webrtc multicast

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WEBRTC native app optimized for low bandwidth

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.When using WEBRTC Base library to develop Android,ios native application, because the mobile terminal is not like PC side, in bandwidth stability, system performance is very big difference, so for mobile device WEBRTC need to do some optimization to improve the call effect,For example, WE

The simplest example of WEBRTC

The online example of WEBRTC is mostly code, the following is an example of a WEBRTC-to-one sample of code simplification, which is tested under Chrome 37. Where iceserver can be omitted, there is no iceserver when the same LAN can still communicate.Client code:When WEBRTC is implemented, the signaling server is required to facilitate communication between client

WEBRTC Start-APPRTC Server Setup

Introduction APPRTC is what, webrtc.org official Experience App Ingredients: ubuntu14.04, other Linux versions are not limited, the official does not specify Chrome M51+stunnle Https://www.stunnel.org/ind Ex.htmlrfc5766-turn-server https://code.google.com/archive/p/rfc5766-turn-server/Google App Engine SDK for Pythonapp RTC HTTPS://GITHUB.COM/WEBRTC/APPRTCsteps:Set up proxyBecause of the particularity of the domestic network, this step is very importa

Sketchy WEBRTC ns (Noise suppression) module

Noise suppression, is what everyone said noise reduction. This noise reduction is a distinction between vocals and non-human voices, which is a noise.A piece of audio that contains vocals and noise is processed by the module, and in theory, only the vocal is left.WEBRTC NS In the industry is still famous, through the actual comparison test, we found that webrtc noise reduction is indeed performance and stabilityare higher than similar open source algo

Sketchy WEBRTC audio processing algorithm written in front of the words

Recently work used to WEBRTC, found that WEBRTC is a treasure trove, there are many things worth studying.Search this area a lot of information, found that the introduction of the use of WEBRTC, but for some of the algorithm researchNot much. In particular, the algorithm can be said to be concise and clear is even rarer.In fact, I would like to carefully study ea

Small knowledge of WEBRTC

First, WEBRTC related APIReference: Https://github.com/ChenYilong/WebRTC/blob/master/WebRTC Getting Started tutorial/webrtc Getting Started tutorial. MD1.1 Functional Divisions Get audio and video data Transmitting audio and video data Transfer arbitrary binary data 1.2 API partition: Three JS int

Unicast, broadcast, and Multicast

Unicast, broadcast, and Multicast I. Introduction 1. Three IP addresses are available: unicast address, broadcast address, and multicast address. 2. broadcast and multicast are only applied to UDP. TCP is a connection-oriented protocol that means two processes (determined by the port number) run on two hosts (determined by the IP address) there is a connection b

IPv6 multicast in Network Interconnection (1)

In the network overuse phase, we will use the dual-stack technology to connect the two networks. In the double stack technology, IPv6 multicast provides powerful functions. It can run on both the router and host, and can write v4 and v6 to form a convergence point. In transition technology, IPv6 multicast is very important. We will discuss this in depth. IPv6 multicast

Multicast unicast broadcast

http://blog.csdn.net/bloghome/article/details/4682984 Multicast Overview: Multimedia applications integrate sound, graphics, animation, text, and video, and this application is more and more in the current network environment. Multimedia TrafficThere are three main modes of communication in the network: 1, Unicast (unicast) 2, broadcast (broadcast) 3, multicast (multica

WEBRTC compilation Details

WEBRTC Compilation Details--record + reprintOriginal address: http://blog.csdn.net/temotemo/article/details/7056581WEBRTC compilingMy environment:Operating system: XP SP3VS 2013Tools required before compiling the source codeGet the source code tool:1, first need to install the source of the tool SVN (Project code version management tools, Google also use this)TortoiseSVN 1.6.12http://sourceforge.net/projects/tortoisesvn/2. Download and install Msysgit

WEBRTC demo in the browser

WEBRTC in the Chrome browser demo Many examples, WebRTC source, but in the Firefox browser, the example can not be used, the information on the web said to set the media.peerconnection.enabled to True, However, in the Firefox browser, the default value is True, using the WEBRTC example in Firefox or can not capture local video. Here are the examples found online:

Compile and install WebRTC in Ubuntu

Compile and install WebRTCsvncheckouthttp in Ubuntu: // configure. Then gclientconfighttps: // webrtc. Compile and install WebRTC in Ubuntu Svn checkout http://webrtc.googlecode.com/svn/trunk/ After the download is complete, WebRTC will get a folder named trunk by default, which contains the WebRTC source code, which i

Google Announces WebRTC Technology Roadmap

Google's first integration of WebRTC in the Chrome Dev release released this January was a source of widespread concern. Today, Google published a roadmap for the development of WebRTC technology in its blog.WebRTC is a technology for real-time video and audio communication inside the browser, and Google acquired a technology in 2010 to acquire Global IP Solutions. The technology is based on the WHATWG prot

Session Border Controler (SBC) based on WEBRTC technology

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 intention of

Use delegation in C # events and Multicast

();}}} (3) about delegation and Multicast Multicast delegates include references to two or more methods. For multicast, use + = to add a delegate and use-= to remove the delegate. The method included in the multicast delegate must return the void; otherwise, a run-time exception is thrown. Because the method does not

UNP summary Chapter 18 ~ 21 route socket, key management socket, broadcast, Multicast

= recvfrom(sockfd, recvline, MAXLINE, 0, preply_addr, len); if (n Iv. Multicasting 1. multicast address 1). IPv4 multicast address, IPv6 multicast address, and IPv6 multicast address Class D addresses in IPv4 (from 224.0.0.0 to 239.255.255.255) are multicast

Introduction to multicast programming under Linux

Here's how to start our multicast programming: First, the concept of multicast Multicast, also known as "multicast", the network of the same business type of host logically grouped, the data sent and received only in the same group, other hosts did not join this group can not send and receive the corresponding data.

WEBRTC Echo Cancellation (AEC, AECM) algorithm introduction (RPM)

WEBRTC Echo Cancellation (AEC, AECM) algorithm introduction NBSP;WEBRTC echo Cancellation (AEC, AECM) algorithm mainly includes the following important modules: 1. Echo Delay Estimation 2.NLMS ( Normalized minimum mean square adaptive Algorithm) 3.NLP (nonlinear filtering) 4.CNG (Comfort Noise generation), the general classic AEC algorithm should also include double-ended detection (DT). Considering that

Brief analysis of WEBRTC echo cancellation module

Welcome to Join WEBRTC Learning Group (659922087) to obtain free learning resources, mutual communication and growth. WEBRTC of the Echo Cancellation (AEC, AECM) algorithms mainly include the following important modules: Echo delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC algorithm should also inc

WEBRTC Audio-related neteq (i)

The previous article (latency) and the reduction method at the end of the speech communication, said that from the beginning of this article will cut into the WebRTC Neteq theme, Neteq is one of the two core technologies of audio technology in WEBRTC (another core technology is the front and back processing of audio, including AEC, ANS, AGC etc, commonly known as 3 a algorithm).

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