webrtc framework

Crosswalk QuickStart, using WEBRTC (HTML) to start developing video callsInstall PythonDownload the installer from http://www.python.org/downloads/After the installation is complete, add the environment variable again.Installing Oracle JDK Download page:http://www.oracle.com/technetwork/java/javase/downloads/Select the Java version to download (recommended Java 7). Select a JDK to download and accept the license agreement. Once downlo

Some time ago in the audio and video version of iOS, so the title changed to Android IOS WebRTC audio and Video development summary, the following summarizes some of the experience in the development process:1. iOS WEBRTC audio and video compilation and download: have android WebRTC compile download experience and then go to get IOS, you will find a lot easier, a

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

It is not recommended to compile individual modules in the WEBRTC separately for use. Yesterday, I was fortunate enough to ask the Google forum about the delay in computing the AECM module, and Project member said churn this delay actually didn't help the AECM effect, which only sped up the convergence of the built-in latency estimator when the AECM started, and if the delay in the update was incorrect, it would even make AE The CM built-in delay

When deploying WEBRTC or SIP-to-peer scenarios, you will often encounter environments that are not penetrated by peerThis is where the tunserver comes in.Here we use turnserver-0.7.3Download confuse dependent librarieswget http://savannah.nongnu.org/download/confuse/confuse-2.7.tar.gzTar zxvf confuse-2.7.tar.gzCD confuse*./configureMake make installDownloadwget http://downloads.sourceforge.net/project/turnserver/turnserver-0.7.3.tar.bz2Tar jxvf turns

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 client-to-

Transferred from: http://www.cnblogs.com/lingyunhu/p/4232348.htmlToss for one weeks to finally set up the environment of Kurento (development stage of the product, some bugs to solve their own), so write a separate document to introduce.The following starts to introduce Kurento, the article from the blog Garden Rtc.blacker, reproduced please explain the source.What is Kurento?Video conferencing involves a one-to-many, many-to-many, broadcast, transcoding, mixing, co-screen, recording, which requ

WebRTC Point-to-point video calling system Main functions:1, based on the WebSocket online user list;2, use WebSocket as signaling channel, build WEBRTC video call.Github:https://github.com/graceup/webrtcDevelopment Ide:myeclipse 8.6 Engineering Code: UTF-8Environmental requirements: 1, TOMCAT requires more than 7.0 of the versionNote: When deploying, you need to change "ws://localhost:8080/" in the Js/con

WebRTC really is not very good to get, currently only the PC-side web page and mobile phone-side web page video. But there are still some problems. 1, both must use Firefox 2, feel pc-side camera shot out of the screen can also, the phone side a little bit of spending 3, enter the room after a period of time to show two video ~~~~APPRTC demo has not been tuned, the problem in Turnserver , and then sent the article. There are a lot of APPRTC on the Int

Google's Turnserver download method:svn checkout http://rfc5766-turn-server.googlecode.com/svn/branches/v3.2/ Rfc5766-turn-server-read-onlyAbout the application of WEBRTC Google gives an example:https://apprtc.appspot.com/(need FQ, sometimes fq may not be able to land, it is estimated that the use of too many people)I was always curious about the way he used turn, and then finally figured out what was going on. Take a look at the following characters:

H264 code Stream parsing, online has a lot of open source files; The general analysis is: Obtain Nalu,sps,pps,nalu type,slice type, obtain QP and so on; The computation can be obtained by the bitwise operation of C + +, but the structure can be obtained directly. Here is the WEBRTC in the H264 parsing Related: In the WEBRTC, about the H264 related source files in: webrtc58\src\

reproduced in the original: http://www.cnblogs.com/mod109/p/5469799.html thank you very much. The WEBRTC audio processing module is divided into noise reduction ns, Echo cancellation AEC(Echo control Acem), Automatic control gain AGC, Mute detection section. In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve

First, an overview of the WEBRTC audio processing flow, see:WEBRTC The audio session is abstracted into a channel channels, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. There are three channel, each channel contains codec and rtp/rtcp send function.In the case of a channel, the application will contain three active threads, a recording thread, an audio receive thread, and a playback thread.1) Recor

WEBRTC IntroductionWebRTC (Web real-time Communications) is a protocol that allows us to implement peer-to-peer on the browser. We can use this protocol to transfer text, voice, video and file content. This article has recorded some personal understanding of my learning process. It is highly recommended to read the documentation for MDN for systematic learning.Simple processFirst, we have a bit a and point B want to communicate with each other. At the

WEBRTC's echo Cancellation algorithm (AEC,AECM) has several important modules:1. Echo Delay estimation2.NLMS3.NLP4.CNG5. Double-ended detection (DT)The following are respectively described:(1) Echo delay estimationecho Delay Length: Based on correlated time delay estimation algorithm (including: Based on the speech signal autocorrelation pitch period): Echo cancellation site, time delay search range is large.WEBRTC's echo delay estimation, which is based on the algorithm of Gips chief scientist

The most recent time you've been doing WEBRTC-based Android apps has encountered some problems releasing resources, which are now recorded for memos.The official Apprtcdemo is too simplistic and many questions are not involved.1. Releasing the peerconnection resource problem.Scenario: A and B make a call (Video call)Now stop the call in B.Error: After B terminates the call, the terminal a program will exit unexpectedly.Analysis: When A and b make a ca

This article is mainly their own previous research WEBRTC code structure when some information (including ANDROID,IOS,PC), the article from the blog Garden Rtc.blacker, reproduced please explain the source.1, WEBRTC module: Audio data acquisition, sending, receiving, playback call process:2, WEBRTC module: Video data acquisition, sending, receiving, playback call

. 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 browsers.var datachanneloptions = { Ordered:false,

We have introduced WebRTC and read the example of https://apprtc.appspot.com/on the Internet (which may need to be accessed through a wall). This example is an application deployed on the Google App Engine and relies on the Gae environment, the background language is Python and also relies on the Google App Engine channel API. Therefore, it cannot be run locally or be expanded. After studying the Python source code in the example, I decided to impleme