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
WEBRTC rtp/rtcp Protocol family2017-02-22 20:15 Reading (144) Comments (0) WebRTC congestion control based on GCC (bottom)2017-02-22 15:44 Reading (108) Comments (0) WebRTC congestion control based on GCC (upper)2017-02-22 11:37 Review (0) WebRTC video receive buffer based on Kalmanfilter delay model2017-02-22 11:2
A recent study on how WebRTC speech runs on iOS found that the voice_engine of WebRTC has implemented iOS-related classes, but encountered a series of problems in specific applications. After several days of hard work, finally, we solved a series of problems and successfully realized recording and playing local loop in the simulator.
Compile the testProgramIn the process, we plan to use the libjingle Libr
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 (recommend
??WebRTCdirectory (?) [+]1. Official website:Http://sites.google.com.sixxs.org/site/webrtc/homeHttp://sites.google.com.sixxs.org/site/webrtc/reference/webrtc-componentsHttp://sites.google.com.sixxs.org/site/webrtc/referenceHttp://code.google.com.sixxs.org/p/webrtc/source/che
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 the domestic network, this step is very importa
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
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
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;
Canceller for Mobile acoustic echo cancellation), and AGC (auto Gain control automatic gain controls) and other modules to compile and use the method.
After their own two months of churn and testing, finally on Android and IOS successfully compiled modules and in the project using the NS/VAD/AECM three modules, the effect is relatively good.
Looking back, the compilation of these modules is actually very simple, but two months ago, I really took some effort.
Body
Because several large modules
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
reproduced in the original: Http://www.cnblogs.com/mod109/p/5767867.html#top thank you very much.
The WEBRTC's audio processing module is divided into noise reduction ns (NSX), echo cancellation AEC (Echo control Acem), Audio gain AGC, and 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 other more complex modules, it is best
servers, signaling servers, penetrating servers1, APPRTC Room server HTTPS://GITHUB.COM/WEBRTC/APPRTC2, Collider signaling server above the source code in the own3. Coturn Penetrating server Https://github.com/coturn/coturn4, need to implement their own Coturn connection information interface, the main return user name, password and turn configuration information, usually called Turn REST API, do not implement this interface Apprtcdemo not connected
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
Continue with the unfinished part of the previous article, including the following three sections:1, extension: WEBRTC multiparty calls.2,mcu Multipoint Control Unit.2, Extension: VOIP, telephone, message communication.Note: Translation is not verbatim, but in accordance with their own understanding of the translation, at the same time, in order to facilitate understanding, but also to join some of their own organization language.Reprint please indica
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/
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 ar
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