Use nginx and nginx-rtmp-module in Ubuntu to set up the correct posture of the Streaming Media Server. nginxrtmpmodule
When I used nginx and nginx-rtmp-module to build a Streaming Media Server, I encountered an embarrassing problem: when I added the nginx-rtmp-module to nginx, I first uninstalled the original nginx, and then downloaded the source code of nginx an
The simplest librtmp-based example: receive (RTMP is saved as FLV), librtmprtmpThis document records a libRTMP-based program for receiving streaming media: Simplest libRTMP Receive. This program can save the RTMP stream to FLV files at a cost. In fact, the program recorded in this article is a "streamlined" RTMPDump. RTMPDump has many functions, so the code is complicated, so many beginners do not know wher
RTMP is designed for transport network streaming, requires support from servers such as Fms,awaza, and provides better copyright protection for streaming media content, and it also needs to pay royalties to adobe itself.
First, the two work differently:
RTMP data requires a dedicated server to receive, such as FMS, Awazal, etc., and then play through the local Flash
to real-time requirements are very high, such as 0.5s or less, this is a good choice. The former mimic Spydroid wrote a proposed RTSP server, in fact, is options,describe,setup,play,pause,teardown these steps, the agreement with the most extensive, on-line introduction is more. To really understand the RTSP protocol, the C + + language is good to see live555.RTMP protocol, own recent research, if interested, can look at my other articles.RELATED Link
on the Adobe platform. MSS, only seen in a few places. Dash, at present, is only the product of the paper.The above agreement, mainly to solve a problem, is adaptive bitrate switching, the above dynamic,adaptive are similar meaning. The main is to use the index file at the same time to give different bitrate video file address, so that the player side, according to the bandwidth situation, adaptive selection of different bitrate video files.At the sa
A similar article above found the time: Problems in rtmp handshake
In FFMPEG, rtmp is implemented. In handshake, C1 is, the time field is filled with 0, and the zero field is filled with client_ver. The generated 1528 is treated as follows: Enter the pseudo-random number first, then, encrypt a key in a certain location. Because we do not pay attention to the pseudo-random number generation algorithm and En
NginxOfRtmpProtocol server
By ahuner
The following configuration allows nginx to receive rtmp streams and play real-time videos on the web.1. OpenSSL Installation
Nginx requires the http_ssl_module module and the OpenSSL library.
Opensll: http://www.openssl.org/
Latest stable version: openssl-1.0.1e
Modify the code for the three files, md2test. C, rc5test. C, jpaketest. c In openssl-1.0.1e \ test
Change dummytest. C to # include "dummytest. c ".
Compi
Standard Specification Learning:
RTMP message structure, including several parts: timestamp: 4 byte, in milliseconds. is flipped when the maximum value is exceeded. Length: The length of the message payload. Type Id:type ID part of the ID range used for rtmp control signaling. There is also a part to use for the upper layer, rtmp is just a pass. This makes
Standard Specification Learning:
RTMP message structure, including several parts: timestamp: 4 byte, in milliseconds. is flipped when the maximum value is exceeded. Length: The length of the message payload. Type Id:type ID part of the ID range used for rtmp control signaling. There is also a part to use for the upper layer, rtmp is just a pass. This makes
Standard Flash Player ACTIONSCRIPT3 statement that plays a flash publish rtmp stream,Netconnection--->netstream--->play--->attachnetstreamThe project, however, has been in a state of stalling.Later added a sentenceNsplayer.buffertime = 0.1;I don't even have a card.The help document says:The default value is 0.1 (One-tenth of a second). To determine the number of seconds currently in the buffer with the Buff
Simple RTMP Server is an open source Rtmp/hls streaming media server written by people in the ROC. Functions are similar to nginx-rtmp-module, enabling the distribution of Rtmp/hls.Reference to the Nginx-rtmp-module: http://blog.csdn.net/redstarofsleep/article/details/450921
Your Nginx already has the RTMP live function, if you also want to count a live channel currently viewing the user volume, you can join the With-http_xslt_module module. The steps are as follows:1. View the original parameters/usr/local/nginx/sbin/nginx-vThe output can be obtained from the original compile-time parameters, such as the author obtained:--user=nginx--group=nginx--with-http_stub_status_module--with-http_gzip_static_module--with-http_ssl_
a user to submit a presentation description to the media server via HTTP or other means. If the representation is multicast, the description contains the multicast address and port number for the media stream, and if the representation is unicast, only the destination address should be provided for security in the presentation description.Invitation to join: A media server can be invited to participate in an ongoing meeting, or play back the media in a presentation, or record all media or a sub
HTTP (Hypertext Transfer Protocol), RTSP (Real time Streaming protocol live streaming protocol), RTMP (Routing Table Maintenance Protocol Routing Tables Maintenance Protocol) is the application layer protocol, Theoretically all can do live, on-demand, in fact, live more than rtmp and RTSP, on-demand is more use RTSP and HTTP.First, common areas:
HTTP (HTTPS) all data as text processing, widely used
The process of doing a lot of problems, the environment actually needs Nginx can, and then is in the playback of the problem, m3u8 format, Mac direct access to support, Apple system native H5 support m3u8, there is also direct access to mobile phone support! But other PC side does not support, tried a lot of not, finally found a support m3u8 format H5 play (https://github.com/huangyaoxin/hLive download on the line js.css loading OK)!Reference: http://blog.csdn.net/zph1234/article/details/5284622
rtmp/flv Learning points of attention
1.RTMP and FLV format-friendly compatibility, mainly embodied in the RTMP package playable audio and video streams, when carefully studied, you will find that the RTMP packet in the package of audio and video data flow, in fact, and Flv/tag package the way the video data is the sa
HTTP (Hyper-Text Transfer Protocol), RTSP (Real Time Streaming protocol live Stream Transfer Protocol), RTMP (Routing Table Maintenance Protocol Routing Tables Maintenance Protocolis the application layer protocol, theoretically can do live, on-demand, in fact, live more than rtmp and RTSP, on-demand and more with RTSP and HTTP. First, common areas:
HTTP (HTTPS) all data as text processing, widely
Recently in window is a platform to do a function to capture audio and video through Obs, and through the RTMP protocol to its encoded compressed data into its own program, since the OBS software with very powerful game recording and desktop recording functions, as well as input, output audio device data acquisition and mixing function , the current fight fish game Live is also used by this software as a recording tool.OBS software because of the use
to the console. Copy the information and save it as a. SDP text file suffix. It can also be used to receive the RTP stream. After "> test. SDP" is added, you can directly Save the SDP information as text.
2.2. Play the RTP that carries the H.264 bare stream.
[Plain]View plaincopy
Ffplay test. SDP
3. rtmp3.1. send the H.264 bare stream to the rtmp server (flashmedia server, red5, etc)
Run the command to send the "Chunwan. h264" of the H.264 bare str
. SDP" is added, you can directly Save the SDP information as text.
2.2. Play the RTP that carries the H.264 bare stream.
ffplay test.sdp
3. rtmp3.1. send the H.264 bare stream to the rtmp server (flashmedia server, red5, etc)
Run the command to send the "Chunwan. h264" of the H.264 bare stream to the rtmp URL where the host is localhost, the application is oflademo, and the path is livestream.
ffmpeg -re
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