window rtcpeerconnection

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How to obtain an IP address via JS

Through the JS to get your real external network IP and intranet IP, even if the agent is not used, think it is too scary, but also can be happy to install force!Code:Get the IP addresses associated with a accountfunction Getips (callback) {var ip_dups = {};Compatibility for Firefox and Chromevar rtcpeerconnection = window. Rtcpeerconnection|| Window.mozrtcpeerco

Android IOS WebRTC Audio Video Development Summary (vii)

}}};//Constraints object for standard resolution Videovar vgaconstraints = {video: {mandatory: {m axwidth:640, maxheight:480}}};//Constraints object for high resolution Videovar hdconstraints = {video: {mans Datory: {minwidth:1280, minheight:960}}};function getmedia (constraints) {if (!! Stream) {video.src = null; Stream.stop ();} Navigator.getusermedia (Constraints, Successcallback, errorcallback);}3. Use the Rtcpeerconnection object to exchange med

WEBRTC Introductory article

get a navigatorusermediaerror error, display Permissiondeniederror, the simplest way is the CD to the directory where the HTML file is located, and then Python-m Simplehttpserver (loaded with Python), and then enter the http://localhost:8000/{file name in the browser}.html After using Getusermedia to get the stream, you need to output it, typically binding it to the output on the video label, and you need to use window. Url.createobjecturl (Localmedi

Using WEBRTC to build a front-end video chat room-introductory article

is like thisvar getUserMedia = (navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia);A super-simple exampleHere is a super simple example to show the effect of Getusermedia:Htmllang="ZH-CN" >Head>Metacharset="UTF-8" >Title>getusermedia instancesTitle>Head>Body>VideoId="Video"Autoplay>Video>Body>ScriptType="Text/javascript" > var Getusermedia = (Navigator.getusermedia | | navig

WebRTC Past Life

, free, standardized, and built-in to the browser, and more efficient than existing technologies.Where are weWebRTC is used in a variety of apps, including WhatsApp, Facebook Manager, appear.in and TokBox platforms. Even the experimental WebRTC on the IOS browser. WebRTC is also used in webkitgtk+ and QT.Microsoft added the MediaCapture and Stream APIs to the Edge.The WebRTC implements three APIs: Meidastream (aka Getusermedia) Rtcpeerconnection

JS Bypass proxy, VPN access to real IP and intranet IP, the implementation of reverse tracking method _javascript Skills

Firefox and Chrome support WEBRTC can request the stun server to return to the intranet IP, unlike the XMLHttpRequest request, stun request is not seen in the developer tool. Get the IP addresses associated with a account function Getips (callback) {var ip_dups = {}; Compatibility for Firefox and chrome var rtcpeerconnection = window. rtcpeerconnection | |

Using WEBRTC to build a front-end video chat room-introductory article

= (Navigator.getusermedia | | Navigator.webkitgetusermedia | | Navigator.mozgetusermedia | | Navigator.msgetusermedia); A super-simple exampleHere is a super simple example to show the effect of Getusermedia: html lang="ZH-CN"> head> meta charset="UTF-8"> title>Getusermedia instancestitle> head> body> video id="video" autoplay>video> body> script type="Text/javascript"> varGetusermedia = (Navigator.getusermedia | | navigator.webkitget

HTML5 new characteristics of the webrtc[turn]

Original: http://www.cnblogs.com/jscode/p/3601648.html?comefrom=http://blogread.cn/news/1. OverviewWEBRTC is the acronym for "Network Real Time Communication", which is used primarily to allow the browser to capture and Exchange video, audio, and data in real time.WEBRTC a total of three APIs. MediaStream (also known as Getusermedia) Rtcpeerconnection Rtcdatachannel Getusermedia is primarily used to obtain video and audio informa

Using WEBRTC to build a front-end video chat room-introductory article

compatible version is like thisvar getUserMedia = (navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia);A super-simple exampleHere is a super simple example to show the effect of Getusermedia:Htmllang="ZH-CN" >Head>Metacharset="UTF-8" >Title>getusermedia instancesTitle>Head>Body>VideoId="Video"Autoplay>Video>Body>ScriptType="Text/javascript" > var Getusermedia = (Navigator.get

Test 178 Intranet through a storage XSS

Test 178 Intranet through a storage XSS Test the 178 Intranet attachment payload through a storage XSS Site: http://apt.178.com/The input is not filtered when an app or ringtone resource is added.As follows: After the upload, it will be displayed on the front-end only after the Administrator reviews it.Then you can manage it blindly.I'm so excited that I have successfully received cookies and other information. However, restrictions are imposed on the background. Okay. In this way, the end is

xss-using WEBRTC to get the intranet IP

Xss.jsfunctiongetips (callback) { varip_dups={}; //compatibilityforfirefoxandchrome var Rtcpeerconnection=window. rtcpeerconnection | | window.mozRTCPeerConnection | | window.webkitRTCPeerConnection; varusewebkit=!! window.webkitrtcpeerconnection;//bypassnaivewebrtcblockingusing aniframe if (! Rtcpeerconnection) { //NO

WEBRTC Introduction and simple Application

multiple browsers, there are two modes to establish such sessions:4. New WEBRTC Features How to use WEBRTCWEBRTC is easy to use and requires minimal steps to establish a media session. Some messages flow between the browser and the server, while others flow directly between the two browsers (which become peers).1. Establish WEBRTC sessionThere are several steps required to establish a WEBRTC connection: Get local media ( getUserMedia() ,mediastream API) Establishing a

"Reprint" WEBRTC Basic introduction

the browser-side WEBRTC Video Call process.WEBRTC structureFirst of all, from the WEB-RTC aspect analysis WEBRTC on the browser interface structure, in the browser side, WEBRTC mainly implemented three interfaces MediaStream, to achieve the encapsulation of local audio and video resources, such as from camera, microphone, remote stream and so on, MediaStream represents a media data stream, a Mediastreamtrack represents a media source in the MediaStream, such as audio, video, subtitles,

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 Webkitrtcpeerconnection,firefox with the name Mozrtcpeerconnection.Rtcdatachannel is only available in Chrome, Opera 18 and Firefox 22

WEBRTC Basic Introduction

the browser-side WEBRTC Video Call process.  WEBRTC structureFirst of all, from the WEB-RTC aspect analysis WEBRTC on the browser interface structure, in the browser side, WEBRTC mainly implemented three interfaces MediaStream, to achieve the encapsulation of local audio and video resources, such as from camera, microphone, remote stream and so on, MediaStream represents a media data stream, a Mediastreamtrack represents a media source in the MediaStream, such as audio, video, subtitle

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 WEBRTC component that addresses the stability and efficiency of data flow communication. The following is an architectural diagram of WebRTC , showing the rules of rt

Using WEBRTC to build front-end video chat room--Data channel Chapter

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, which can transfer data directly betw

WEBRTC Study Summary

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 same time, we need an intermediary s to tran

Real-time video communication via WebRTC (I.)

display even on another computer. Here is a specific tutorial for this application: later in this article.BeginIf you don't have time to read this series of articles and want to encode directly, you can do this: Take a look at Gooogle's slide show about WebRTC (here) You have not used Getusermedia, you have to learn it first, tutorial: HTML5 Rocks article, Demo:simpl.info/gum. Master rtcpeerconnection API, tutorial: simple examp

Real-time video communication via WebRTC (I.)

display even on another computer. Here is a specific tutorial for this application: later in this article.BeginIf you don't have time to read this series of articles and want to encode directly, you can do this: Take a look at Gooogle's slide show about WebRTC (here) You have not used Getusermedia, you have to learn it first, tutorial: HTML5 Rocks article, Demo:simpl.info/gum. Master rtcpeerconnection API, tutorial: simple examp

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