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2012-08-24 20:46:40 +04:00
doc implemented encrypted handshake 2012-05-03 02:28:21 +04:00
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AUTHORS improved frame timing & added meta files 2012-03-20 03:52:39 +04:00
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ngx_rtmp.c added amf3_* handlers; this adds compatibility with amf3 players (srobe etc) 2012-06-11 01:02:53 +04:00
ngx_rtmp.h added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
ngx_rtmp_access_module.c push is working 2012-05-16 18:04:35 +04:00
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ngx_rtmp_bandwidth.c implemented http/xml/xsl RTMP stats 2012-05-07 15:41:03 +04:00
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ngx_rtmp_cmd_module.c added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
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ngx_rtmp_codec_module.c fixed compilation on FreeBSD 2012-08-13 17:09:05 +04:00
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ngx_rtmp_core_module.c fixed ipv6only field to comply with the new semantics 2012-08-15 15:54:25 +04:00
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ngx_rtmp_handshake.c improved sesion epoch storage; added live stream time field & added time field to statistics 2012-05-25 16:34:42 +04:00
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ngx_rtmp_live_module.h added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
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ngx_rtmp_netcall_module.h added forcing detached netcalls when handler is NULL 2012-04-12 21:18:15 +04:00
ngx_rtmp_notify_module.c added addr=<ip> to HTTP notifications 2012-07-12 16:53:16 +04:00
ngx_rtmp_play_module.c added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
ngx_rtmp_receive.c Merge remote branch 'origin/safe-amf-parser' into video-on-demand 2012-07-06 12:38:24 +04:00
ngx_rtmp_record_module.c added path checks 2012-07-23 17:19:25 +04:00
ngx_rtmp_relay_module.c fixed compilation 2012-07-18 13:36:17 +04:00
ngx_rtmp_relay_module.h reimplemented push reconnect using target tags 2012-07-13 20:08:28 +04:00
ngx_rtmp_send.c added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
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ngx_rtmp_stat_module.c added SwfUrl to statistics 2012-06-29 15:29:18 +04:00
ngx_rtmp_streams.h added AMF shortcuts & project-wide stream numbers; updated mp4 streamer 2012-08-24 20:46:40 +04:00
README.md updated README 2012-08-20 08:40:23 +04:00
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TODO updated TODO list 2012-07-23 17:40:00 +04:00

NGINX-based RTMP server

nginx-rtmp-module

Project page:

http://arut.github.com/nginx-rtmp-module

Wiki manual:

https://github.com/arut/nginx-rtmp-module/wiki

Features:

  • Live streaming of video/audio

  • Video on demand (FLV)

  • Stream relay support for distributed streaming: push & pull models

  • Recording published streams in FLV file

  • H264/AAC support

  • Online transcoding with FFmpeg

  • HLS (HTTP Live Streaming) support; experimental; requires recent libavformat (>= 53.31.100) from ffmpeg (ffmpeg.org)

  • HTTP callbacks on publish/play/record

  • Advanced buffering techniques to keep memory allocations at a minimum level for faster streaming and low memory footprint

  • Proved to work with Wirecast,FMS,Wowza, JWPlayer,FlowPlayer,StrobeMediaPlayback, ffmpeg,avconv,rtmpdump,flvstreamer and many more

  • Statistics in XML/XSL in machine- & human- readable form

Build:

cd to NGINX source directory & run this:

./configure --add-module=<path-to-nginx-rtmp-module>
make
make install

Known issue:

The module does not share data between workers. Because of this live streaming is only available in one-worker mode so far. Video-on-demand has no such limitations.

You can try auto-push branch with multi-worker support if you really need that.

RTMP URL format:

rtmp://rtmp.example.com/app[/name]

app - should match one of application {} blocks in config

name - interpreted by each application can be empty

Example nginx.conf:

rtmp {

    server {

        listen 1935;

        chunk_size 4000;

        # TV mode: one publisher, many subscribers
        application mytv {

            # enable live streaming
            live on;

            # record first 1K of stream
            record all;
            record_path /tmp/av;
            record_max_size 1K;

            # append current timestamp to each flv
            record_unique on;

            # publish only from localhost
            allow publish 127.0.0.1;
            deny publish all;

            #allow play all;
        }

        # Transcoding (ffmpeg needed)
        application big {
            live on;

            # On every pusblished stream run this command (ffmpeg)
            # with substitutions: $app/${app}, $name/${name} for application & stream name.
            #
            # This ffmpeg call receives stream from this application &
            # reduces the resolution down to 32x32. The stream is the published to
            # 'small' application (see below) under the same name.
            #
            # ffmpeg can do anything with the stream like video/audio
            # transcoding, resizing, altering container/codec params etc
            #
            # Multiple exec lines can be specified.

            exec /usr/bin/ffmpeg -re -i rtmp://localhost:1935/$app/$name -vcodec flv -acodec copy -s 32x32 -f flv rtmp://localhost:1935/small/${name};
        }

        application small {
            live on;
            # Video with reduced resolution comes here from ffmpeg
        }

        application mypush {
            live on;

            # Every stream published here
            # is automatically pushed to 
            # these two machines
            push rtmp1.example.com;
            push rtmp2.example.com:1934;
        }

        application mypull {
            live on;

            # Pull all streams from remote machine
            # and play locally
            pull rtmp://rtmp3.example.com pageUrl=www.example.com/index.html;
        }

        # video on demand
        application vod {
            play /var/flvs;
        }

        # Many publishers, many subscribers
        # no checks, no recording
        application videochat {

            live on;

            # The following notifications receive all 
            # the session variables as well as 
            # particular call arguments in HTTP POST
            # request

            # Make HTTP request & use HTTP retcode
            # to decide whether to allow publishing
            # from this connection or not
            on_publish http://localhost:8080/publish;

            # Same with playing
            on_play http://localhost:8080/play;

            # Publish/play end (repeats on disconnect)
            on_done http://localhost:8080/done;

            # All above mentioned notifications receive
            # standard connect() arguments as well as 
            # play/publish ones. If any arguments are sent
            # with GET-style syntax to play & publish
            # these are also included.
            # Example URL:
            #   rtmp://localhost/myapp/mystream?a=b&c=d

            # record 10 video keyframes (no audio) every 2 minutes
            record keyframes;
            record_path /tmp/vc;
            record_max_frames 10;
            record_interval 2m;

            # Async notify about an flv recorded
            on_record_done http://localhost:8080/record_done;

        }


        # HLS (experimental)

        # HLS requires libavformat & should be configured as a separate
        # NGINX module in addition to nginx-rtmp-module:
        # ./configure ... --add-module=/path/to/nginx-rtmp-module/hls ...

        # For HLS to work please create a directory in tmpfs (/tmp/app here)
        # for the fragments. The directory contents is served via HTTP (see
        # http{} section in config)
        #
        # Incoming stream must be in H264/AAC/MP3. For iPhones use baseline H264
        # profile (see ffmpeg example).
        # This example creates RTMP stream from movie ready for HLS:
        #
        # ffmpeg -loglevel verbose -re -i movie.avi  -vcodec libx264 
        #    -vprofile baseline -acodec libmp3lame -ar 44100 -ac 1 
        #    -f flv rtmp://localhost:1935/hls/movie
        #
        # If you need to transcode live stream use 'exec' feature.
        #
        application hls {
            hls on;
            hls_path /tmp/app;
            hls_fragment 5s;
        }

    }
}

# HTTP can be used for accessing RTMP stats
http {

    server {

        listen      8080;

        # This URL provides RTMP statistics in XML
        location /stat {
            rtmp_stat all;

            # Use this stylesheet to view XML as web page
            # in browser
            rtmp_stat_stylesheet stat.xsl;
        }

        location /stat.xsl {
            # XML stylesheet to view RTMP stats.
            # Copy stat.xsl wherever you want
            # and put the full directory path here
            root /path/to/stat.xsl/;
        }

        location /hls {
            # Serve HLS fragments
            alias /tmp/app;
        }

    }
}