webrtc framework

this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds

first, the network topology structureWEBRTC can also be used as multiparty calls, such as video conferencing, in addition to peer-to-peer communication. When it comes to multi-party calls, we need to select a schema for our application. This is a very important decision, because how to organize users is related to the scale of the conference system. Corresponding to WEBRTC, there are two common network topologies: Mesh networks and star-shaped netwo

This article mainly introduces WEBRTC (we translate and collation, translator: Weizhenwei, check: Blacker), the earliest published in the "Weaving wind net"Support original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).Technically speaking, using a webcam for online broadcasting does not require WEBRTC. The camera itself is a server that can

the server.Rtcdatachannel is a completely different approach:* It can establish point-to-point interconnection through the rtcpeerconnection API. Because there is no need for a mediation server, the median "hop count" is reduced and the latency is lower.* Rtcdatachannel uses the stream Control transmission Protocol (SCTP) protocol, allowing us to configure delivery semantics: We can configure the order of packet transfers and provide some configuration for retransmission.SCTP-based Rtcdatachann

stackReal Time ProtocolB. Stun/iceCall connections between different types of networks can be established through stun and ice components.c. Session ManagementAn abstract session layer that provides session building and management capabilities. This layer protocol is left to the application developer to customize the implementation. (5) VoiceengineThe audio engine is a framework that includes a range of audio multimedia processing, ranging from video

based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), IL

again, even if WEBRTC provides a point-to-point channel between browsers for data transmission, it is necessary for the server to be involved in establishing this channel. WEBRTC requires the server to support four aspects of its functionality:1. User Discovery and communication2. Signaling Transmission3. nat/Firewall traversal4. If the point-to-point communication fails to establish, it can be used as a t

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

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374610.htmlThe first two articles describe the running process of WEBRTC and the use of the framework interface, and then begin to analyze the local audio and video collection process. Due to the large space, video capture and audio capture are divided into two blog posts, where the video capture process is analyzed first. Analysis of the time

depletion. NAT technology protects the security of the intranet address, so this raises the question of how NAT blocks access to the extranet address when I use peer-to-peer connectivity, and we have to use NAT to penetrate. So we have the following ideas:We use a public network IP server, a, B are sent to the public network Ip/port contract, the public network server can be learned that a, B ip/port, and because a, b active to the public network IP server contracted, so the public network serv

Go What is a WebRTC Gateway anyway? (Lorenzo Miniero)https://webrtchacks.com/webrtc-gw/as I mentioned in my ' WebRTC meets Telecom ' article a couple of weeks ago, at Quobis we ' re currently involved In30+ WebRTC field trials/pocs which involve in one or another a telco networ K. In the most cases service providers is

What is WEBRTC. As we all know, the browser itself does not support each other directly to establish channels for communication, are through the server relay. For example, now there are 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 message sent to the server, the server to a message relay, sent to B, and vice versa. In this way a message between A

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. ”

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, 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!

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 technology was right for the project.

This article original from Http://blog.csdn.net/voipmaker reprint annotated source. I built a communication learning Exchange Group, 45211986, Welcome to join. 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, dedicat

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

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