Window环境下 海康视频RTMP推流方法
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Window环境下 海康视频RTMP推流方法,使用到的工具。希望能够帮到大家。
# NGINX-based Media Streaming Server
## nginx-rtmp-module
### Project blog
http://nginx-rtmp.blogspot.com
### Wiki manual
https://github.com/arut/nginx-rtmp-module/wiki/Directives
### Google group
https://groups.google.com/group/nginx-rtmp
https://groups.google.com/group/nginx-rtmp-ru (Russian)
### Donation page (Paypal etc)
http://arut.github.com/nginx-rtmp-module/
### Features
* RTMP/HLS/MPEG-DASH live streaming
* RTMP Video on demand FLV/MP4,
playing from local filesystem or HTTP
* Stream relay support for distributed
streaming: push & pull models
* Recording streams in multiple FLVs
* H264/AAC support
* Online transcoding with FFmpeg
* HTTP callbacks (publish/play/record/update etc)
* Running external programs on certain events (exec)
* HTTP control module for recording audio/video and dropping clients
* 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
* Linux/FreeBSD/MacOS/Windows
### Build
cd to NGINX source directory & run this:
./configure --add-module=/path/to/nginx-rtmp-module
make
make install
Several versions of nginx (1.3.14 - 1.5.0) require http_ssl_module to be
added as well:
./configure --add-module=/path/to/nginx-rtmp-module --with-http_ssl_module
For building debug version of nginx add `--with-debug`
./configure --add-module=/path/to-nginx/rtmp-module --with-debug
[Read more about debug log](https://github.com/arut/nginx-rtmp-module/wiki/Debug-log)
### Windows limitations
Windows support is limited. These features are not supported
* execs
* static pulls
* auto_push
### 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
### Multi-worker live streaming
Module supports multi-worker live
streaming through automatic stream pushing
to nginx workers. This option is toggled with
rtmp_auto_push directive.
### 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 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 webcam {
live on;
# Stream from local webcam
exec_static ffmpeg -f video4linux2 -i /dev/video0 -c:v libx264 -an
-f flv rtmp://localhost:1935/webcam/mystream;
}
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;
}
application mystaticpull {
live on;
# Static pull is started at nginx start
pull rtmp://rtmp4.example.com pageUrl=www.example.com/index.html name=mystream static;
}
# video on demand
application vod {
play /var/flvs;
}
application vod2 {
play /var/mp4s;
}
# 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
# For HLS to work please create a directory in tmpfs (/tmp/hls here)
# for the fragments. The directory contents is served via HTTP (see
# http{} section in config)
#
# Incoming stream must be in H264/AAC. 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 {
live on;
hls on;
hls_path /tmp/hls;
}
# MPEG-DASH is similar to HLS
application dash {
live on;
dash on;
dash_path /tmp/dash;
}
}
}
# HTTP can be used for accessing RTMP stats
http {
server {
listen 8080;
# This URL provides RTMP statistics in XML
location /stat {
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