webrtc free

WebRTC FEC (forward error correcting code) is an important part of its QoS, which is used to recover original packets when network drops, reduce retransmission times, reduce delay and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its principles. redundant Coding To understand the FEC in WEBRTC, you first need to understand the red Packet. The so-called Re

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC clients establish a call through interactive candidate to achieve NAT traversal, if these interactions candidate in the Offer/anwser SDP will lead to through the establishment time is very slow,Because the WEBRTC client needs to communicate with the Stun/turn server to get these candidate, the entire process is block, in order to increase the speed of call establ

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WebRTC-Peer part uses the ice framework, the ICE framework includes the Stun,turn, and one problem with the actual use of WebRTC to develop audio and video applications is that calls are built very slowly because the ice process takes too much time for the client to communicate with the stun server before initiating the c

This article mainly introduces the last encounter of some machines do not see the video problem, the article from the blog Park Rtc.blacker, reproduced please indicate the source.Before doing the video chat app has been running well, a few days ago customer feedback said in Samsung 9100. You can't see your own image on the Android4.0.3.After a search to find the WEBRTC is the bottom of the bug, has been repaired and feedback to the community, the foll

Transferred from: http://www.xuebuyuan.com/1248366.htmlThe bandwidth adaptive algorithm in WEBRTC is divided into two types:1, the originator bandwidth control, the principle is the RTCP in the packet loss statistics to dynamically increase or decrease the bandwidth, in the reduction of bandwidth using the TFRC algorithm to increase the smoothness.2, the receiver bandwidth estimation, the principle is and by the receipt of RTP data, the estimated band

based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each client needs to establish peerconnection with other N o

Recently the major live sites are compared to fire, want to explore how to play. But read a few Daniel's answer, feel there are too many unfamiliar things, try to get up a little higher cost. Found that there is a thing called WEBRTC, someone has analyzed he is not suitable for the flow of large numbers of live. But I'm just playing with it and feeling the video connectivity.The first thing I saw on GitHub was a demo of all the APIs, and an example of

Recently in the docking WebRTC to Android phone, there is a demand is the mobile phone horizontal screen when the other side of the image rotation, study the code of WEBRTC Video_render found that the remote video rendering using OPENGLES20 or Surfaceview implementation, Where OPENGLES20 uses hardware rendering, so performance is better, so simply add the VideoRenderOpenGles20 class in the video_render_open

1. Introduction It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So a message between A and b through

WEBRTC FEC (forward error-correcting code) is an important part of its QoS, which can be used to recover original data packets when packet loss is lost, reduce retransmission times, reduce latency and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its rationale. Redundant coding To understand the FEC in WEBRTC, you need to first understand red Packet. The

Quality Scaler is WEBRTC in accordance with the video quality, adaptive adjustment resolution of the scheme, the idea is generally: to observe the video encoding loss frame rate and QP changes, determine whether the capturer in the adaptor to adjust the encoding resolution. Its code is located at:$ (ROOT)/src/webrtc/modules/video_coding/utilityThe implementation is fairly simple,Observe the average QP and d

The main reference in this document is [1], which takes the code from the reference article. But [1] did not upload the complete code. Environment configuration can refer to the previous article [2] The main realization of the video in the local transmission. It took a day. As for WEBRTC video capture, codec, please refer to the online blog, here is not mentioned. Compile using CMake to generate makefile, step cmake. Note the points that follow, an

Recently in-house training for companies, mainly on instant messaging and mobile video calls, including Android and Android,ios with Ios,android and iOS, as well as mobile and PCDo not fight unprepared for the war, so carefully collated a more detailed outline, the following outline for mobile video calls,Although the content of the talk is not directly shared, but this outline for everyone to clear the idea of the video call will be helpful,Of course, if you think this outline is missing or fee

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