Said Nat before the penetration of a few about the concept of WEBRTC, may have been the same as the author of the WEBRTC concept of the wrong understanding of the classmate. WebRTC (Network real-time communication) It is a Web browser to support real-time voice dialogue or video dialogue technology, it provides us with video conferencing core technology, includin
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-time streaming media application quality assura
Real-time video communication via WebRTC (I.)
Real-time video communication via WEBRTC (II.)
Real-time video communication via WEBRTC (iii)
In this article we continue to learn about WebRTC 's related Api,rtcpeerconnectiont and Rtcdatachannel.RtcpeerconnectionRtcpeerconnection is a
reproduced in the original: http://blog.csdn.net/u012931018/article/details/17045077 thank Bo Master.
WEBRTC Echo Cancellation (Acoustic ECHOCANCELLATION,AEC, acoustic echocancellation for MOBILE,AECM) algorithm mainly includes the following important modules: echo Time delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC
The IP address consists of three types: unicast address (destination is a single host), broadcast address (destination is all hosts on a given network), and multicast address (destination is all hosts within the same group). broadcast and multicast only apply to UDP, It is important that they transmit messages to multiple receivers simultaneously. TCP is a connection-oriented protocol that means that ther
Digital video transmission is a communication method that people use vision to obtain information. Compared with other information transmission methods, digital video transmission is accurate, intuitive, and efficient. Due to the large amount of information transmitted by digital videos and limited bandwidth, the compression and encoding of videos, selection of transmission channels and network protocols, and IP multicast technology (IP
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), ILBC (audio), pcma,pcmu,g722,g729). NAT supports Ice (Stun+turn)2) Effect measurementTest en
Recently experiment how to let WEBRTC support H264 code, record, for people who need reference.
To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code.
The end result is that the browser can send a video with H264 or receive H264 video.
Note t
Learning Method:WhyWhatHowWhy does this technology come into being. Without this technology, what is the current situation?What is this technology? Its principle, operating process, its working process, its functions, and so onHow to Implement --- basic configuration, optimized configuration, advanced configuration, security features and summary.Advantages of Multicast:One-to-multiple, send data to multiple receivers at the same time.Bandwidth savingSaves hardware resources and so on.Vro with IP
IPV4 Multicast (multicast) mechanism refers to: the mechanism of sending and receiving IP multicast traffic. IP multicast traffic is sent to a single destination IP address, but is received and processed by multiple IP hosts, regardless of where the host is located on the IP internetwork. A host listens on a specific I
To accomplish useful work, peers in peer-to-peer applications must be able to discover each other and interact with each other. This article provides an implementation of a discovery based on IP multicast.
Before a software entity can participate in a direct peer-to-peer interaction with a Peer-to-peer application feature, the entity must discover the appropriate peers to interact with. All available Peer-to-peer architectures provide a solution for
WEBRTC reply content: I am in development and have a basic understanding of the WebRTC source code stack. It mainly consists of two key technologies: 1. webRTC Video/Voice Engine, including camera microphone operations, Video preprocessing, VP8 coding/decoding, and streaming media transmission (RTP/RTCP); 2. implement the P2P channel and use libjingle to complete
Original address: http://segmentfault.com/a/1190000000436544 what is WEBRTC?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
Recently experiment how to let WEBRTC support H264 code, record, for people who need reference.To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code.The end result is that the browser can send a video with H264 or receive H264 video.Note that
UDP multicast can be used to broadcast data in the intarnet or Internet datagram mode (multicast over the Internet requires the router to support IGMP (Internet gateway management protocol, this Protocol appears after the IP address appears to support multicast )). compared with broadcasts that consume extremely bandwidth (broadcast can only be broadcast within t
UDP, broadcast and Multicast (7), udp Broadcast
(Reference) TCP/IP explanation, Volume 1: Protocol
UDP is a simple datagram-oriented transport layer protocol. Each output operation of a process generates a UDP datagram and assembles it into an IP datagram to be sent. UDp datagram is encapsulated into IP datagram format, for example:
UDP Header
The fields of the UDP header are as follows:
The port number indicates the sending and receiving processes.
ObjectiveBefore writing the article "separate compilation using WEBRTC audio processing module-Android", I have been trying to compile the WEBRTC audio processing engine Voe the whole to use for his project, but limited to the poor technology, time is tight, so did not succeed. The AECM, AGC, NS, and VAD modules in the engine were compiled separately to make it work. Although it can achieve a certain effect
In the information age, network technology has been widely used. Many applications depend on the ability to send the same information from one host to multiple hosts or from multiple hosts to multiple hosts. The number of applications distributed on the Internet can reach 100,000, these require higher bandwidth and greatly exceed the unicast capability. IP multicast is an important technology that can maximize the use of existing bandwidth.1. Concepts
Recently, due to the needs of the project, I began to touch the WEBRTC thing. Unexpectedly the threshold of this thing is still pretty high, next share I stepped on the pit, hoping for the first contact with this thing in the future to help people.WEBRTC official websiteThe first step of course is to see the official homepage (www.webrtc.org), first the content of the homepage was roughly swept over, probably a little bit of understanding of this thin
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