webrtc chrome

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The simplest example of WEBRTC

The online example of WEBRTC is mostly code, the following is an example of a WEBRTC-to-one sample of code simplification, which is tested under Chrome 37. Where iceserver can be omitted, there is no iceserver when the same LAN can still communicate.Client code:When WEBRTC is implemented, the signaling server is requir

Cross-platform WEBRTC client framework: OPENWEBRTC

WEBRTC IOS Framework compilation http://www.th7.cn/Program/IOS/201502/390418.shtml WebRTC in webkit:http://www.webrtcinwebkit.org/ OPENWEBRTC is designed for flexibility and modularity. The bulk of the API layer is implemented in JavaScript, making it super fast to modify and extend with new functionality. Below is a simplified sketch of the architecture. OPENWEBRTC is an open-source, cross-platform,

Google open real-time communication framework WebRTC source code

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

WEBRTC Start-APPRTC Server Setup

Introduction APPRTC is what, webrtc.org official Experience App Ingredients: ubuntu14.04, other Linux versions are not limited, the official does not specify Chrome M51+stunnle Https://www.stunnel.org/ind Ex.htmlrfc5766-turn-server https://code.google.com/archive/p/rfc5766-turn-server/Google App Engine SDK for Pythonapp RTC HTTPS://GITHUB.COM/WEBRTC/APPRTCsteps:Set up proxyBecause of the particularity of th

WEBRTC Source Code

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

The frame and interface of WEBRTC

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

Why always recommend WEBRTC

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;

TELEMCU Video Conferencing Android version WEBRTC client Support

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version of TELEMCU adds Android phone-side WEBRTC video conferencing capabilities, Android phone installs Chrome browser after loading TELEMCU WEBRTC client TeleWeb can directly participate in video conferencing,At the same time, TeleWeb can support two

WebRTC Demo-getusermedia ()

WEBRTC IntroductionWEBRTC provides three types of APIs: MediaStream, namely Getusermedia Rtcpeerconnection Rtcdatachannel Getusermedia has been supported by Chrome, opera and Firefox.rtcpeerconnection is currently supported by Chrome, opera and Firefox. Chrome and opera provide an interface named

Let WebRTC use external audio and video codecs

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

Let WebRTC use external audio and video codecs

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 WebRTC

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/

HTML5 new characteristics of the webrtc[turn]

. RecordeR.connect (context.destination); }3. Real-time data exchangeWEBRTC's other two api,rtcpeerconnection are used for point-to-point connections between browsers, Rtcdatachannel for Point-to-point data communication.The rtcpeerconnection has a browser prefix and is mozrtcpeerconnection in the Chrome browser for the Webkitrtcpeerconnection,firefox browser. Google maintains a library of adapter.js that is used to pump out differences between browse

Vulnerability: WebRTC leaking User IP

point 1 detection point 2If you see the situation shown, there is a risk of vulnerability. The repair of the bug is still very simple.If you are a Firefox browser, then the latest version has been fixed, of course, you can also perform the following steps to troubleshoot.1. Input: About:config2. media.peerconnection.enabled3. Modify its property to FalseIf you are a Google browser, please download install Scriptsafe plugin for repair, if you are unable to download, you can click on the li

WEBRTC video engine with client create code for the daytime

src\webrtc\examples\peerconnection\client\conductor.ccboolconductor::initializepeerconnection()1 webrtc::createpeerconnectionfactory ();src\talk\app\webrtc\peerconnectionfactory.cc1.1 New Rtc::refcountedobject1.2 bool Peerconnectionfactory::initialize ()1.2.1 cricket::mediaengineinterface* media_engine =Peerconnectionfactory::createmediaengine_w()src\talk\media\

Compile WebRTC For Android code in Ubuntu 14.04

Compile WebRTC For Android code in Ubuntu 14.04 Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who encounter similar problems while downloading and

Analysis of NAT penetration in WEBRTC

Said Nat before the penetration of a few about the concept of WEBRTC, may have been the same as the author of the WEBRTC concept of the wrong understanding of the classmate. WebRTC (Network real-time communication) It is a Web browser to support real-time voice dialogue or video dialogue technology, it provides us with video conferencing core technology, includin

"Reprint" WEBRTC congestion control based on GCC (upper)-Algorithm analysis

The greatest feature of real-time streaming media applications is real-time, while latency is the biggest enemy of real-time sex. The processing speed of media data is the important reason of delay, and the network congestion is the main cause of delay from the point of transmission. Network congestion can cause packet loss, and may result in longer data transfer times and increased latency.Congestion control is one of the important methods in real-time streaming media application quality assura

Real-time video communication via WEBRTC (iii)

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In this article we continue to learn about WebRTC 's related Api,rtcpeerconnectiont and Rtcdatachannel.RtcpeerconnectionRtcpeerconnection is a

An introduction to WebRTC's echo cancellation (AEC, AECM) algorithm

reproduced in the original: http://blog.csdn.net/u012931018/article/details/17045077 thank Bo Master. WEBRTC Echo Cancellation (Acoustic ECHOCANCELLATION,AEC, acoustic echocancellation for MOBILE,AECM) algorithm mainly includes the following important modules: echo Time delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC

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