webrtc streaming

between a browser and the WebSocket server, but through a series of signaling, creating a channel between the browser and the browser (Peer-to-peer), which can send any data, Without having to go through the server. and WEBRTC through the implementation of MediaStream, through the browser to invoke the device's camera, microphone, so that the browser can transfer audio and video. Three interfacesThe WEBRTC

The greatest feature of real-time streaming media applications is real-time, while latency is the biggest enemy of real-time sex. The processing speed of media data is the important reason of delay, and the network congestion is the main cause of delay from the point of transmission. Network congestion can cause packet loss, and may result in longer data transfer times and increased latency.Congestion control is one of the important methods in real-ti

This paper focuses on the WEBRTC-based direct-to-peer streaming technology (Shi, Pro Gajun CTO, Editor: Dora), first published in " here "Support the original, reprint must indicate the source, welcome attention to the public number blacker (Id:blackerteam or WEBRTCORGCN)So far, the live industry continues as expected in full swing development, in the competition after the delay, HD, beauty, seconds open an

reproduced in the original: http://www.cnblogs.com/lidabo/p/6842765.html thank you very much. "WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of Global in 2010 IP Solutions A technology that Google has made available to the company. ”

Scene:1, A call B2, B answer3, A connected with BCommon steps:Both A and B need to initialize the WEBRTC module to create the PeerconnectionfactoryStatus of a in step 11. Create Peerconnection instances through Peerconnectionfactory2. Call Peerconnection.createoffer3, PeerConnection.Observer.onCreateSuccess (final sessiondescription ORIGSDP)4. Send SDP to B5, the following is the collection of Icecandidate, send the mobile phone icecandidate informati

On the WWDC17, Apple has brought us a big surprise-its browser kernel WebKit will formally support WEBRTC, and future Apple browsers based on the WebKit kernel, such as the MacOS high Sierra, Safari in IOS 11 Browser and Safari Technology Preview version 32, will be used to WEBRTC technology. The news has attracted countless webrtc developers, even more so that

"WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of global IP in 2010 Solutions company to get a technology, Google Open source of this technology! ”Just for the more than 60 million dollar technology, it is worth studying!

1,HTTP://WWW.WEBRTC.ORG/WEBRTC official website, god Horse compilation, God horse download, the solution here is the most authoritative.－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－2,HTTPS://CODE.GOOGLE.COM/P/WEBRTC/WEBRTC Source download location, you can also pay attention to the latest changes anywhere.－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－－3,https://webrtchacks.com/is an arti

Recently flv.js things seem to have ignition, and again to the MSE this thing to bring up.MSE (Media source extensions) is a new function of HTML5, and the general function is to realize streaming media function.If the MSE with WEBRTC and JS binary processing, then you can implement the server to send video to one of the browser users, the browser users will then transfer video

signaling protocol selection. Google will also provide a JavaScript library to reduce the complexity of using the API.2. Topologies (topology)Multiple independent peerconnections will be supported, and each peerconnections will be able to send and receive multiple independent media sources.3. Ice/stun/turnICE and STUN are standard ways to establish point-to-point connections on the Internet. Chrome's current stack deviates from official standards, and Google says it is trying to solve the probl

and video module respectively, as the lower limit of the future rendering delay. So far, one audio-video synchronization is complete. This operation is performed periodically in the Moduleprocessthread thread. Iv. Summary This paper analyzes in detail the implementation details of WEBRTC internal audio and video synchronization, including the generation of RTP timestamp, the construction, sending and receiving of RTCP SR message, the initialization

Webrtc ios framework compilation and Webrtcios framework Compilation1. WebRTC iOS framework Selection Currently, two active open-source WebRTC implementations are available. Google WebRTC: Project address: https://code.google.com/p/webrtc/ Ericsson Research OpenWebRTC:

NDK develops live streaming program for Android-side rtmp After a toss-up, the success of the rtmp live streaming code, through the NDK cross-compiled way, ported to Android, thus realizing the Android side acquisition camera and mic seam data, then the H264 video encoding and AAC audio encoding, and sent to the RTMP server, To enable live Android camera. The program, called Ndkrtmpencoder, introdu

the lower bound value of future rendering delay. So far, one audio and video sync operation is complete. This operation is performed periodically in the Moduleprocessthread thread. Summing up the detailed analysis of the implementation of WEBRTC internal audio and video synchronization, including the production of RTP timestamp, RTCP SR message Construction, send and receive, audio and video synchronization initialization and synchronization process.

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is an API that enables Web browsers to make real-time voice

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is an API that enables Web browsers to make real-time voice

video module respectively, as the lower limit of the future rendering delay. So far, one audio-video synchronization is complete. This operation is performed periodically in the Moduleprocessthread thread. Iv. Summary This paper analyzes in detail the implementation details of WEBRTC internal audio and video synchronization, including the generation of RTP timestamp, the construction, sending and receiving of RTCP SR message, the initialization and

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

WEBRTC IOS Framework compilation http://www.th7.cn/Program/IOS/201502/390418.shtml WebRTC in webkit:http://www.webrtcinwebkit.org/ OPENWEBRTC is designed for flexibility and modularity. The bulk of the API layer is implemented in JavaScript, making it super fast to modify and extend with new functionality. Below is a simplified sketch of the architecture. OPENWEBRTC is an open-source, cross-platform,

Transferred from: http://www.cnblogs.com/lingyunhu/p/4232348.htmlToss for one weeks to finally set up the environment of Kurento (development stage of the product, some bugs to solve their own), so write a separate document to introduce.The following starts to introduce Kurento, the article from the blog Garden Rtc.blacker, reproduced please explain the source.What is Kurento?Video conferencing involves a one-to-many, many-to-many, broadcast, transcoding, mixing, co-screen, recording, which requ