webrtc to rtmp

Nginx itself is a very excellent HTTP server, FFmpeg is a very good audio and video solutions. These two things can be set up by a nginx module Nginx-rtmp-module, a relatively perfect streaming media server.This streaming media server can support RTMP and HLS (Live Http Stream)Starting from InstallationNginx installation refer to my previous one: http://redstarofsleep.iteye.com/blog/2010391The difference is

Nginx rtmp ffmpeg combo mimic HLS LiveA few days ago, Mister said to engage in a LAN live broadcast, think of the rtmp module mentioned before, holding a try to open the mentality of theSystem environment:[Email protected] html]# uname-a Linux localhost.localdomain 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT i686 ATHL On i386 Gnu/linux[[email protected] html]# getconf Long_bit32Nginx Environment:1, Nginx:h

This load test tool is a tool for sharing online games and can be used in http://blog.csdn.net/win_lin/article/details/11835011 or https://github.com/winlinvip/ St-load found a very detailed introduction, including how to install, how to use. Server load Test Tool (St-load): Analog huge concurrency: 2G of memory can open 300k connections. The States-threads-based process. Supports HLS parsing and testing, downloading TS slices after waiting for a slice length, impersonating the clie

The simplest FFmpeg-based streamer (taking pushing RTMP as an example) and ffmpegrtmpThis document records a simple FFmpeg-based streamer (simplest ffmpeg streamer ). The streamer is used to push local video data to the Streaming Media Server. The streamer recorded in this article can use streaming media protocols (such as RTMP, HTTP, UDP, TCP, RTP, etc.) to transfer local media files in MOV/AVI/MKV/MP4/FLV

simple-rtmp-server Server Setup1. Server Image Acquisition GitHub Source Address git clone https://github.com/winlinvip/simple-rtmp-server.git CSDN Mirror Address git clone https://code.csdn.net/winlinvip/srs-csdn.git Oschina Mirroring git clone https://git.oschina.net/winlinvip/srs.oschina.git2. Turn off firewalls and selinux or open ports Shutting down th

1. Compile nginx that can be used in android and add the option -- add-module =/path/nginx-rtmp-module, prepare ffmpeg files and related library files that can be used by shell on android and m3u8 segmenter files.2. Put the compiled nginx files in the data/android. nginx directory of the mobile phone to this directory.3. Compile the configuration file--------------------------------- Nginx. conf start here ---------------------------------------------

IOS RTMP -based video push streamFirst, the basic introductioniOS live a birth, immediately fiery not, a variety of live platforms springing up, because of this, also led to the rapid development of live broadcasting technology, in the IT industry proficient in live technology monkeys are very valuable. Broadcast technology involves a wide range of knowledge, the most important is probably the following: soft and hard decoding. H264, beauty processing

WEBRTC Introduction and simple Application WebRTC, web Real-time communication, Web real-time communication technology. In short, the introduction of real-time communication in a Web browser, including audio and video calls. WEBRTC Real-time communication technology Introduction How to use Media Introduction Signaling Stun

rtmp Protocol Real Time messaging Protocol (Real-time Messaging Protocol protocol) is a private protocol developed by Adobe for Audio, video, and data transmission between flash players and servers. The RTMP protocol is an application layer protocol in the TCP/IP five-tier architecture. The basic data unit in the RTMP protocol is called a message, and when the

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

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_

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC.CreatepeerconnectionfactoryIn Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows:inline rtc::scoped_refptrAs you can see, the last four parameters of Createpeercon

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC. createpeerconnectionfactory In Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows: Inline rtc::scoped_refptr As you

Which framework or library is the best for use WebRTCHttp://stackoverflow.com/questions/24857637/current-state-of-javascript-webrtc-librariesI want to know the which framework or library is the best for use WebRTC. Here are a small and incomplete list of libraries/sdk out there. Any lib that I forgot, feel free to let me know:Libraries: Simplertc Rtcmulticonnection Crocodilertc Lynckia/

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

Itools a bit of a card, but pushing to the server is fast. Preface to push FlowThis blog is a supplement to the:简信 and 微博 ask me how to push the flow this part of how to achieve. So, I've re-integrated RTMP the push flow and merged it into the previous project.Download linkGitHub (full version video live)Blog details (suggested that you join this article to see together)Mac Build Nginx+rtmp Server

To publish FLV data using RTMP, you first need to be aware of the FLV encapsulation format. can view: FLV video Encapsulation Format before you run this instance, you need to build a rtmp server and build an environment where Rtmpdump compiles and runs. can refer to: Nginx build rtmp Streaming Media server Linux compilation Installation Trmpdump (LIBRTMP)

Continue with the unfinished part of the previous article, including the following three sections:1, extension: WEBRTC multiparty calls.2,mcu Multipoint Control Unit.2, Extension: VOIP, telephone, message communication.Note: Translation is not verbatim, but in accordance with their own understanding of the translation, at the same time, in order to facilitate understanding, but also to join some of their own organization language.Reprint please indica

1.WebRTC Backend Service: Room server for callsThe room server is used to create and manage call session status maintenance, is the two sides call or multiparty calls, join and leave the room and so on, we temporarily follow the Google deployment on the Gae platform APPRTC this room server implementation, the Gae The app's source code can be obtained on the github.com. The implementation is a Python-based Gae app that we need to download Goog