webrtc server

WEBRTC IntroductionWebRTC (Web real-time Communications) is a protocol that allows us to implement peer-to-peer on the browser. We can use this protocol to transfer text, voice, video and file content. This article has recorded some personal understanding of my learning process. It is highly recommended to read the documentation for MDN for systematic learning.Simple processFirst, we have a bit a and point B want to communicate with each other. At the

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC.CreatepeerconnectionfactoryIn Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows:inline rtc::scoped_refptrAs you can see, the last four parameters of Createpeercon

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC. createpeerconnectionfactory In Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows: Inline rtc::scoped_refptr As you

Some personal understanding about WEBRTC Just participated in the sound network presided over the first WEBRTC conference in Beijing, coupled with reading "hundred asked Freeswtich" written by Daniel, to it has more understanding, record for later review: 1, simple understanding, WEBRTC is a way to achieve web-to-business dial audio and video telephony technolog

Author: Gustavo Garcia (original link) Translation: Liu Tong Bandwidth estimation is probably the most important part of the WEBRTC video engine. The task of the Bandwidth estimation (BWE) module is to determine how much video stream you can send and not network congestion to ensure that video quality is not reduced. In the previous bandwidth estimation algorithm is very basic, in general, based on the design of packet loss. Usually we start to slowl

SchemaDescription. CAddWhatever isNecessary toEnableStart-up Daemon forthe/usr/Local/bin/turnserver.2)IfYou Do notWant the Turnserver toBe a system service, ThenYou canStart/stop it"Manually",usingThe"Turnserver"Executable withAppropriate options (see the documentation).3) to Create Database Schema, useSchema inchfile/usr/Local/share/turnserver/Schema.SQL.4) forAdditional information, run: $ Man Turnserver $ mans turnadmin $ man turnutilsCreate a user.db file in the root directoryStart with Tu

Google's Turnserver download method:svn checkout http://rfc5766-turn-server.googlecode.com/svn/branches/v3.2/ Rfc5766-turn-server-read-onlyAbout the application of WEBRTC Google gives an example:https://apprtc.appspot.com/(need FQ, sometimes fq may not be able to land, it is estimated that the use of too many people)I was always curious about the way he used turn, and then finally figured out what was going

servers, signaling servers, penetrating servers1, APPRTC Room server HTTPS://GITHUB.COM/WEBRTC/APPRTC2, Collider signaling server above the source code in the own3. Coturn Penetrating server Https://github.com/coturn/coturn4, need to implement their own Coturn connection information interface, the main return user nam

src\webrtc\examples\peerconnection\client\conductor.ccboolconductor::initializepeerconnection()1 webrtc::createpeerconnectionfactory ();src\talk\app\webrtc\peerconnectionfactory.cc1.1 New Rtc::refcountedobject1.2 bool Peerconnectionfactory::initialize ()1.2.1 cricket::mediaengineinterface* media_engine =Peerconnectionfactory::createmediaengine_w()src\talk\media\

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WEBRTC standard customizes how real-time media is transmitted to and from the web, but there is no specification for multiplayer video and is also a challenge for WEBRTC in enterprise solutions, WEBRTC technical VideoThe conference programme can be summed up in several1. Mesh S

Cross-platform WEBRTC WEBRTC is Google Open source of a plug-in real-time video communication technology, which is divided into web development and native development; currently supports Chrome,firefox,android,ios,opera,edge. is a true sense of cross-platform plug-in real-time video communication technology. Video applications are generally based on web-level development. This paper is mainly about the cod

In "Let WEBRTC support H264 codec" I provide a priority to use the H264 codec thinking. We can verify it on the browser side. There are three ways to verify: In JS print SDP view Chrome's log chrome_debug.log (see Open Chrome Log) Grab bag using webrtc-internals The first three kinds are no longer introduced, we look at the webrtc-internals. The

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

WEBRTC Technology Group: 234795279Original Address: http://blog.csdn.net/temotemo/article/details/7530504 1, WebRTC purpose WebRTC (Web real-time communication) The ultimate purpose of the project The main is to allow web developers to be based on the browser (chrome\firefox\ ... Fast and easy to develop rich real-time multimedia applications, without the need to

We have introduced WebRTC and read the example of https://apprtc.appspot.com/on the Internet (which may need to be accessed through a wall). This example is an application deployed on the Google App Engine and relies on the Gae environment, the background language is Python and also relies on the Google App Engine channel API. Therefore, it cannot be run locally or be expanded. After studying the Python source code in the example, I decided to impleme

As early as 2014 through the WebRTC realized the PC client real-time video voice, then the establishment of peer-to WEBRTC with the Libjingle library, using the Peerconnection API implementation. Later in the Remote Desktop, file transfer requires point-to-point connection, the Libjingle library for a period of time, found a few problems:The 1.libjingle library is built using the XMPP protocol, but our clie

WEBRTC, also known as "Web Instant Messaging", is a set of API functions that have been certified by the organization to support voice calls, video chats, and peer sharing files between browsers.This protocol mainly includes: Getusermedia,rtcpeerconnection,rtcdatachannels,getstats these modules. The function of Getusermediagon is to allow the browser to use other devices such as a webcam, a mobile phone, and rtcpeerconnection to create a connection be