webrtc server

Recently, our team is developing a app to help people solve problem face to face.We Choose WEBRTC Protocol as our bridge among different platform (Android, IOS, browser etc).But there are a hole in Android 6.0 system, the protocol can not support Android 6.0 system.As we known, Libjingle (Http://mvnrepository.com/artifact/io.pristine) was built in December, 2015,It hasn ' t been updated in least one year. I do not know if

In the next is WEBRTC development novice, at present encountered a problem, turned over to have not understood. Maybe English is not good, look at the document to see blindfolded, so has not found a solution.Development environment:node. JS Server builtI'm using Socket.io to do communications now.Development Purpose:A classmate to B students to initiate a request, B received after the two sides live video.I

write on the frontA: The purpose of writing a blog1. Self-study of the hard self-evident.2. All kinds of information on the Internet is a mixed bag, many are outdated.3. Based on the latest WEBRTC source to share some experience in their work.4. If you write a good people clap, write bad don't spray. Money to hold a field, no money ...Two: Compile compile or compile1. It is best to prepare a VPN, do not think of someone to copy the code to upload to t

, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen

WebRTC represents the best technology in the field of real-time communication since the date of birth. But for a long time, it supported the video encoder only VP8, and later with H265/VP9 as the representative of the birth of the next generation of video Encoders, WebRTC appeared VP9 Codec. The most widely used H264 has been kept out of sight. Until Cisco announces its H264 codec open source for OpenH264,

app Google launched), we'll see what chemistry can produce.So regardless of whether duo succeeds or not, at least we see Google's focus on social and video. In other words, even if duo is unsuccessful, Google will definitely launch other relevant apps to get into this area.2, Google is not always pushing the HTML5 standard? And there is a very important element in HTML5 is WEBRTC, on such an important occasion to show duo (Duo is based on

The newer WEBRTC source code does not have the corresponding vidoeengine with the voiceengine structure, instead of the meidaengine. Mediaengine includes the Mediaengineinterface interface and the fact that the compositemediaengine,compositemediaengine itself is also a template class, and two template references are audio engines and video engines respectively. The compositemediaengine derived class Webrtcmediaengine depends on the template parameters

?? Next Tuesday launch of "Audio and video technology WEBRTC" Open class, Welcome to join!Open Course Links: http://edu.csdn.net/huiyiCourse/detail/90The course's explanatory material and code will be uploaded at the end of the Open class http://download.csdn.net/user/yangzhenpingThe following is the course information:Course Brief IntroductionThe core of WEBRTC originates from Gips.Gips (Global IP Sound) w

1,HTTP://WWW.WEBRTC.ORG/WEBRTC official website, god Horse compilation, God horse download, the solution here is the most authoritative.－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－2,HTTPS://CODE.GOOGLE.COM/P/WEBRTC/WEBRTC Source download location, you can also pay attention to the latest changes anywhere.－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－3,https://webrtchacks.com/is an arti

reproduced in the original: http://www.cnblogs.com/lidabo/p/6842765.html thank you very much. "WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of Global in 2010 IP Solutions A technology that Google has made available to the company. ”

The newer WEBRTC source has no voiceengine structure corresponding to the vidoeengine, replaced by Meidaengine. Mediaengine contains the Mediaengineinterface interface and its implementation compositemediaengine,compositemediaengine itself is also a template class, two template parameters are the audio engine and video engine respectively. Compositemediaengine derived classes Webrtcmediaengine dependent template parameters are Webrtcvoiceengine and We

, YY streaming media Services is basically this function, the structure is as follows.Advantages of the program:1.1, compared with mixer, the service-side pressure is relatively small. and easy to expand.1.2, low latency, especially in combination with SVC, can greatly improve the client experience.Disadvantages of the program:1.1, considering that different clients need different receiving ability, so the architecture of the server is really not simp

This article mainly introduces WEBRTC in each platform debug or log viewing mode, to facilitate troubleshooting, including Bs,pc,android,ios (this series of articles reproduced please indicate the source, blog Park rtc.blacker).1, Browser development:This development method does not need to download and compile WEBRTC source code (many people are "dead" here, but it is really troublesome, the reason is not

 Harnessing Open Source Library WebRTC Fourth chapter-Compiling Macios edition Author: Adam Acknowledgements: Lao Zhang Date: 2015-4-6 Version: 1.0.0 Welcome reprint, has the question feedback q:2780113541, as far as possible consummates series of tutorials. Update Address: Https://github.com/wpc320/webrtc_doc.git Depot_tools proxy settings Reference old Zhang "the best wall in history download WEBRTC co

2016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very mu

ObjectiveThe purpose of this article is to save you 10 hours (or more) of your life, or to waste 10 minutes. WEBRTC's compilation has been called a nightmare as a large cross-platform base library that Google has frequently updated. If you happen to want to compile WEBRTC under Windows, you'd better evaluate your patience in advance.As of now, I have tried almost all the articles in the Chinese blog community, which can be said to have failed. I do no

"WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of global IP in 2010 Solutions company to get a technology, Google Open source of this technology! ”Just for the more than 60 million dollar technology, it is worth studying!

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In the previous article, we explained WebRTC 's overview, history, security, and developer tools. The next step is to explain the process of building

Compile and install WebRTCsvncheckouthttp in Ubuntu: // configure. Then gclientconfighttps: // webrtc. Compile and install WebRTC in Ubuntu Svn checkout http://webrtc.googlecode.com/svn/trunk/ After the download is complete, WebRTC will get a folder named trunk by default, which contains the WebRTC source code, which i

Google's first integration of WebRTC in the Chrome Dev release released this January was a source of widespread concern. Today, Google published a roadmap for the development of WebRTC technology in its blog.WebRTC is a technology for real-time video and audio communication inside the browser, and Google acquired a technology in 2010 to acquire Global IP Solutions. The technology is based on the WHATWG prot