-01)
* [Savoury implementation of the QUIC transport protocol and HTTP/3](https://github.com/cloudflare/quiche)
* [nginx-quic开源实现](https://github.com/evansun922/nginx-quic)
* [mvfst - An implementation of the QUIC transport protocol.](https://github.com/facebookincubator/mvfst)
* [Experiment with HTTP/3 using NGINX and quiche](https://blog.cloudflare.com/experiment-with-http-3-using-nginx-and-quiche/)
* [msquic](https://github.com/microsoft/msquic)
* [阿里XQUIC：标准QUIC实现自研之路](https://mp.weixin.qq.com/s/pBv_DnG05YWl4ZYRHThaTw)
* [QUIC协议在BIGO的实践与优化](https://mp.weixin.qq.com/s/885WDIFohS7rh3avwFRYkA)
## SRT学习资料链接
* [ffmpeg enable-libsrt问题解决](https://blog.csdn.net/liwf616/article/details/99215608)
* [srt-live-server](https://github.com/Edward-Wu/srt-live-server)
* [拆解SRT：新UDP视频传输协议](https://mp.weixin.qq.com/s/1VcmQgYHEH4oih9BnRv_Og)
* [SRT-GO](https://github.com/Haivision/srtgo)
## HLS学习资料快速链接
* [HTTP Live Streaming(rfc8216)](https://tools.ietf.org/html/rfc8216)
* [hls之m3u8、ts流格式详解](https://my.oschina.net/u/727148/blog/666824)
* [HLS fmp4 h264点播播放地址](https://bitdash-a.akamaihd.net/content/MI201109210084_1/m3u8s-fmp4/f08e80da-bf1d-4e3d-8899-f0f6155f6efa.m3u8)
* [HLS fmp4 h265点播播放地址](http://bitmovin-a.akamaihd.net/content/dataset/multi-codec/hevc/stream_fmp4.m3u8)
* [第一章：TS预备知识](https://www.onelib.biz/doc/stb/course/prereading.html)
* [第二章：从TS到PAT和PMT](https://www.onelib.biz/doc/stb/course/begin.html)
* [第三章：深入学习PSI](https://www.onelib.biz/doc/stb/course/psi.html)
* [第四章：深入学习SI](https://www.onelib.biz/doc/stb/course/si.html)
* [HLS vs DASH vs HDS vs MSS](https://bitmovin.com/mpeg-dash-vs-apple-hls-vs-microsoft-smooth-streaming-vs-adobe-hds/)
## HLS协议规范
* [HLS 最新的协议（cbcs）](https://tools.ietf.org/html/rfc8216)
* [mp4 规范(亚马逊的fmp4实现参照这个方案)](http://l.web.umkc.edu/lizhu/teaching/2016sp.video-communication/ref/mp4.pdf)
* [HLS对4K技术的支持](http://www.streamingmedia.com/Articles/Editorial/Featured-Articles/Apple-Got-It-Wrong-Encoding-Specs-for-HEVC-in-HLS--121878.aspx)
* [HLS对HDR技术的支持](https://streaminglearningcenter.com/blogs/apple-updates-hls-authoring-spec-4k-hdr.html)
## 通用工具
* [DASH 播放器](http://reference.dashif.org/dash.js/nightly/samples/dash-if-reference-player/index.html)
* [HLS fmp4播放器](https://bitmovin.com/hls-news-wwdc-2016)
* [mp4box 工具](https://gpac.wp.imt.fr/mp4box/dash/)
* [hls.js 播放器](http://video-dev.github.io/hls.js/demo/)
* [mp4box 工具](https://gpac.wp.imt.fr/mp4box/dash/)
* [qt实现的mp4分析工具](https://github.com/ksvc/MediaParser)
* [弱网模拟的工具-network-emulator-toolkit](https://blog.mrpol.nl/2010/01/14/network-emulator-toolkit/)
* [弱网模拟的工具-clumsy](https://jagt.github.io/clumsy/)
* [webrtc munge-sdp](https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp/)
* [obs.ninja](https://obs.ninja/)
* [obsninja](https://github.com/steveseguin/obsninja)
## 音视频峰会
* [WWDC16: HLS Supports Fragmented MP4](https://bitmovin.com/hls-news-wwdc-2016/)
* [WWDC17 – HEVC with HLS](https://bitmovin.com/wwdc17-hevc-hls-apple-just-announced-feature-support-box/)
* [2017腾讯LIVE开发者大会](https://github.com/iv-web/ppts/tree/master/2017_TLC_ppts)
* [2018腾讯LIVE开发者大会](https://github.com/iv-web/ppts/tree/master/2018_TLC_ppts)
* [2017杭州云栖大会100位大咖视频+讲义全分享](https://yq.aliyun.com/articles/231065)
* [FOSDEM 2019 - Real Time Communications (RTC) devroom](https://fosdem.org/2019/schedule/track/real_time_communications_rtc/)
* [2019杭州云栖大会回顾](https://yunqi.youku.com/2019/hangzhou/review?spm=a2c4e.11165380.1395223.1)
* [URTC万人直播互动实践之路](https://mp.weixin.qq.com/s/l9rmV0fNm2UKRcFB-1tPxw)
## 行业报告
* [Bitmovin: 视频开发者报告 2018](https://mp.weixin.qq.com/s/2o-lt3RKwybE3iXXLc7eqQ)
* [2019年低延迟直播技术展望](https://mp.weixin.qq.com/s/Q0TbJCkr_wNPkX1VpsXr5g)
* [On the Road to WebRTC 1.0, Including VP8](https://webkit.org/blog/8672/on-the-road-to-webrtc-1-0-including-vp8/)
* [低延时HLS直播（苹果公司）](https://developer.apple.com/videos/play/wwdc2019/502/)
* [【杭州云栖】AliQUIC：场景化高性能传输网络实践](http://blog.itpub.net/31550522/viewspace-2215505/)
* [WebRTC project updates 2019年11月15日](https://www.youtube.com/watch?v=avtlQeaxd_I&feature=youtu.be)
## Webrtc rfc
* [rfc5245（ICE)](http://www.faqs.org/rfcs/rfc5245.html)
* [RTP Payload for Redundant Audio Data](https://datatracker.ietf.org/doc/html/rfc2198)
* [rfc3550（RTP: A Transport Protocol for Real-Time Applications）](http://www.ietf.org/rfc/rfc3550.txt)
* [Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)(NACK/PLI/SLI/RPSI/TSTR/TSTN/VBCM)rfc4585](https://tools.ietf.org/html/rfc4585)
* [Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)(TMMBR/TMMBN)](https://tools.ietf.org/html/rfc5104)
* [RTP Extensions for Transport-wide Congestion Control draft-holmer-rmcat-transport-wide-cc-extensions-01(TCC format)](https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01)
* [RTP Payload Format for H.264 Video](https://tools.ietf.org/html/rfc6184)
* [rfc4566（SDP: Session Description Protocol）](https://tools.ietf.org/html/rfc4566)
* [Annotated Example SDP for WebRTC draft-ietf-rtcweb-sdp-09](https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-09)
* [rfc3711 (The Secure Real-time Transport Protocol (SRTP))](https://www.ietf.org/rfc/rfc3711.txt)
* [rfc5285 (A General Mechanism for RTP Header Extensions）](https://tools.ietf.org/html/rfc5285)
* [Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)](https://tools.ietf.org/html/rfc5763)
* [WebRTC MediaStream Identification in the Session Description Protocol draft-ietf-mmusic-msid-16](https://tools.ietf.org/html/draft-ietf-mmusic-msid-16)
* [Using Simulcast in SDP and RTP Sessions(draft-ietf-mmusic-sdp-simulcast-11)](https://tools.ietf.org/id/draft-ietf-mmusic-sdp-simulcast-11.html)
* [Selective Forwarding Middlebox](https://tools.ietf.org/html/rfc7667#section-3.7)
* [Scalable Video Coding (SVC) Extension for WebRTC](https://w3c.github.io/webrtc-svc/)
* [RTP Payload Format for Flexible Forward Error Correction (FEC)](https://tools.ietf.org/id/draft-ietf-payload-flexible-fec-scheme-06.html)
* [ICE Renomination: Dynamically selecting ICE candidate pairs draft-thatcher-ice-renomination-01](https://tools.ietf.org/html/draft-thatcher-ice-renomination-01)
* [A Real-Time Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication
draft-lennox-avt-rtp-audio-level-exthdr-02](https://tools.ietf.org/pdf/draft-lennox-avt-rtp-audio-level-exthdr-02.pdf)
* [Frame Marking RTP Header Extension draft-ietf-avtext-framemarking-10](https://tools.ietf.org/html/draft-ietf-avtext-framemarking-10)
* [TCP Candidates with Interactive Connectivity Establishment (ICE)](https://tools.ietf.org/html/rfc6544)
* [Datagram Transport Layer Security Version 1.2](https://tools.ietf.org/html/rfc6347)
* [RTP Payload Format for MPEG-4 Audio/Visual Streams](https://tools.ietf.org/html/rfc3016)
* [RTP Control Protocol Extended Reports (RTCP XR)](https://tools.ietf.org/html/rfc3611)
* [Reed-Solomon Forward Error Correction (FEC) Schemes](https://tools.ietf.org/html/rfc5510)
* [RTP Stream Identifier Source Description (SDES) draft-ietf-avtext-rid-09](https://tools.ietf.org/html/draft-ietf-avtext-rid-09)
* [RTP Topologies](https://datatracker.ietf.org/doc/html/rfc7667)
* [Sending Multiple RTP Streams in a Single RTP Session](https://datatracker.ietf.org/doc/rfc8108/)
## DASH常用的命令
1. 用mp4box将ss.mp4切割成fragment mp4。
```shell
mp4box -dash 5000 -frag 5000 -rap -frag-rap -profile dashavc264:live ss.mp4 -out ss_dash.mpd
```
2. 用mp4将ss.mp4切割成fragment mp4。
```shell
1. mp4fragment ss.mp4 ss_fragment.mp4
2. mp4dash --use-segment-timeline ss_fragment.mp4
```
3. mp4dump工具
## Webrtc工具集
* [internals-parameters](https://testrtc.com/webrtc-internals-parameters/)