The simple-rtmp-server from han4235

#Simple-RTMP-Server

SRS/2.0，开发代号：ZhouGuowen

SRS定位是运营级的互联网直播服务器集群，追求更好的概念完整性和最简单实现的代码。

Download from github.io: Centos6-x86_64 more...

Download from ossrs.net: Centos6-x86_64 more...

Contact by QQ or Weixin, read Contact

Why SRS?

Completely rewrite HLS following m3u8/ts spec, and HLS support h.264+aac/mp3.
High efficient RTMP deliverying support 7k+ concurrency, vhost based, both origin and edge.
Embeded simplified media HTTP server for HLS, api and HTTP flv/ts/mp3/aac streaming.
Variety input: RTMP, pull by ingest file or stream(HTTP/RTMP/RTSP), push by stream caster RTSP/MPEGTS-over-UDP.
Popular internet delivery: RTMP/HDS for flash, HLS for mobile(IOS/IPad/MAC/Android), HTTP flv/ts/mp3/aac streaming for user prefered.
Enhanced DVR and hstrs: segment/session/append plan, customer path and HTTP callback. the hstrs(http stream trigger rtmp source) enable the http-flv stream standby util encoder start publish, similar to rtmp, which will trigger edge to fetch from origin.
Multiple feature: transcode, forward, ingest, http hooks, dvr, hls, rtsp, http streaming, http api, refer, log, bandwith test and srs-librtmp.
Best maintainess: simple arch over state-threads(coroutine), single thread, single process and for linux/osx platform, common server x86-64/i386/arm/mips cpus, rich comments, strictly follows RTMP/HLS/RTSP spec.
Easy to use: both English and Chinese wiki, typically config files in trunk/conf, traceable and session based log, linux service script and install script.
MIT license, open source with product management and evolution.

Enjoy it!

About

SRS(SIMPLE RTMP Server) over state-threads created in 2013.10.

SRS delivers rtmp/hls/http/hds live on x86/x64/arm/mips linux/osx, supports origin/edge/vhost and transcode/ingest and dvr/forward and http-api/http-callback/reload, introduces tracable session-oriented log, exports client srs-librtmp, with stream caster to push MPEGTS-over-UDP/RTSP to SRS, provides EN/CN wiki and the most simple architecture.

SRS focus on small problem domain, which is the most complex for all software(see OOAD). Because of lack of deveoper resource, SRS only provides features which is the most popular for internet. SRS is simple for and only for problem domain is simplified.

SRS is a simple, RTMP( CN, EN ), HLS( CN, EN ), HDS( CN, EN ), HTTP( CN, EN ), high-performance(10k+ clients)( CN, EN ), low-latency(0.1s+)( CN, EN ), single processes, edge/origin live server, x86/x64/arm( CN, EN ), compile depends on st( CN, EN )(required), ssl and http-parser, use nginx, ffmpeg and cherrypy as external tools. that is, only need st to run srs for minimum run. see Build( CN, EN ).

SRS supports vhost( CN, EN ), rtmp(encoder push( CN, EN ), client/edge( CN, EN, ) pull), ingester(srs pull)( CN, EN ), HLS( CN, EN ), HLS audio only( CN, EN ), transcoding( CN, EN ), forward( CN, EN ), http hooks( CN, EN ), http api( CN, EN ), http server( CN, EN ), dvr( CN, EN ) and SRS-librtmp( CN, EN ).

SRS-librtmp( CN, EN ) is a client library, only depends on c++ and socket, with examples( CN, EN )(to play, publish, ingest flv/rtmp, inject flv, publish h264 raw stream( CN, EN ), exported as seperate project or single cpp file by configure( CN, EN ). SRS-librtmp( CN, EN ) provides apis to support RTMP, FLV, AMF0 and h.264 raw stream( CN, EN ).

Report Bug: https://github.com/winlinvip/simple-rtmp-server/issues/new
WebSite: http://ossrs.net
Release: http://winlinvip.github.io/srs.release
Blog: http://blog.csdn.net/win_lin
Wiki: https://github.com/winlinvip/simple-rtmp-server/wiki
StreamServers：BLS/BLE, NGINX-RTMP, CRTMPD, RED5, WOWZA, FMS/AMS

AUTHORS

There are three types of people that have contributed to the SRS project:

PRIMARY: Contribute important features and >10% code. Names of all PRIMARY response in NetConnection.connect and metadata.
AUTHORS: Contribute features and 1%~10% code. Names of all PRIMARY response in NetConnection.connect and metadata.
CONTRIBUTORS: Submit patches, report bugs, add translations, help answer newbie questions, and generally make SRS that much better.

About all PRIMARY, AUTHORS and CONTRIBUTORS, read AUTHORS.txt.

A big THANK YOU goes to:

All friends of SRS for big supports.
Genes amd Mabbott for creating st(state-threads).
Michael Talyanksy for introducing us to use st.
Roman Arutyunyan for creating nginx-rtmp for SRS to refer to.
Joyent for creating http-parser for http-api for SRS.
Igor Sysoev for creating nginx for SRS to refer to.
FFMPEG and libx264 group for SRS to use to transcode.
Guido van Rossum for creating Python for api-server for SRS.

Mirrors

Github: https://github.com/winlinvip/simple-rtmp-server, the GIT usage( CN, EN )

git clone https://github.com/winlinvip/simple-rtmp-server.git

CSDN: https://code.csdn.net/winlinvip/srs-csdn , the GIT usage( CN, EN )

git clone https://code.csdn.net/winlinvip/srs-csdn.git

OSChina: http://git.oschina.net/winlinvip/srs.oschina , the GIT usage( CN, EN )

git clone https://git.oschina.net/winlinvip/srs.oschina.git

Gitlab: https://gitlab.com/winlinvip/srs-gitlab , the GIT usage( CN, EN )

git clone https://gitlab.com/winlinvip/srs-gitlab.git

Usage

Step 1: get SRS

git clone https://github.com/winlinvip/simple-rtmp-server &&
cd simple-rtmp-server/trunk

Step 2: build SRS, Requires Centos6.x/Ubuntu12 32/64bits, others see Build( CN, EN ).

./configure && make

Step 3: start SRS

./objs/srs -c conf/srs.conf

See also:

Usage: How to delivery RTMP?( CN, EN )
Usage: How to delivery HLS?( CN, EN )
Usage: How to delivery HLS for other codec?( CN, EN )
Usage: How to transode RTMP stream by SRS?( CN, EN )
Usage: How to forward stream to other server?( CN, EN )
Usage: How to deploy low lantency application?( CN, EN )
Usage: How to deploy SRS on ARM?( CN, EN )
Usage: How to ingest file/stream/device to SRS?( CN, EN )
Usage: How to use SRS-HTTP-server to delivery HTTP/HLS stream?( CN, EN )
Usage: How to show the demo of SRS?( CN, EN )
Usage: How to publish h.264 raw stream to SRS?( CN, EN )
Usage: Solution using SRS?( CN, EN )
Usage: Why SRS?( CN, EN )

Wiki

SRS 1.0 wiki

Please select your language:

SRS 2.0 wiki

Please select your language:

Donation

Donation:
http://winlinvip.github.io/srs.release/donation/index.html OR
http://www.ossrs.net/srs.release/donation/index.html

Donations:
[https://github.com/winlinvip/simple-rtmp-server/blob/develop/DONATIONS.txt] (https://github.com/winlinvip/simple-rtmp-server/blob/develop/DONATIONS.txt)

System Requirements

Supported operating systems and hardware:

All Linux , both 32 and 64 bits
Apple OSX(Darwin), both 32 and 64bits.
All hardware with x86/x86_64/arm/mips cpu.

Summary

Simple, also stable enough.
High-performance( CN, EN ): single-thread, async socket, event/st-thread driven.
High-concurrency( CN, EN ), 6000+ connections(500kbps), 900Mbps, CPU 90.2%, 41MB
Support RTMP Origin Server( CN, EN )
Support RTMP Edge Server( CN, EN ) for CDN, push/pull stream from any RTMP server
Support single process; no multiple processes.
Support Vhost( CN, EN ), support __defaultVhost__.
Support RTMP( CN, EN ) live streaming; no vod streaming.
Support Apple HLS(m3u8)( CN, EN ) live streaming.
Support HLS audio-only( CN, EN ) live streaming.
Support Reload( CN, EN ) config to enable changes.
Support cache last gop( CN, EN ) for flash player to fast startup.
Support listen at multiple ports.
Support long time(>4.6hours) publish/play.
Support Forward( CN, EN ) in master-slave mode.
Support live stream Transcoding( CN, EN ) by ffmpeg.
Support ffmpeg( CN, EN ) filters(logo/overlay/crop), x264 params, copy/vn/an.
Support audio transcode( CN, EN ) only, speex/mp3 to aac
Support http callback api hooks( CN, EN )(for authentication and injection).
Support bandwidth test( CN, EN ) api and flash client.
Player, publisher(encoder), and demo pages(jquery+bootstrap)( CN, EN ).
Demo( CN, EN ) video meeting or chat(SRS+cherrypy+jquery+bootstrap).
Full documents in wiki( CN, EN ), both Chinese and English.
Support RTMP(play-publish) library: srs-librtmp( CN, EN )
Support ARM cpu arch( CN, EN ) with rtmp/ssl/hls/librtmp.
Support init.d( CN, EN ) and packge script, log to file.
Support RTMP ATC( CN, EN ) for HLS/HDS to support backup(failover)
Support HTTP RESTful management api( CN, EN ).
Support Ingest( CN, EN ) FILE/HTTP/RTMP/RTSP(RTP, SDP) to RTMP using external tools(e.g ffmepg).
Support DVR( CN, EN ), record live to flv file for vod.
Support tracable log, session based log( CN, EN ).
Support DRM token traverse( CN, EN ) for fms origin authenticate.
Support system full utest on gtest.
Support embeded HTTP server( CN, EN ) for hls(live/vod)
Support vod stream(http flv/hls vod stream)( CN, EN ).
Stable 1.0release branch.
Support publish h264 raw stream( CN, EN ) by srs-librtmp.
Support 6k+ clients, 3Gbps per process.
Suppport English wiki.
Research and simplify st, bug #182.
Support compile srs-librtmp on windows, bug #213.
Support 10k+ clients, 4Gbps per process.
Support publish aac adts raw stream( CN, EN ) by srs-librtmp.
Support 0.1s+ latency, read #257.
Support allow/deny publish/play for all or specified ip( CN, EN ).
Support custom dvr path and http callback, read #179 and 274.
Support rtmp remux to http flv/mp3/aac/ts live stream, read #293( CN, EN ).
Support HLS(h.264+mp3) streaming, read #301.
Rewrite HLS(h.264+aac/mp3) streaming, read #304.
Support Adobe HDS(f4m)( CN, EN ) dynamic streaming.
[experiment] Support push MPEG-TS over UDP to SRS, read #250.
[experiment] Support push RTSP to SRS, read #133.
Start 2.0release branch.
[no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
[no-plan] Support RTMP 302 redirect #92.
[no-plan] Support multiple processes, for both origin and edge
[no-plan] Support adobe RTMFP(flash p2p) protocol.
[no-plan] Support adobe flash refer/token/swf verification.
[no-plan] Support adobe amf3 codec.
[no-plan] Support encryption: RTMPE/RTMPS, HLS DRM
[no-plan] Support RTMPT, http to tranverse firewalls
[no-plan] Support file source, transcoding file to live stream

Releases

2015-02-12, Release v1.0r2, bug fixed, 1.0.27, 59507 lines.
2015-01-15, Release v1.0r1, bug fixed, 1.0.21, 59472 lines.
2014-12-05, Release v1.0r0, all bug fixed, 1.0.10, 59391 lines.
2014-10-09, Release v1.0-beta, all bug fixed, 1.0.0, 59316 lines.
2014-08-03, Release v1.0-mainline7, config utest, all bug fixed. 57432 lines.
2014-07-13, Release v1.0-mainline6, core/kernel/rtmp utest, refine bandwidth(as/js/srslibrtmp library). 50029 lines.
2014-06-27, Release v1.0-mainline5, refine perf 3k+ clients, edge token traverse, srs monitor, 30days online. 41573 lines.
2014-05-28, Release v1.0-mainline4, support heartbeat, tracable log, fix mem leak and bugs. 39200 lines.
2014-05-18, Release v1.0-mainline3, support mips, fms origin, json(http-api). 37594 lines.
2014-04-28, Release v1.0-mainline2, support dvr, android, edge. 35255 lines.
2014-04-07, Release v1.0-mainline, support arm, init.d, http server/api, ingest. 30000 lines.
2013-12-25, Release v0.9, support bandwidth test, player/encoder/chat demos. 20926 lines.
2013-12-08, Release v0.8, support http hooks callback, update st_load. 19186 lines.
2013-12-03, Release v0.7, support live stream transcoding. 17605 lines.
2013-11-29, Release v0.6, support forward stream to origin/edge. 16094 lines.
2013-11-26, Release v0.5, support HLS(m3u8), fragment and window. 14449 lines.
2013-11-10, Release v0.4, support reload config, pause, longtime publish/play. 12500 lines.
2013-11-04, Release v0.3, support vhost, refer, gop cache, listen multiple ports. 11773 lines.
2013-10-25, Release v0.2, support rtmp flash publish, h264, time jitter correct. 10125 lines.
2013-10-23, Release v0.1, support rtmp FMLE/FFMPEG publish, vp6. 8287 lines.
2013-10-17, Created.

History

SRS 2.0 history

v2.0, 2015-04-04, for #304, rewrite annexb mux for ts, refer to apple sample. 2.0.157.
v2.0, 2015-04-03, enhanced avc decode, parse the sps get width+height. 2.0.156.
v2.0, 2015-04-03, for #372, support transform vhost of edge 2.0.155.
v2.0, 2015-03-30, for #366, config hls to disable cleanup of ts. 2.0.154.
v2.0, 2015-03-31, support server cycle handler. 2.0.153.
v2.0, 2015-03-31, support on_hls for http hooks. 2.0.152.
v2.0, 2015-03-31, enhanced hls, support deviation for duration. 2.0.151.
v2.0, 2015-03-30, for #351, support config the m3u8/ts path for hls. 2.0.149.
v2.0, 2015-03-17, for #155, osx(darwin) support demo with nginx and ffmpeg. 2.0.143.
v2.0, 2015-03-15, start 2.0release branch, 80773 lines.
v2.0, 2015-03-14, fix #324, support hstrs(http stream trigger rtmp source) edge mode. 2.0.140.
v2.0, 2015-03-14, for #324, support hstrs(http stream trigger rtmp source) origin mode. 2.0.139.
v2.0, 2015-03-12, fix #328, support adobe hds. 2.0.138.
v2.0, 2015-03-10, fix #155, support osx(darwin) for mac pro. 2.0.137.
v2.0, 2015-03-08, fix #316, http api provides stream/vhost/srs/server bytes, codec and count. 2.0.136.
v2.0, 2015-03-08, fix #310, refine aac LC, support aac HE/HEv2. 2.0.134.
v2.0, 2015-03-06, for #322, fix http-flv stream bug, support multiple streams. 2.0.133.
v2.0, 2015-03-06, refine http request parse. 2.0.132.
v2.0, 2015-03-01, for #179, revert dvr http api. 2.0.128.
v2.0, 2015-02-24, for #304, fix hls bug, write pts/dts error. 2.0.124
v2.0, 2015-02-19, refine dvr, append file when dvr file exists. 2.0.122.
v2.0, 2015-02-19, refine pithy print to more easyer to use. 2.0.121.
v2.0, 2015-02-18, fix #133, support push rtsp to srs. 2.0.120.
v2.0, 2015-02-17, the join maybe failed, should use a variable to ensure thread terminated. 2.0.119.
v2.0, 2015-02-15, for #304, support config default acodec/vcodec. 2.0.118.
v2.0, 2015-02-15, for #304, rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.
v2.0, 2015-02-12, for #304, use stringstream to generate m3u8, add hls_td_ratio. 2.0.116.
v2.0, 2015-02-11, dev code ZhouGuowen for 2.0.115.
v2.0, 2015-02-10, for #311, set pcr_base to dts. 2.0.114.
v2.0, 2015-02-10, fix the bug of ibmf format which decoded in annexb.
v2.0, 2015-02-10, for #310, downcast aac SSR to LC. 2.0.113
v2.0, 2015-02-03, fix #136, support hls without io(in ram). 2.0.112
v2.0, 2015-01-31, for #250, support push MPEGTS over UDP to SRS. 2.0.111
v2.0, 2015-01-29, build libfdk-aac in ffmpeg. 2.0.108
v2.0, 2015-01-25, for #301, hls support h.264+mp3, ok for vlc. 2.0.107
v2.0, 2015-01-25, for #301, http ts stream support h.264+mp3. 2.0.106
v2.0, 2015-01-25, hotfix #268, refine the pcr start at 0, dts/pts plus delay. 2.0.105
v2.0, 2015-01-25, hotfix #151, refine pcr=dts-800ms and use dts/pts directly. 2.0.104
v2.0, 2015-01-23, hotfix #151, use absolutely overflow to make jwplayer happy. 2.0.103
v2.0, 2015-01-22, for #293, support http live ts stream. 2.0.101.
v2.0, 2015-01-19, for #293, support http live flv/aac/mp3 stream with fast cache. 2.0.100.
v2.0, 2015-01-18, for #293, support rtmp remux to http flv live stream. 2.0.99.
v2.0, 2015-01-17, fix #277, refine http server refer to go http-framework. 2.0.98
v2.0, 2015-01-17, for #277, refine http api refer to go http-framework. 2.0.97
v2.0, 2015-01-17, hotfix #290, use iformat only for rtmp input. 2.0.95
v2.0, 2015-01-08, hotfix #281, fix hls bug ignore type-9 send aud. 2.0.93
v2.0, 2015-01-03, fix #274, http-callback support on_dvr when reap a dvr file. 2.0.89
v2.0, 2015-01-03, hotfix to remove the pageUrl for http callback. 2.0.88
v2.0, 2015-01-03, fix #179, dvr support custom filepath by variables. 2.0.87
v2.0, 2015-01-02, fix #211, support security allow/deny publish/play all/ip. 2.0.86
v2.0, 2015-01-02, hotfix #207, trim the last 0 of log. 2.0.85
v2.0, 2014-01-02, fix #158, http-callback check http status code ok(200). 2.0.84
v2.0, 2015-01-02, hotfix #216, http-callback post in application/json content-type. 2.0.83
v2.0, 2014-01-02, fix #263, srs-librtmp flv read tag should init size. 2.0.82
v2.0, 2015-01-01, hotfix #270, memory leak for http client post. 2.0.81
v2.0, 2014-12-12, fix #266, aac profile is object id plus one. 2.0.80
v2.0, 2014-12-29, hotfix #267, the forward dest ep should use server. 2.0.79
v2.0, 2014-12-29, hotfix #268, the hls pcr is negative when startup. 2.0.78
v2.0, 2014-12-22, hotfix #264, ignore NALU when sequence header to make HLS happy. 2.0.76
v2.0, 2014-12-20, hotfix #264, support disconnect publish connect when hls error. 2.0.75
v2.0, 2014-12-12, fix #257, support 0.1s+ latency. 2.0.70
v2.0, 2014-12-08, update wiki for mr(EN, CN) and mw(EN, CN).
v2.0, 2014-12-07, fix #251, 10k+ clients, use queue cond wait and fast vector. 2.0.67
v2.0, 2014-12-05, fix #251, 9k+ clients, use fast cache for msgs queue. 2.0.57
v2.0, 2014-12-04, fix #241, add mw(merged-write) config. 2.0.53
v2.0, 2014-12-04, for #241, support mr(merged-read) config and reload. 2.0.52.
v2.0, 2014-12-04, enable #241 and #248, +25% performance, 2.5k publisher. 2.0.50
v2.0, 2014-12-04, fix #248, improve about 15% performance for fast buffer. 2.0.49
v2.0, 2014-12-03, fix #244, conn thread use cond to wait for recv thread error. 2.0.47.
v2.0, 2014-12-02, merge #239, traverse the token before response connect. 2.0.45.
v2.0, 2014-12-02, srs-librtmp support hijack io apis for st-load. 2.0.42.
v2.0, 2014-12-01, for #237, refine syscall for recv, supports 1.5k clients. 2.0.41.
v2.0, 2014-11-30, add qtcreate project file trunk/src/qt/srs/srs-qt.pro. 2.0.39.
v2.0, 2014-11-29, fix #235, refine handshake, replace union with template method. 2.0.38.
v2.0, 2014-11-28, fix #215, add srs_rtmp_dump tool. 2.0.37.
v2.0, 2014-11-25, update PRIMARY, AUTHORS, CONTRIBUTORS rule. 2.0.32.
v2.0, 2014-11-24, fix #212, support publish aac adts raw stream. 2.0.31.
v2.0, 2014-11-22, fix #217, remove timeout recv, support 7.5k+ 250kbps clients. 2.0.30.
v2.0, 2014-11-21, srs-librtmp add rtmp prefix for rtmp/utils/human apis. 2.0.29.
v2.0, 2014-11-21, refine examples of srs-librtmp, add srs_print_rtmp_packet. 2.0.28.
v2.0, 2014-11-20, fix #212, support publish audio raw frames. 2.0.27
v2.0, 2014-11-19, fix #213, support compile srs-librtmp on windows, bug #213. 2.0.26
v2.0, 2014-11-18, all wiki translated to English. 2.0.23.
v2.0, 2014-11-15, fix #204, srs-librtmp drop duplicated sps/pps(sequence header). 2.0.22.
v2.0, 2014-11-15, fix #203, srs-librtmp drop any video before sps/pps(sequence header). 2.0.21.
v2.0, 2014-11-15, fix #202, fix memory leak of h.264 raw packet send in srs-librtmp. 2.0.20.
v2.0, 2014-11-13, fix #200, deadloop when read/write 0 and ETIME. 2.0.16.
v2.0, 2014-11-13, fix #194, writev multiple msgs, support 6k+ 250kbps clients. 2.0.15.
v2.0, 2014-11-12, fix #194, optmized st for timeout recv. pulse to 500ms. 2.0.14.
v2.0, 2014-11-11, fix #195, remove the confuse code st_usleep(0). 2.0.13.
v2.0, 2014-11-08, fix #191, configure --export-librtmp-project and --export-librtmp-single. 2.0.11.
v2.0, 2014-11-08, fix #66, srs-librtmp support write h264 raw packet. 2.0.9.
v2.0, 2014-10-25, fix #185, AMF0 support 0x0B the date type codec. 2.0.7.
v2.0, 2014-10-24, fix #186, hotfix for bug #186, drop connect args when not object. 2.0.6.
v2.0, 2014-10-24, rename wiki/xxx to wiki/v1_CN_xxx. 2.0.3.
v2.0, 2014-10-19, fix #184, support AnnexB in RTMP body for HLS. 2.0.2
v2.0, 2014-10-18, remove supports for OSX(darwin). 2.0.1.
v2.0, 2014-10-16, revert github srs README to English. 2.0.0.

SRS 1.0 history

v1.0, 2015-02-17, the join maybe failed, should use a variable to ensure thread terminated. 1.0.28.
v1.0, 2015-02-12, 1.0r2 release(1.0.27) released. 59507 lines.
v1.0, 2015-02-11, dev code HuKaiqun for 1.0.27.
v1.0, 2015-02-10, for #310, the aac profile must be object plus one. 1.0.26
v1.0, 2015-01-25, hotfix #268, refine the pcr start at 0, dts/pts plus delay. 1.0.25
v1.0, 2015-01-25, hotfix #151, refine pcr=dts-800ms and use dts/pts directly. 1.0.24
v1.0, 2015-01-23, hotfix #151, use absolutely overflow to make jwplayer happy. 1.0.23
v1.0, 2015-01-17, hotfix #290, use iformat only for rtmp input. 1.0.22
v1.0, 2015-01-15, 1.0r1 release(1.0.21) released. 59472 lines.
v1.0, 2015-01-08, hotfix #281, fix hls bug ignore type-9 send aud. 1.0.20
v1.0, 2015-01-03, hotfix to remove the pageUrl for http callback. 1.0.19
v1.0, 2015-01-02, hotfix #207, trim the last 0 of log. 1.0.18
v1.0, 2015-01-02, hotfix #216, http-callback post in application/json content-type. 1.0.17
v1.0, 2015-01-01, hotfix #270, memory leak for http client post. 1.0.16
v1.0, 2014-12-29, hotfix #267, the forward dest ep should use server. 1.0.15
v1.0, 2014-12-29, hotfix #268, the hls pcr is negative when startup. 1.0.14
v1.0, 2014-12-22, hotfix #264, ignore NALU when sequence header to make HLS happy. 1.0.12
v1.0, 2014-12-20, hotfix #264, support disconnect publish connect when hls error. 1.0.11
v1.0, 2014-12-05, 1.0 release(1.0.10) released. 59391 lines.
v1.0, 2014-10-09, 1.0 beta(1.0.0) released. 59316 lines.
v1.0, 2014-10-08, fix #151, always reap ts whatever audio or video packet. 0.9.223.
v1.0, 2014-10-08, fix #162, failed if no epoll. 0.9.222.
v1.0, 2014-09-30, fix #180, crash for multiple edge publishing the same stream. 0.9.220.
v1.0, 2014-09-26, fix hls bug, refine config and log, according to clion of jetbrains. 0.9.216.
v1.0, 2014-09-25, fix #177, dvr segment add config dvr_wait_keyframe. 0.9.213.
v1.0, 2014-08-28, fix #167, add openssl includes to utest. 0.9.209.
v1.0, 2014-08-27, max connections is 32756, for st use mmap default. 0.9.209
v1.0, 2014-08-24, fix #150, forward should forward the sequence header when retry. 0.9.208.
v1.0, 2014-08-22, for #165, refine dh wrapper, ensure public key is 128bytes. 0.9.206.
v1.0, 2014-08-19, for #160, support forward/edge to flussonic, disable debug_srs_upnode to make flussonic happy. 0.9.201.
v1.0, 2014-08-17, for #155, refine for osx, with ssl/http, disable statistics. 0.9.198.
v1.0, 2014-08-06, fix #148, simplify the RTMP handshake key generation. 0.9.191.
v1.0, 2014-08-06, fix #147, support identify the srs edge. 0.9.190.
v1.0, 2014-08-03, 1.0 mainline7(0.9.189) released. 57432 lines.
v1.0, 2014-08-03, fix #79, fix the reload remove edge assert bug. 0.9.189.
v1.0, 2014-08-03, fix #57, use lock(acquire/release publish) to avoid duplicated publishing. 0.9.188.
v1.0, 2014-08-03, fix #85, fix the segment-dvr sequence header missing. 0.9.187.
v1.0, 2014-08-03, fix #145, refine ffmpeg log, check abitrate for libaacplus. 0.9.186.
v1.0, 2014-08-03, fix #143, fix retrieve sys stat bug for all linux. 0.9.185.
v1.0, 2014-08-02, fix #138, fix http hooks bug, regression bug. 0.9.184.
v1.0, 2014-08-02, fix #142, fix tcp stat slow bug, use /proc/net/sockstat instead, refer to 'ss -s'. 0.9.183.
v1.0, 2014-07-31, fix #141, support tun0(vpn network device) ip retrieve. 0.9.179.
v1.0, 2014-07-27, support partially build on OSX(Darwin). 0.9.177
v1.0, 2014-07-27, api connections add udp, add disk iops. 0.9.176
v1.0, 2014-07-26, complete config utest. 0.9.173
v1.0, 2014-07-26, fix #124, gop cache support disable video in publishing. 0.9.171.
v1.0, 2014-07-23, fix #121, srs_info detail log compile failed. 0.9.168.
v1.0, 2014-07-19, fix #119, use iformat and oformat for ffmpeg transcode. 0.9.163.
v1.0, 2014-07-13, 1.0 mainline6(0.9.160) released. 50029 lines.
v1.0, 2014-07-13, refine the bandwidth check/test, add as/js library, use srs-librtmp for linux tool. 0.9.159
v1.0, 2014-07-12, complete rtmp stack utest. 0.9.156
v1.0, 2014-07-06, fix #81, fix HLS codec info, IOS ok. 0.9.153.
v1.0, 2014-07-06, fix #103, support all aac sample rate. 0.9.150.
v1.0, 2014-07-05, complete kernel utest. 0.9.149
v1.0, 2014-06-30, fix #111, always use 31bits timestamp. 0.9.143.
v1.0, 2014-06-28, response the call message with null. 0.9.137
v1.0, 2014-06-28, fix #110, thread start segment fault, thread cycle stop destroy thread. 0.9.136
v1.0, 2014-06-27, fix #109, fix the system jump time, adjust system startup time. 0.9.135
v1.0, 2014-06-27, 1.0 mainline5(0.9.134) released. 41573 lines.
v1.0, 2014-06-27, SRS online 30days with RTMP/HLS.
v1.0, 2014-06-25, fix #108, support config time jitter for encoder non-monotonical stream. 0.9.133
v1.0, 2014-06-23, support report summaries in heartbeat. 0.9.132
v1.0, 2014-06-22, performance refine, support 3k+ connections(270kbps). 0.9.130
v1.0, 2014-06-21, support edge token traverse, fix #104. 0.9.129
v1.0, 2014-06-19, add connections count to api summaries. 0.9.127
v1.0, 2014-06-19, add srs bytes and kbps to api summaries. 0.9.126
v1.0, 2014-06-18, add network bytes to api summaries. 0.9.125
v1.0, 2014-06-14, fix #98, workaround for librtmp ping(fmt=1,cid=2 fresh stream). 0.9.124
v1.0, 2014-05-29, support flv inject and flv http streaming with start=bytes. 0.9.122
v1.0, 2014-05-28, 1.0 mainline4(0.9.120) released. 39200 lines.
v1.0, 2014-05-27, fix #87, add source id for full trackable log. 0.9.120
v1.0, 2014-05-27, fix #84, unpublish when edge disconnect. 0.9.119
v1.0, 2014-05-27, fix #89, config to /dev/null to disable ffmpeg log. 0.9.117
v1.0, 2014-05-25, fix #76, allow edge vhost to add or remove. 0.9.114
v1.0, 2014-05-24, Johnny contribute ossrs.net. karthikeyan start to translate wiki to English.
v1.0, 2014-05-22, fix #78, st joinable thread must be stop by other threads, 0.9.113
v1.0, 2014-05-22, support amf0 StrictArray(0x0a). 0.9.111.
v1.0, 2014-05-22, support flv parser, add amf0 to librtmp. 0.9.110
v1.0, 2014-05-22, fix #74, add tcUrl for http callback on_connect, 0.9.109
v1.0, 2014-05-19, support http heartbeat, 0.9.107
v1.0, 2014-05-18, 1.0 mainline3(0.9.105) released. 37594 lines.
v1.0, 2014-05-18, support http api json, to PUT/POST. 0.9.105
v1.0, 2014-05-17, fix #72, also need stream_id for send_and_free_message. 0.9.101
v1.0, 2014-05-17, rename struct to class. 0.9.100
v1.0, 2014-05-14, fix #67 pithy print, stage must has a age. 0.9.98
v1.0, 2014-05-13, fix mem leak for delete[] SharedPtrMessage array. 0.9.95
v1.0, 2014-05-12, refine the kbps calc module. 0.9.93
v1.0, 2014-05-12, fix bug #64: install_dir=DESTDIR+PREFIX
v1.0, 2014-05-08, fix #36: never directly use *(int32_t*) for arm.
v1.0, 2014-05-08, fix #60: support aggregate message
v1.0, 2014-05-08, fix #59, edge support FMS origin server. 0.9.92
v1.0, 2014-05-06, fix #50, ubuntu14 build error.
v1.0, 2014-05-04, support mips linux.
v1.0, 2014-04-30, fix bug #34: convert signal to io thread. 0.9.85
v1.0, 2014-04-29, refine RTMP protocol completed, to 0.9.81
v1.0, 2014-04-28, 1.0 mainline2(0.9.79) released. 35255 lines.
v1.0, 2014-04-28, support full edge RTMP server. 0.9.79
v1.0, 2014-04-27, support basic edge(play/publish) RTMP server. 0.9.78
v1.0, 2014-04-25, add donation page. 0.9.76
v1.0, 2014-04-21, support android app to start srs for internal edge. 0.9.72
v1.0, 2014-04-19, support tool over srs-librtmp to ingest flv/rtmp. 0.9.71
v1.0, 2014-04-17, support dvr(record live to flv file for vod). 0.9.69
v1.0, 2014-04-11, add speex1.2 to transcode flash encoder stream. 0.9.58
v1.0, 2014-04-10, support reload ingesters(add/remov/update). 0.9.57
v1.0, 2014-04-07, 1.0 mainline(0.9.55) released. 30000 lines.
v1.0, 2014-04-07, support ingest file/stream/device.
v1.0, 2014-04-05, support http api and http server.
v1.0, 2014-04-03, implements http framework and api/v1/version.
v1.0, 2014-03-30, fix bug for st detecting epoll failed, force st to use epoll.
v1.0, 2014-03-29, add wiki Performance for RaspberryPi.
v1.0, 2014-03-29, add release binary package for raspberry-pi.
v1.0, 2014-03-26, support RTMP ATC for HLS/HDS to support backup(failover).
v1.0, 2014-03-23, support daemon, default start in daemon.
v1.0, 2014-03-22, support make install/install-api and uninstall.
v1.0, 2014-03-22, add ./etc/init.d/srs, refine to support make clean then make.
v1.0, 2014-03-21, write pid to ./objs/srs.pid.
v1.0, 2014-03-20, refine hls code, support pure audio HLS.
v1.0, 2014-03-19, add vn/an for FFMPEG to drop video/audio for radio stream.
v1.0, 2014-03-19, refine handshake, client support complex handshake, add utest.
v1.0, 2014-03-16, fix bug on arm of st, the sp change from 20 to 8, for respberry-pi, @see commit
v1.0, 2014-03-16, support ARM(debian armhf, v7cpu) with rtmp/ssl/hls/librtmp.
v1.0, 2014-03-12, finish utest for amf0 codec.
v1.0, 2014-03-06, add gperftools for mem leak detect, mem/cpu profile.
v1.0, 2014-03-04, add gest framework for utest, build success.
v1.0, 2014-03-02, add wiki srs-librtmp, SRS for arm, product
v1.0, 2014-03-02, srs-librtmp, client publish/play library like librtmp.
v1.0, 2014-03-01, modularity, extract core/kernel/rtmp/app/main module.
v1.0, 2014-02-28, support arm build(SRS/ST), add ssl to 3rdparty package.
v1.0, 2014-02-28, add wiki BuildArm, FFMPEG, Reload
v1.0, 2014-02-27, add wiki LowLatency, HTTPCallback, ServerSideScript, IDE
v1.0, 2014-01-19, add wiki DeliveryHLS
v1.0, 2014-01-12, add wiki HowToAskQuestion, RtmpUrlVhost
v1.0, 2014-01-11, fix jw/flower player pause bug, which send closeStream actually.
v1.0, 2014-01-05, add wiki Build, Performance, Forward
v1.0, 2014-01-01, change listen(512), chunk-size(60000), to improve performance.
v1.0, 2013-12-27, merge from wenjie, the bandwidth test feature.
v0.9, 2013-12-25, v0.9 released. 20926 lines.
v0.9, 2013-12-25, fix the bitrate bug(in Bps), use enhanced microphone.
v0.9, 2013-12-22, demo video meeting or chat(SRS+cherrypy+jquery+bootstrap).
v0.9, 2013-12-22, merge from wenjie, support banwidth test.
v0.9, 2013-12-22, merge from wenjie: support set chunk size at vhost level
v0.9, 2013-12-21, add players for play and publish.
v0.9, 2013-12-15, ensure the HLS(ts) is continous when republish stream.
v0.9, 2013-12-15, fix the hls reload bug, feed it the sequence header.
v0.9, 2013-12-15, refine protocol, use int64_t timestamp for ts and jitter.
v0.9, 2013-12-15, support set the live queue length(in seconds), drop when full.
v0.9, 2013-12-15, fix the forwarder reconnect bug, feed it the sequence header.
v0.9, 2013-12-15, support reload the hls/forwarder/transcoder.
v0.9, 2013-12-14, refine the thread model for the retry threads.
v0.9, 2013-12-10, auto install depends tools/libs on centos/ubuntu.
v0.8, 2013-12-08, v0.8 released. 19186 lines.
v0.8, 2013-12-08, support http hooks: on_connect/close/publish/unpublish/play/stop.
v0.8, 2013-12-08, support multiple http hooks for a event.
v0.8, 2013-12-07, support http callback hooks, on_connect.
v0.8, 2013-12-07, support network based cli and json result, add CherryPy 3.2.4.
v0.8, 2013-12-07, update http/hls/rtmp load test tool st_load, use SRS rtmp sdk.
v0.8, 2013-12-06, support max_connections, drop if exceed.
v0.8, 2013-12-05, support log_dir, write ffmpeg log to file.
v0.8, 2013-12-05, fix the forward/hls/encoder bug.
v0.7, 2013-12-03, v0.7 released. 17605 lines.
v0.7, 2013-12-01, support dead-loop detect for forwarder and transcoder.
v0.7, 2013-12-01, support all ffmpeg filters and params.
v0.7, 2013-11-30, support live stream transcoder by ffmpeg.
v0.7, 2013-11-30, support --with/without -ffmpeg, build ffmpeg-2.1.
v0.7, 2013-11-30, add ffmpeg-2.1, x264-core138, lame-3.99.5, libaacplus-2.0.2.
v0.6, 2013-11-29, v0.6 released. 16094 lines.
v0.6, 2013-11-29, add performance summary, 1800 clients, 900Mbps, CPU 90.2%, 41MB.
v0.6, 2013-11-29, support forward stream to other edge server.
v0.6, 2013-11-29, support forward stream to other origin server.
v0.6, 2013-11-28, fix memory leak bug, aac decode bug.
v0.6, 2013-11-27, support --with or --without -hls and -ssl options.
v0.6, 2013-11-27, support AAC 44100HZ sample rate for iphone, adjust the timestamp.
v0.5, 2013-11-26, v0.5 released. 14449 lines.
v0.5, 2013-11-24, support HLS(m3u8), fragment and window.
v0.5, 2013-11-24, support record to ts file for HLS.
v0.5, 2013-11-21, add ts_info tool to demux ts file.
v0.5, 2013-11-16, add rtmp players(OSMF/jwplayer5/jwplayer6).
v0.4, 2013-11-10, v0.4 released. 12500 lines.
v0.4, 2013-11-10, support config and reload the pithy print.
v0.4, 2013-11-09, support reload config(vhost and its detail).
v0.4, 2013-11-09, support reload config(listen and chunk_size) by SIGHUP(1).
v0.4, 2013-11-09, support longtime(>4.6hours) publish/play.
v0.4, 2013-11-09, support config the chunk_size.
v0.4, 2013-11-09, support pause for live stream.
v0.3, 2013-11-04, v0.3 released. 11773 lines.
v0.3, 2013-11-04, support refer/play-refer/publish-refer.
v0.3, 2013-11-04, support vhosts specified config.
v0.3, 2013-11-02, support listen multiple ports.
v0.3, 2013-11-02, support config file in nginx-conf style.
v0.3, 2013-10-29, support pithy print log message specified by stage.
v0.3, 2013-10-28, support librtmp without extended-timestamp in 0xCX chunk packet.
v0.3, 2013-10-27, support cache last gop for client fast startup.
v0.2, 2013-10-25, v0.2 released. 10125 lines.
v0.2, 2013-10-25, support flash publish.
v0.2, 2013-10-25, support h264/avc codec by rtmp complex handshake.
v0.2, 2013-10-24, support time jitter detect and correct algorithm
v0.2, 2013-10-24, support decode codec type to cache the h264/avc sequence header.
v0.1, 2013-10-23, v0.1 released. 8287 lines.
v0.1, 2013-10-23, support basic amf0 codec, simplify the api using c-style api.
v0.1, 2013-10-23, support shared ptr msg for zero memory copy.
v0.1, 2013-10-22, support vp6 codec with rtmp protocol specified simple handshake.
v0.1, 2013-10-20, support multiple flash client play live streaming.
v0.1, 2013-10-20, support FMLE/FFMPEG publish live streaming.
v0.1, 2013-10-18, support rtmp message2chunk protocol(send_message).
v0.1, 2013-10-17, support rtmp chunk2message protocol(recv_message).

Performance

Performance benchmark history, on virtual box.

See also: Performance for x86/x64 Test Guide
See also: Performance for RaspberryPi

Play benchmark

The play benchmark by st-load:

Update	SRS	Clients	Type	CPU	Memory	Commit
2013-11-28	0.5.0	1.8k(1800)	players	90%	41MB	-
2014-07-12	0.9.156	1.8k(1800)	players	68%	38MB	-
2014-07-12	0.9.156	2.7k(2700)	players	89%	61MB	commit
2014-11-11	1.0.5	2.7k(2700)	players	85%	66MB	-
2014-11-11	2.0.12	2.7k(2700)	players	85%	66MB	-
2014-11-12	2.0.14	2.7k(2700)	players	69%	59MB	-
2014-11-12	2.0.14	3.5k(3500)	players	95%	78MB	commit
2014-11-13	2.0.15	6.0k(6000)	players	82%	203MB	commit
2014-11-22	2.0.30	7.5k(7500)	players	87%	320MB	commit
2014-12-05	2.0.55	8.0k(8000)	players	89%	360MB	(mw_sleep=350) commit
2014-12-05	2.0.57	9.0k(9000)	players	90%	468MB	commit
2014-12-07	2.0.67	10k(10000)	players	95%	656MB	commit

Publish benchmark

The publish benchmark by st-load:

Update	SRS	Clients	Type	CPU	Memory	Commit
2014-12-03	1.0.10	1.2k(1200)	publishers	96%	43MB	-
2014-12-03	2.0.12	1.2k(1200)	publishers	96%	43MB	-
2014-12-03	2.0.47	1.2k(1200)	publishers	84%	76MB	commit
2014-12-03	2.0.47	1.4k(1400)	publishers	95%	140MB	-
2014-12-03	2.0.48	1.4k(1400)	publishers	95%	140MB	commit
2014-12-04	2.0.49	1.4k(1400)	publishers	68%	144MB	-
2014-12-04	2.0.49	2.5k(2500)	publishers	95%	404MB	commit
2014-12-04	2.0.51	2.5k(2500)	publishers	91%	259MB	commit
2014-12-04	2.0.52	4.0k(4000)	publishers	80%	331MB	(mr_sleep=350) commit

Latency benchmark

The latency between encoder and player with realtime config( CN, EN ):

Update	SRS	VP6	H.264	VP6+mp3	H.264+mp3
2014-12-03	1.0.10	0.4s	0.4s	0.9s	1.2s
2014-12-12	2.0.70	0.1s	0.4s	1.0s	0.9s
2014-12-16	2.0.72	0.1s	0.4s	0.8s	0.6s

We use FMLE as encoder for benchmark. The latency of server is 0.1s+, and the bottleneck is the encoder. For more information, read bug #257.

Architecture

SRS always use the most simple architecture to support complex transaction.

System arch: the system structure and arch.
Modularity arch: the main modularity of SRS.
Stream arch: the stream dispatch arch of SRS.
RTMP cluster arch: the RTMP origin and edge cluster arch.
Multiple processes arch (by wenjie): the multiple process of SRS.
CLI arch: the cli arch for SRS, api to manage SRS.
Bandwidth specification: the bandwidth test specification of SRS.

System Architecture

+------------------------------------------------------+
|             SRS(Simple RTMP Server)                  |
+---------------+---------------+-----------+----------+
|   API/hook    |   Transcoder  |    HLS    |   RTMP   |
|  http-parser  |  FFMPEG/x264  |  NGINX/ts | protocol |
+---------------+---------------+-----------+----------+
|              Network(state-threads)                  |
+------------------------------------------------------+
|      All Linux(RHEL,CentOS,Ubuntu,Fedora...)         |
+------------------------------------------------------+

Modularity Architecture

+------------------------------------------------------+
|             Main(srs/bandwidth/librtmp)              |
+------------------------------------------------------+
|           App(Server/Client application)             |
+------------------------------------------------------+
|               RTMP(Protocol stack)                   |
+------------------------------------------------------+
|      Kernel(depends on Core, provides error/log)     |
+------------------------------------------------------+
|         Core(depends only on system apis)            |
+------------------------------------------------------+

Stream Architecture

                   +---------+              +----------+
                   + Publish +              +  Deliver |
                   +---|-----+              +----|-----+
+----------------------+-------------------------+----------------+
|     Input            | SRS(Simple RTMP Server) |     Output     |
+----------------------+-------------------------+----------------+
|    Encoder(1)        |   +-> RTMP protocol ----+-> RTMP player  |
|  (FMLE,FFMPEG, -rtmp-+->-+-> HLS/HTTP ---------+-> M3u8 player  |
|  Flash,XSPLIT,       |   +-> FLV/MP3/Aac/Ts ---+-> HTTP player  |
|  ......)             |   +-> Fowarder ---------+-> RTMP server  |
|                      |   +-> Transcoder -------+-> RTMP server  |
|                      |   +-> DVR --------------+-> Flv file     |
|                      |   +-> BandwidthTest ----+-> flash        |
+----------------------+                         |                |
|  MediaSource(2)      |                         |                |
|  (RTSP,FILE,         |                         |                |
|   HTTP,HLS,   --pull-+->-- Ingester ----(rtmp)-+-> SRS          |
|   Device,            |                         |                |
|   ......)            |                         |                |
+----------------------+                         |                |
|  MediaSource(2)      |                         |                |
|  (RTSP,FILE,         |                         |                |
|   HTTP,HLS,   --push-+->-- Streamer ----(rtmp)-+-> SRS          |
|   Device,            |                         |                |
|   ......)            |                         |                |
+----------------------+-------------------------+----------------+

Remark:
(1) Encoder: encoder must push RTMP stream to SRS server.
(2) MediaSource: any media source, which can be ingest by ffmpeg.
(3) Ingester: SRS will fork a process to run ffmpeg(or your application) 
to ingest any input to rtmp, push to SRS. Read Ingest.
(4) Streamer: SRS will listen for some protocol and accept stream push 
over some protocol and remux to rtmp to SRS. Read Streamer.

HDS/HLS origin backup

                        +----------+        +----------+
               +--ATC->-+  server  +--ATC->-+ packager +-+   +---------+
+----------+   | RTMP   +----------+ RTMP   +----------+ |   | Reverse |    +-------+
| encoder  +->-+                                         +->-+  Proxy  +-->-+  CDN  +
+----------+   |        +----------+        +----------+ |   | (nginx) |    +-------+
               +--ATC->-+  server  +--ATC->-+ packager +-+   +---------+
                 RTMP   +----------+ RTMP   +----------+

RTMP cluster(origin/edge) Architecture

Remark: cluster over edge, see Edge Remark: cluster over forward, see Forward

+---------+       +-----------------+     +-----------------------+ 
+ Encoder +--+-->-+  SRS(RTMP Edge) +--->-+     (RTMP Origin)     | 
+---------+  |    +-----------------+     |   SRS/FMS/NGINX-RTMP  |
             |                            |    Red5/HELIX/CRTMP   |
             +-------------------------->-+         ......        |
                                          +-----------------------+ 
Schema#1: Any RTMP encoder push RTMP stream to RTMP (origin/edge)server,
    where SRS RTMP Edge server will forward stream to origin.


+-------------+    +-----------------+      +--------------------+
| RTMP Origin +-->-+  SRS(RTMP Edge) +--+->-+  Client(RTMP/HLS)  |
+-------------+    +-----------------+  |   |  Flash/IOS/Android |
                                        |   +--------------------+
                                        |
                                        |   +-----------------+
                                        +->-+  SRS(RTMP Edge) +
                                            +-----------------+
Schema#2: SRS RTMP Edge server pull stream from origin (or upstream SRS 
    RTMP Edge server), then delivery to Client.

Bandwidth Test Workflow

   +------------+                    +----------+
   |  Client    |                    |  Server  |
   +-----+------+                    +-----+----+
         |                                 |
         |   connect vhost------------->   |
         |   <-----------result(success)   |
         |                                 |
         |   <----------call(start play)   |
         |   result(playing)---------->    |
         |   <-------------data(playing)   |
         |   <-----------call(stop play)   |
         |   result(stopped)---------->    |
         |                                 |
         |   <-------call(start publish)   |
         |   result(publishing)------->    |
         |   data(publishing)--------->    |
         |   <--------call(stop publish)   |
         |   result(stopped)(1)------->    |
         |                                 |
         |   <--------------------report   |
         |   final(2)----------------->    |
         |           <END>                 |
         
@See: class SrsBandwidth comments.

Beijing, 2013.10
Winlin

han4235 / simple-rtmp-server Goto Github PK

simple-rtmp-server's Introduction