jitsi webrtc

The HTML5 SIP client is an open-source client that fully utilizes JavaScript to integrate social networking (Facebook, Twitter, Google +), online games, and e-commerce applications. No extensions, no plug-ins, or necessary gateways. The video stack technology relies on WebRTC. Like Demo Video demo on the home page, you can easily implement real-time video/audio calls between Chrome and IOS/Android mobile devices. This client is a technology that can

Chromium is the core of the Google Chrome browser, the first from Apple's WebKit development, because webkit differences in development, and Google wants to have greater freedom in the developer, 2013 Google decided to develop its own branch of WebCore, called the Blink engine, and then Google to the BSD Berkeley Licensing Open source, BSD licensing restrictions more lenient, many browsers are based on chromium development, for example, after the omission of 100 words. Google has made further st

Foreword: Recently the specialized course has the work, has chosen the IP tracking and the anti-tracking this question, thus, has this article.First, look at the concept of STUN . STUN (simple traversal of the UDP over Nats,nat) is a network protocol that allows clients located on a NAT (or multiple NAT) to find their own public address, find out what type of NAT they are in and NAT Internet-side port that is bound to a local port. This information is used to establish UDP traffic between two ho

Building an chromium-based applicationChromium is the kernel used by Google Chrome, which started out with Apple's WebKit, because of WebKit's divergence in development and Google's desire for greater freedom in development, 2013 Google decided to develop its own branch of WebCore, called the Blink engine, and then Google to the BSD Berkeley Licensing Open source, BSD licensing restrictions more lenient, many browsers are based on chromium development, for example, after the omission of 100 word

HeadtrackrIs a JS library for Real-Time Face Tracking and head tracking. Track the user's head position on the computer screen through the webcam and webRTC/getUserMedia standards. If you want to watch the example, you can watch this video. Or take a notebook with a camera or a browser supporting webRTC/getUserMedia and try the following example.Check whether your browser supports

Author: seven Cow Cloud Original link: Click to open the link push protocol Here's a look at what the push protocols are, their status and pros and Cons in the live broadcast field. RTMP WebRTC UDP-based private protocol 1. RTMP RTMP is the acronym for Real Time Messaging Protocol (Live Message Transfer Protocol). The protocol is based on TCP and is a protocol family, including various variants such as RTMP Basic protocol and Rtmpt/rtmps/rtmpe. RTMP

through the keyboard to select the desktop icon, you can press on, down, left, right, home,end, select the icon 8, add the silver skin Features to be developed: 1. Task Manager 2. Object Viewer 3, command line terminal November 2012 Unknowingly has been updated for more than 1 years, but also intermittent changes, due to the reasons for the work can not be promoted. Here's the latest progress: Added to the page split-screen processing increased desktop pendants, such as weather, clock app

install GRUNT-CLI 4. Engage in APPRTC source code Sudo-i cd/root git clone https://github.com/webrtc/apprtc.git Backup source Tar czf apprtc.tar.gz apprtc/ Installation dependencies NPM Install Sometimes it is wrong to report permissions, you need to add sudo Compile Grunt Build To see more problems, copy the error log Google can5. Modify the configuration fileconstant.py Modify it as follows, and pay attention to replacing it with your own I

RTP is transferred in the H264 according to the NALU group packageThere's WEBRTC.Single NAL Unit PackageStap-aFu-ahttp://blog.csdn.net/chen495810242/article/details/39207305When packing, remove Startcode, receive data and add Startcode The recent need to modify the WEBRTC processing process needs to change the way StartcodeRecord it: Where the receiving end adds Startcodewebrtc\src\webrtc\modules\video_co

RTP is transmitted at the time of the h264 is in accordance with the NALU group packageThere's WEBRTC.Single NAL Unit PackageStap-aFu-ahttp://blog.csdn.net/chen495810242/article/details/39207305Remove Startcode when packing, receive data and add Startcode Recently need to modify the way the WEBRTC processing process needs to change StartcodeRecord it: Where the receiving side adds Startcodewebrtc\src\webrtc