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WebRTC's Past Life
This article is translated by RWEBRTC
WebRTC 技术是激烈的开放的 Web 战争中一大突破。-Brendan Eich, inventor of JavaScriptNo plug-in real-time communicationImagine mobile phones, TVs, and computers communicating through a unified platform. Just imagine, it's easy to add video chat and peer-to data sharing to your site. This is the vision of WebRTC technology.Want to have a try? WEBRT c is ava
In fact, as early as June 2 ago, a friend who worked at Google told me this information. I also got all the source code from WebRTC for the first time, but since my recent work was really busy, this information was not immediately reproduced here. Now, I want to keep an eye on the multimedia applications, learn the technologies in the WebRTC Framework earlier, and use them in actual projects.
Google today
Selection of iOS framework for 1.WebRTCCurrently two more active open source WEBRTC implementations.
Google WebRTC:
Project address is: https://code.google.com/p/webrtc/
Ericsson OPENWEBRTC:
Project address is: HTTPS://GITHUB.COM/ERICSSONRESEARCH/OPENWEBRTCOur Camp David Education is designed to build the iOS app development framework
Google open real-time communication framework WEBRTC source code
In fact, as early as June 2, in Google work friends told me this message, I also first time to get WEBRTC all source code, but because the recent work is busy, not the first time in this reprint this information. Now, I hope to pay attention to the peer of multimedia application, can learn the technology of
Original link: Introduction to WebRTC on Android
Original Author: Dag-inge Aas
Translated by: appear.in
Translator: Dorisminmin
Status: Complete
WebRTC is regarded as a New of web long-term open source development, and is the most important innovation in web development in recent years. WEBRTC allows web developers to add video chats or point
What is WEBRTC?It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So
This article in order to remember for the audio and video communications to make outstanding contributions to the young talent-Lei Hua, really jealous talent!!!A bit sad at the beginning, as a work on the front-line it technical personnel, heard the news is always a bit bad, if you are fortunate to read this article please remember: Pay attention to rest, work is not finished, the body is the capital of the revolution. The last blog interaction with Comrade Ray is as follows:650) this.width=650;
Author: Gustavo Garcia (original link)
Translation: Liu Tong
Bandwidth estimation is probably the most important part of the WEBRTC video engine. The task of the Bandwidth estimation (BWE) module is to determine how much video stream you can send and not network congestion to ensure that video quality is not reduced.
In the previous bandwidth estimation algorithm is very basic, in general, based on the design of packet loss. Usually we start to slowl
Transferred from: http://www.cnblogs.com/fangkm/p/4370492.htmlReprint Please specify source: http://www.cnblogs.com/fangkm/p/4370492.htmlThe previous article simply introduced the next WEBRTC protocol process, which begins with the introduction of frameworks and interfaces.When it comes to frames, instinctively don't know where to start. Once directly from the chromium project on the integration of the source of W
Switch from using WEBRTC to build front-end video chat room--Data channel ChapterIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rt
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
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/
This article is translated from WEBRTC data channelsIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rtcdatachannel API provided by
WEBRTC Technology Group: 234795279Original Address: http://blog.csdn.net/temotemo/article/details/7530504 1, WebRTC purpose WebRTC (Web real-time communication) The ultimate purpose of the project The main is to allow web developers to be based on the browser (chrome\firefox\ ... Fast and easy to develop rich real-time multimedia applications, without the need to
based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), ILBC (audio), pcma,pcmu,g722,g729). NAT supports Ice (Stun+turn)2) Effect measurementTest en
Recently experiment how to let WEBRTC support H264 code, record, for people who need reference.
To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code.
The end result is that the browser can send a video with H264 or receive H264 video.
Note t
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