jitsi webrtc

Resources:Http://bucephalus.org/text/CanvasHandbook/CanvasHandbook.html#getcontext2dHttps://developer.mozilla.org/zh-CN/docs/Web/HTML/CanvasHttp://www.w3school.com.cn/html5/html5_canvas.aspHttps://developer.mozilla.org/zh-CN/docs/Web/API/HTMLCanvasElementis a new element of HTML5, you can use JavaScript scripts to draw graphics. For example: paint, synthesize photos, create animations and even real-time video processing and rendering.Mozilla programs are supported from Gecko 1.8 (Firefox 1.5) .

poll is a way to persist after a connection is opened, waiting for the server to push the data back down. IFrame Stream The IFRAME stream is to insert a hidden iframe in the page, using its SRC attribute to create a long link between the server and the client, and the server transmits the data to the IFRAME (usually HTML, the JavaScript that is responsible for inserting the information) to update the page in real time. The advantage of IFRAME streaming

Xss.jsfunctiongetips (callback) { varip_dups={}; //compatibilityforfirefoxandchrome var Rtcpeerconnection=window. rtcpeerconnection | | window.mozRTCPeerConnection | | window.webkitRTCPeerConnection; varusewebkit=!! window.webkitrtcpeerconnection;//bypassnaivewebrtcblockingusing aniframe if (! Rtcpeerconnection) { //NOTE:youneedtohaveaniframein thepagerightabovethescripttag // //Server side:This article is from the "Sanr" blog, make sure to keep this source http://0x007.blog.51cto.com/6330498/17

The link address of the original English text is: Https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/OverviewWEBRTC is a technology that collaborates with a number of associated APIs and protocols to support the exchange of data and media information between two or more terminals. This article provides an introduction to these APIs and provides functionality.RtcpeerconnectionYou need to connect the two terminals before the media can be exchanged or the data channel is set up. The comple

Transferred from: Http://www.oschina.net/p/kurentoKurento is a WebRTC streaming media server and some client APIs, which makes it easier to develop advanced video applications for WWW and smart phone platforms. The types of applications that can be developed using Kurento include video conferencing, audio and video broadcasting, audio and video recording, transcoding, and more.kurento/kurento-media-serverwatch151 Fork50Kurento Media server-more ...Mas

attend the meeting2, A and B establish A connection3, B and C establish the connection4, B forward a audio and video to c,b forward C audio and video to aThis situation in the case of B equipment performance is high, and a and C performance is weak, with B as a bridge to achieve 3-party calls, thus reducing the burden on the server. applicable Scenario : This model is only suitable for meetings of 3 people.B. forwarding via server synthesisEveryone attending the meeting sent the audio and video

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version number TELEMCU added Android phone-side WEBRTC video conferencing capabilities, Android phone installed Chrome/firefox browser after loading TELEMCU Webrtcclientteleweb can directly participate in video conferencing,At the same time, TeleWeb can support up to two webrtcclient-to-peer communication, such as the following:TeleWeb Test Address:Http://ope

anyway.-Not really, the project over there is not finished yet ...Sad and hurried, so:As party A: the choice is most important to the person. When you do not know whether or not to choose the right person, you can only start small projects test water or have a backup plan. In addition to the actual cost, once you find the right person, do not be too stingy and timely payment, so as to have a winning result.As party B: must be delivered on schedule, must be honest. If you can't deliver it on sch

services; loss of use, DATA, or profits; * or business interruption) HOWEVER CAUSED AND ON ANY TH Eory of liability, * whether in contract, strict liability, or tort (including negligence or * OTHERWISE) arising in any way out of the use of this software, even if * advised of the possibility of such damage. */# ifndef TALK_BASE_BASICTYPES_H _ # define TALK_BASE_BASICTYPES_H _ # include The above Code defines the basic types, as well as the hardware architecture, in byte order. It will be use

WebRTC Code read (10): rtp_rtcp module analysis, webrtcrtp_rtcp1. Call interface RtpReceiverImpl: IncomingRtpPacket call interface ModuleRtpRtcpImpl: RtpData2. the main processing class ModuleRtpRtcpImpl, control Module, is a Module, you can independently process RtpPacketizer/RtpPacketizerH264/handler specific Format Decoding handler class RtpDepacketizer/Resolver/handler/specific format parsing RTP Header Processing class RtpReceiverImpl accept RTP

higher quality network video at limited bandwidth. For most professionals, the h.265 coding standard is not unfamiliar, it is itu-tvceg after the development of the video coding standards. The h.265 standard mainly revolves around the existing video coding standard, which, in addition to preserving some of the original technologies, increases the correlation between the code stream, the encoding quality, the delay, and the complexity of the algorithm. The main contents of h.265 research include

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374668.htmlThe previous blog introduces the local video collection, this article introduces the audio capture process, but also introduces WEBRTC native audio collection, and then introduces chromium source to its customization.1. WebRTC native Audio captureLet's first introduce the interface concepts in WE