need to is a browser, but could very well is an application. The reasons for have such an application is several:it May is acting as an MCU, a media recorder, an IVR applicatio n, a Bridge towards a more or less different technology (e.g., SIP, RTMP, or any legacy streaming PLATF ORM) or something else. Such an application, which should implement most, if not all, the WebRTC protocols and technologies, a
ability to record and store media streams for any purpose, which is hard to do on the mesh architecture.Integration with other communication technologiesAnother advantage of using media servers is the ability to communicate with other systems outside the web system, such as the PSTN via SIP trunking or a service streamed over rtmp (like Fackbook live or YouTube live streaming).You can see an example of a previous blog in which the Kurento media serve
QuickTime plugin, but we want a pure browser video stream.Another notable option is Flash Player, which can receive rtmp streams obtained through Wowza conversion rtsp/rtp/h.264. But Flash Player is also a browser plugin, although it is more popular than VLC and QuickTime.In our scenario, we test the same RTSP/RTP stream, but use the WEBRTC compatible browser as the player to play the video without any add
First, download the library files that you need to rely on to install Nginx:1.1, select Source DirectorySelected Directory/usr/local/rtmpCd/usr/local/rtmp1.2, installing the Pcre LibraryCd/usr/local/rtmpTo www. pcre. org download pcre-8.37.tar.gz, then copy to/usr/local/hlsTAR-ZXVF pcre-8.37.tar.gzCD pcre-8.37./configureMakeMake install1.3, installing the Zlib libraryCd/usr/local/rtmpTo www. zlib. NET Download zlib-1.2.8.tar.gz, then copy to/usr/local/hlsTAR-ZXVF zlib-1.2.8.tar.gz CD zlib-1.2.8.
: This article mainly introduces how to configure the rtmp Server for NGINX to implement rtmp and hls live broadcast. if you are interested in PHP Tutorial, refer to it. Introduction to NGINX
First configure:
work_processes4;events{ work_connections1024;}http{ include mime.types; default_type application/octet-stream; sendfileon; keepalive_timeout65; server{ listen80; server
"Getting Started with WebRTC" The first chapter WebRTC introduction?This chapter is a conceptual introduction to WEBRTC.after reading this chapter. You will have a clear understanding of the following: . What is WEBRTC . How to use it . which browsers support1.1. WEBRTC IntroductionWorld Wide Web (WWW) is the early day
A simple Rtmp/hls Live server licensed by the MIT protocol, its core value concept is simplicity and efficiency.How to use:TEP 1:build SRSTar XF simple-rtmp-server-*.*.tar.gzCD Simple-rtmp-server-*.*/trunk./configure--with-ssl--with-hlsMakeStep 2:start SRS./objs/simple_rtmp_server-c conf/srs.confStep 3 (Optional): Start Nginx for HLSsudo./objs/nginx/sbin/nginxSte
Keywords: nignx (http server); Rtmp,hls (Streaming media Services)Thanks for the open source, thanks to the fighting people. Now in some of the work of streaming media services, streaming media server set up online tutorials more into the calf, but look carefully, found a lot of comrades posted tutorials have a little problem, there are some are crawling other people's data to get, typesetting also have problems, let novice confused. This article is a
implementations?Answer 1: The second scenario just described is implemented on the client side, of course, the server also needs to do some work; andThe third kind of solution is mainly in the service-side realization; The relevant advantages and disadvantages above have also been answered, we can refer to the following;2: Does the MAI technology have an open source base version?The solution to 2:p2p can be considered on the basis of WEBRTC, while th
http://neue.v2ex.com/t/316766Although the browser with JS decoding ' play ' video scheme has a few ... Why not one more ...Basic principle:
The server uses websockify to relay an rtmp Stream.
The browser uses NODE-RTMPAPI to parse the RTMP protocol to complete the handshake and Communication.
Extract the H264 video stream
Send to Broadway decode
Repo:https://github.com/yingdev/web-rtmpAt pr
Introduction:First declare I was just a small intern, if there is not correct, I hope you help correct me.First, WEBRTC basic structureFigure A WEBRTC overall structure, from Baidu EncyclopediaFirst of all, WEBRTC the general realization of the idea: we create a web app, and then call in the app's JS Api,js API will invoke the C + + layer API in the browser, the
This paper mainly introduces the RTP/RTCP protocol in WEBRTC, Weizhenwei, the earliest published articles in the Wind network, ID:BEFOIOSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).a prefaceThe RTP/RTCP protocol is the cornerstone of streaming media communications. The RTP protocol defines the packet format for streaming media data over the Internet, while the RTCP pr
first, the network topology structureWEBRTC can also be used as multiparty calls, such as video conferencing, in addition to peer-to-peer communication.
When it comes to multi-party calls, we need to select a schema for our application.
This is a very important decision, because how to organize users is related to the scale of the conference system.
Corresponding to WEBRTC, there are two common network topologies:
Mesh networks and star-shaped netwo
this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds
Static Library Connection Compilation issues:Assert is not defined under the C compilerCeill no connection, no-lm.Http://blog.chinaunix.net/uid-20681545-id-3786786.html-c and-lz cannot be used together with-c no connectionhttp://blog.csdn.net/aiwoziji13/article/details/7330333 Static Library ConnectionHttp://www.cnblogs.com/HuangWj/p/5012120.html vs2010 under the libraryHttps://github.com/ossrs/srs/wiki/v1_CN_SampleRTMP ok!Https://github.com/ossrs/srs/wiki/v2_CN_BuildHttps://github.com/ossrs/srs
This article records some examples of flash-based streaming media processing. The most common streaming media protocol for the Flash platform is rtmp. Some of the previously documented RTMP players/boosters are based on C + +, but no rtmp player/ejector based on ActionScript in Flash was recorded. In fact, the RTMP pla
The simplest example of Flash-based streaming media: RTMP push and receipt (ActionScript) and rtmpactionscriptThis article records some Flash-based streaming media processing examples. The most common Streaming Media Protocol for Flash platforms is RTMP. Some RTMP players/streamer Based on C/C ++ recorded previously, but no R
On the WWDC17, Apple has brought us a big surprise-its browser kernel WebKit will formally support WEBRTC, and future Apple browsers based on the WebKit kernel, such as the MacOS high Sierra, Safari in IOS 11 Browser and Safari Technology Preview version 32, will be used to WEBRTC technology.
The news has attracted countless webrtc developers, even more so that
Scene:1, A call B2, B answer3, A connected with BCommon steps:Both A and B need to initialize the WEBRTC module to create the PeerconnectionfactoryStatus of a in step 11. Create Peerconnection instances through Peerconnectionfactory2. Call Peerconnection.createoffer3, PeerConnection.Observer.onCreateSuccess (final sessiondescription ORIGSDP)4. Send SDP to B5, the following is the collection of Icecandidate, send the mobile phone icecandidate informati
reproduced in this document: Http://www.cnblogs.com/my_life/articles/5593892.html, thank the blogger enthusiastic dedication
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In the live app, rtmp and HLS can basically cover all the clients watching,HLS mainly is the delay is relatively large, the
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