README.md



Simple-RTMP-Server

SRS(SIMPLE RTMP Server) over state-threads created in 2013.

SRS is a simple, RTMP/HLS, high-performance, single(plan)/multiple processes, edge(plan)/origin live server.

SRS supports vhost, rtmp, HLS, transcoding, forward, http hooks. 

Blog: http://blog.csdn.net/win_lin 

See also: https://github.com/winlinvip/simple-rtmp-server 

See also: http://winlinvip.github.io/simple-rtmp-server 

TencentQQ: http://url.cn/WAHICw (Group: 212189142)


AUTHORS

The PRIMARY AUTHORS are (and/or have been)(Authors ordered by first contribution): 


winlin(winterserver): http://blog.csdn.net/win_lin 


wenjie(wenjiegit): http://blog.chinaunix.net/uid/25006789.html 


About the primary AUTHORS: 


Contribute important features to SRS. 


Names of all PRIMARY AUTHORS response in NetConnection.connect and metadata. 


And here is an inevitably incomplete list of MUCH-APPRECIATED CONTRIBUTORS --

people who have submitted patches, reported bugs, added translations, helped

answer newbie questions, and generally made SRS that much better: AUTHORS.txt


Wiki

https://github.com/winlinvip/simple-rtmp-server/wiki


Usage(RTMP) for RTMP only

Requires: Centos6.x 64bits, others see Build

Step 0: get SRS 


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


Step 1: build SRS 


./configure --with-ssl --without-hls --without-ffmpeg --without-http && make


Step 2: start SRS 


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


Step 3: Adobe FMLE publish demo live stream 


FMS URL: rtmp://192.168.1.170/live
Stream:  livestream


Step 4: VLC (2.1+) play live stream. 


rtmp url: rtmp://192.168.1.170/live/livestream


Usage(simple) for all features

Requires: Centos6.x 64bits, others see Build

Step -1: get SRS


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


Step 0: build SRS system.


bash scripts/build.sh


Step 1: start SRS all demo features.


bash scripts/run.sh


Step 2: SRS live show: http://your-server-ip 

Step 3: stop SRS demo


bash scripts/stop.sh


Usage(detail) for all features

Requires: Centos6.x 64bits, others see Build

Step 0: get SRS 


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


Step 1: build SRS 


./configure --with-ssl --with-hls --with-ffmpeg --with-http && make


Step 2: start SRS 


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


Step 3(optinal): start SRS listen at 19350 to forward to


./objs/srs -c conf/srs.19350.conf


Step 4(optinal): start nginx for HLS 


sudo ./objs/nginx/sbin/nginx


Step 5(optinal): start http hooks for SRS callback 


python ./research/api-server/server.py 8085


Step 6: publish demo live stream 


FMS URL: rtmp://127.0.0.1/live?vhost=demo.srs.com
Stream:  livestream
FFMPEG to publish the default demo stream:
    for((;;)); do \
        ./objs/ffmpeg/bin/ffmpeg -re -i ./doc/source.200kbps.768x320.flv \
        -vcodec copy -acodec copy \
        -f flv -y rtmp://127.0.0.1/live?vhost=demo.srs.com/livestream; \
        sleep 1; \
    done


Step 7: publish players live stream 


FMS URL: rtmp://127.0.0.1/live?vhost=players
Stream:  livestream
FFMPEG to publish the players demo stream:
    for((;;)); do \
        ./objs/ffmpeg/bin/ffmpeg -re -i ./doc/source.200kbps.768x320.flv \
        -vcodec copy -acodec copy \
        -f flv -y rtmp://127.0.0.1/live?vhost=players/livestream; \
        sleep 1; \
    done


Step 8: add server ip to client hosts as demo. 


# edit the folowing file:
# linux: /etc/hosts
# windows: C:\Windows\System32\drivers\etc\hosts
# where server ip is 192.168.2.111
192.168.2.111 demo.srs.com


Step 9: play live stream. 


players: http://demo.srs.com/players
rtmp url: rtmp://demo.srs.com/live/livestream
m3u8 url: http://demo.srs.com/live/livestream.m3u8
for android: http://demo.srs.com/live/livestream.html


Step 10(optinal): play live stream auto transcoded


rtmp url: rtmp://demo.srs.com/live/livestream_ld
m3u8 url: http://demo.srs.com/live/livestream_ld.m3u8
for android: http://demo.srs.com/live/livestream_ld.html
rtmp url: rtmp://demo.srs.com/live/livestream_sd
m3u8 url: http://demo.srs.com/live/livestream_sd.m3u8
for android: http://demo.srs.com/live/livestream_sd.html


Step 11(optinal): play live stream auto forwarded, the hls dir change to /forward


rtmp url: rtmp://demo.srs.com:19350/live/livestream
m3u8 url: http://demo.srs.com/forward/live/livestream.m3u8
for android: http://demo.srs.com/forward/live/livestream.html
rtmp url: rtmp://demo.srs.com:19350/live/livestream_ld
m3u8 url: http://demo.srs.com/forward/live/livestream_ld.m3u8
for android: http://demo.srs.com/forward/live/livestream_ld.html
rtmp url: rtmp://demo.srs.com:19350/live/livestream_sd
m3u8 url: http://demo.srs.com/forward/live/livestream_sd.m3u8
for android: http://demo.srs.com/forward/live/livestream_sd.html


Step 12(optinal): modify the config and reload it (all features support reload)


killall -1 srs


or use specified signal to reload:


killall -s SIGHUP srs


Architecture

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...)         |
+------------------------------------------------------+


Stream Architecture:

        +---------+              +----------+ 
        + Publish +              +  Deliver | 
        +---|-----+              +----|-----+ 
+-----------+-------------------------+----------------+
| Encoder   | SRS(Simple RTMP Server) |     Client     |
+-----------+-------------------------+----------------+
| (FMLE,    |   +-> RTMP protocol ----+-> Flash Player |
| FFMPEG, --+-> +-> HLS/NGINX --------+-> m3u8 player  |
| Flash,    |   +-> Fowarder ---------+-> RTMP Server  |
| XSPLIT,   |   +-> Transcoder -------+-> RTMP Server  |
|  ...)     |   +-> DVR --------------+-> FILE         |
|           |   +-> BandwidthTest ----+-> Flash/StLoad |
+-----------+-------------------------+----------------+


(plan) RTMP cluster(origin/edge) Architecture:

Remark: cluster over forward, see Cluster

+---------+       +-----------------+     +-----------------------+ 
+ 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.


(plan) SRS Multiple processes Architecture:


                                 +---------------------------+
                           +-----+ worker process(3)         |
+----------------+         |     +---------------------------+
| master process |---(2)---+           
|   (1)manager   |         |     +---------------------------+
+----------------+         +-----+ worker process N          |
                           |     +---------------------------+
                           |
                           |     +---------------------------+
                           +-----+ bandwidth test process(4) |
                                 +---------------------------+
Remark:
(1) master process: to fork processes, schedule fd(client) to the "right" 
    process, forward messages between processes
(2) communication: master process use unix domain socket to communicate 
    with worker processes.
(3) worker process: to provide RTMP streaming service, generate HLS files.
(4) bandwidth test process: for all bandwidth test request, to make the 
    bandwidth testing has minimal effect to streaming service


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.


System Requirements

Supported operating systems and hardware:


All Linux , both 32 and 64 bits
All handware.


Summary


Simple: also stable enough.


High-performance: single-thread, async socket, event/st-thread driven.


With RTMP edge server, support origin server.


NO vod streaming, live streaming only.


With multiple processes, support single process.


Support vhost, support __defaultVhost__.


Support adobe rtmp live streaming.


Support apple HLS(m3u8) live streaming.


Support reload config to enable changes.


Support cache last gop for flash player to fast startup.


Support listen at multiple ports.


Support long time(>4.6hours) publish/play.


High performace, 1800 connections(500kbps), 900Mbps, CPU 90.2%, 41MB


Support forward publish stream to build active-standby cluster.


Support broadcast by forward the stream to other servers(origin/edge).


Support live stream transcoding by ffmpeg.


Support live stream forward(acopy/vcopy) by ffmpeg.


Support ffmpeg filters(logo/overlay/crop), x264 params.


Support audio transcode only, speex/mp3 to aac


Support http callback api hooks(for authentication and injection).


Support bandwidth test api and flash client.


Player, publisher(encoder), and demo pages(jquery+bootstrap). 


Demo video meeting or chat(SRS+cherrypy+jquery+bootstrap). 


[dev] Full documents in wiki, in chineses. 


[plan] Support RTSP(RTP, SDP)


[plan] Support system utest


[plan] Support HLS cluster, use RTMP ATC to generate the TS


[plan] Support RTMP edge server, push/pull stream from any RTMP server


[plan] Support multiple processes, for both origin and edge


[plan] Support network based cli and json result.


[no-plan] Support adobe flash refer/token/swf verification.


[no-plan] Support adobe amf3 codec.


[no-plan] Support dvr(record live to vod file)


[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


Performance


 300 connections, 150Mbps, 500kbps, CPU 18.8%, MEM 5956KB.
 600 connections, 300Mbps, 500kbps, CPU 32.1%, MEM 9808KB.
 900 connections, 450Mbps, 500kbps, CPU 49.9%, MEM 11MB.
1200 connections, 600Mbps, 500kbps, CPU 72.4%, MEM 15MB.
1500 connections, 750Mbps, 500kbps, CPU 81.9%, MEM 28MB.
1800 connections, 900Mbps, 500kbps, CPU 90.2%, MEM 41MB.


[winlin@dev6 srs]$ dstat
----total-cpu-usage---- -dsk/total- ---net/lo-- ---paging-- ---system--
usr sys idl wai hiq siq| read  writ| recv  send|  in   out | int   csw 
 58   9  32   0   0   1|   0  4168k| 277M  277M|   0     0 |  29k   25k
 61   8  30   0   0   1|   0  1168k| 336M  336M|   0     0 |  29k   24k
 63   8  27   0   0   1|   0  2240k| 124M  124M|   0     0 |  32k   33k
 62   8  28   0   0   1|   0  1632k| 110M  110M|   0     0 |  31k   33k
 53   7  40   0   0   1|   0  1360k| 115M  115M|   0     0 |  24k   26k
 51   7  41   0   0   1|   0  1184k| 146M  146M|   0     0 |  24k   27k
 39   6  54   0   0   1|   0  1284k| 105M  105M|   0     0 |  22k   28k
 41   6  52   0   0   1|   0  1264k| 116M  116M|   0     0 |  25k   28k
 48   6  45   0   0   1|   0  1272k| 143M  143M|   0     0 |  27k   27k


See also: Performance Test Guide


Releases


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.


Compare


SRS v0.9: 20926 lines. player/encoder/chat demos. bandwidth test for encoder/CDN.


SRS v0.8: 19186 lines. implements http hooks refer to nginx-rtmp. 


SRS v0.7: 17605 lines. implements transcoding(FFMPEG) feature refer to wowza. 


SRS v0.6: 16094 lines. important feature forward for CDN. 


SRS v0.5: 14449 lines. implements HLS feature refer to nginx-rtmp. 


SRS v0.4: 12500 lines. important feature reload for CDN. 


SRS v0.3: 11773 lines. implements vhost feature refer to FMS. 


SRS v0.2: 10125 lines. implements rtmp protocol stack refer to nginx-rtmp. 


SRS v0.1: 8287 lines. base on state-threads. 


nginx-rtmp v1.0.4: 26786 lines 


nginx v1.5.0: 139524 lines 


History


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, Cluster

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).


Beijing, 2013

Winlin