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WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call

Transferred from: http://blog.csdn.net/nonmarking/article/details/47375849 This series is currently a total of three articles, follow up will also update WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call WebRTC Videoengine Ultra-Detailed tutorial (ii)--integrated OPENH264 codec WebRTC Vide

WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call

General Statement In the previous article, we explained how to integrate the OPENH264 codec into the WEBRTC, but OPENH264 can only encode baseline H264 video, and in terms of encoding quality, X264 is the best, This article will explain how to integrate the X264 encoder into the WEBRTC, in order to achieve decoding, at the same time to use the ffmpeg. The overall process, as before, is divided into the re-

Apple finally got in. WebRTC, what other pits are on the way to the next generation of mobile WEB applications? (6/11)

within the browser. But at the same time, there are many developers questioned whether WEBRTC will be short-lived, after all, although the Chrome, Firefox, Opera and other mainstream browsers have joined the WEBRTC stronghold, but Apple's Safari and Microsoft's IE do not support. Today, Safari is about to integrate WEBRTC in the latest version, what it means for

Android IOS WebRTC Audio Video Development Summary (86)--analysis of implementation of RTP/RTCP protocol in WebRTC

This paper mainly introduces the RTP/RTCP protocol in WEBRTC, Weizhenwei, the earliest published articles in the Wind network, ID:BEFOIOSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).a prefaceThe RTP/RTCP protocol is the cornerstone of streaming media communications. The RTP protocol defines the packet format for streaming media data over the Internet, while the RTCP pr

The first knowledge WEBRTC of WEBRTC's introductory study

Introduction:First declare I was just a small intern, if there is not correct, I hope you help correct me.First, WEBRTC basic structureFigure A WEBRTC overall structure, from Baidu EncyclopediaFirst of all, WEBRTC the general realization of the idea: we create a web app, and then call in the app's JS Api,js API will in

Comparison of advantages and disadvantages of WEBRTC native development and hybrid development

will have some compatibility issues 。 All your users and older mobile phones, you can only use the native development. native Development, compiled WEBRTC development package size of about 3M, and the use of mixed development, the app finally packaged out the size is above 30M, so if your application degree if the program installed package size requirements, you can only use the native development. Finally, if you should also add H265 such as video c

Android IOS WebRTC Audio and Video Development Summary (87)--analysis of the implementation of packet loss retransmission Nack in WebRTC

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

Cross-platform WEBRTC client framework: OPENWEBRTC

due to a lot of features in the WEBRTC, involving platform-related hardware device interaction, media flow, etc. need to be shared between the tabs, making the changes mainly for the rendering layer. Hopefully these changes will allow us to see the effects in Safari or iOS WebView as soon as possible. OPENWEBRTC (1) Server and Android client Demo installation 1, about OPENWEBRTC OPENWEBRTC is an open-source, cross-platform,

WebRTC Past Life

. When a network handler is available, the handle runs. Alice sends the serialized data to Bob, either way: WebSocket or otherwise. When Bob gets the data from Alice, call Addicecandidate to add the remote node description. WebRTC clients (known as peers, such as Alice and BOB) also need to exchange local and remote audio and video media information, such as protocols and encoders used. TOD

Introduction to "WebRTC"

description can be seen here. Network stream Apimediastream:mediastream is used to represent a stream of media data. Mediastreamtrack represents a media source in the browser. Rtcpeerconnectionrtcpeerconnection: A Rtcpeerconnection object allows users to communicate directly between two browsers. Rtcicecandidate: Represents a candidate for an ice protocol. Rtciceserver: Represents an ice Server. Peer-to-peer data apidatachannel: The Datachannel interface represents a bidirectional data channel

WebRTC Advanced-Overview-one: WebRTC The comparison and analysis of networking technology of conference

server, it can handle real-time data very well. However, the cost of the MCU is high and it increases the overall cost of the network. Comparison of mesh networks and star-like networksThis paper compares the network and CPU load of mesh network and star-shaped network to the growth of users. 2.1 test MethodsA mesh network can be composed of two computers connected directly to each other. I found out that https://appear.in/had a way of constructing a mesh network, In it there is a group

"Getting Started with WebRTC" The first chapter WebRTC introduction

"Getting Started with WebRTC" The first chapter WebRTC introduction?This chapter is a conceptual introduction to WEBRTC.after reading this chapter. You will have a clear understanding of the following: . What is WEBRTC . How to use it . which browsers support1.1. WEBRTC IntroductionWorld Wide Web (WWW) is the early day

WEBRTC iOS Framework Compilation

Selection of iOS framework for 1.WebRTCCurrently two more active open source WEBRTC implementations. Google WebRTC: Project address is: https://code.google.com/p/webrtc/ Ericsson OPENWEBRTC: Project address is: HTTPS://GITHUB.COM/ERICSSONRESEARCH/OPENWEBRTCOur Camp David Education is designed to build the iOS app development framework

WebRTC Project Research

??WebRTCdirectory (?) [+]1. Official website:Http://sites.google.com.sixxs.org/site/webrtc/homeHttp://sites.google.com.sixxs.org/site/webrtc/reference/webrtc-componentsHttp://sites.google.com.sixxs.org/site/webrtc/referenceHttp://code.google.com.sixxs.org/p/webrtc/source/che

Introduction of Android WEBRTC

method.AddicecandidateIcecandidates is created when an internal iceframework discovers that candidates allows other parties to connect to you. When passing data to each other through peerconnectionobserver.onicecandidate, you need to get to each other's icecandidates through any signal channel you choose. Use Addicecandidate to add them to peerconnection so that peerconnection can attempt to connect to each other through existing information.Createoffer/createanswerThese two methods are used fo

Uncover the mystery of WEBRTC Media server--WEBRTC Media Server & Open Source project Introduction

Uncover the mystery of WEBRTC Media server--WEBRTC Media Server Open Source project IntroductionThe WEBRTC ecosystem is very large. When I first tried to understand WEBRTC, the number of network resources was unbelievable. This article provides some introduction to WEBRTC m

WEBRTC Introduction and simple Application

WEBRTC Introduction and simple Application WebRTC, web Real-time communication, Web real-time communication technology. In short, the introduction of real-time communication in a Web browser, including audio and video calls. WEBRTC Real-time communication technology Introduction How to use Media Introduction Signaling Stun

Using WEBRTC to build a front-end video chat room-introductory article

the device's camera and microphone* Rtcpeerconnection:rtcpeerconnection is a component that WEBRTC uses to build stable, efficient streaming between point-to-point* Rtcdatachannel:rtcdatachannel enables a high-throughput, low-latency channel between browsers (Point-to-point) for transmitting arbitrary data Here's a general introduction to these three APIs MediaStream (Getusermedia)The MediaStream API provides WEB

WEBRTC interface invocation process for video calls based on WEBRTC

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

WEBRTC Server Setup

1.WebRTC Backend Service: Room server for callsThe room server is used to create and manage call session status maintenance, is the two sides call or multiparty calls, join and leave the room and so on, we temporarily follow the Google deployment on the Gae platform APPRTC this room server implementation, the Gae The app's source code can be obtained on

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