webrtc server

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 streaming to other users of t

The advent of WEBRTC has made it possible for enterprises to quickly develop a full platform-enabled audio and video program. Before WEBRTC, the enterprise wanted to develop a full-platform audio and video program, the difficulty, the workload is very large. After using WEBRTC, some common modules in audio and video programs such as audio and video capture, play

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:

The goal of this project is to develop an online multiplayer game with as few server resources as possible, while expecting to run the game on one user's browser while allowing another player to connect. In addition, it is hoped that the program is as simple as possible for analysis in blogs.Application of the technologyI found WebRTC when I first came into contact with the technology, and I thought the tec

cause based on the error message. I don't know about google. After the compilation is successful, you can see the corresponding demo installation package in the out/Debug or out/Release directory, such as the Debug directory, where there is a AppRTCDemo-debug.apk and WebRTCDemo-debug.apk The code of the AppRTCDemo program is trunk/talk/examples/android /. Note: What is the difference between AppRTCDemo and WebRTCDemo? WebRTCDemo is only a point-to-point in the LAN. If you know the ip address

network customers, for example, some overseas customers directly connected to the domestic server effect is not good enough, can consider through the TURN service to transit, thereby guaranteeing the quality of service;3 The use of UDP will involve network latency, packet loss, so to consider QoS, the main strategy includes:A jitter cache (jitter buffer) is used to eliminate the jitter characteristics of the network packet, and the packet is delivere

Transferred from: http://www.cnblogs.com/fangkm/p/4364553.htmlWEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether usi

WEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether using the

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

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but SBC technology in the media by the server relay, This violates the original in

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

[Transfer to]WEBRTC Learning: Deploying Stun and turn serversHttp://www.cnblogs.com/lingdhox/p/4209659.htmlThe WEBRTC-to-peer penetration part is implemented by Libjingle.The sequence of steps is probably this:1. Try direct Connect.2. Penetrate through the stun server3. Cannot penetrate through the turn server relay.Stun serv