webrtc client

This article is translated from WEBRTC data channelsIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rtcdatachannel API provided by

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 i

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

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

based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each

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

Compile WebRTC For Android code in Ubuntu 14.04 Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who encounter similar problems while downloading and

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

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

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

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WebRTC-Peer part uses the ice framework, the ICE framework includes the Stun,turn, and one problem with the actual use of WebRTC to develop audio and video applications is that calls are built very slowly because the ice process takes too much time for the client to communicate

#!/bin/bashfunction Build_xcode () {Echo "* * * Building WebRTC for the ia32 IOS simulator";Export gyp_generators= "xcode";Export gyp_defines= "build_with_libjingle=1 build_with_chromium=0 libjingle_objc=1 os=ios target_arch=ia32 clang_ Xcode=1 ";Export gyp_generator_flags= "$GYP _generator_flags output_dir=out_ios_ia32";Export gyp_crosscompile=1;Gclient runhooks;Ninja-c Out_ios_ia32/release-iphonesimulator Iossim apprtcdemo;}function Build_iossim_ia3

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

poll is a way to persist after a connection is opened, waiting for the server to push the data back down. IFrame Stream The IFRAME stream is to insert a hidden iframe in the page, using its SRC attribute to create a long link between the server and the client, and the server transmits the data to the IFRAME (usually HTML, the JavaScript that is responsible for inserting the information) to update the page in real

The combination of gstreamer and webrtc is a little breakthrough, gstreamerwebrtc Today, I found a fork killer in gstreamer, and quickly came up with a general framework and solution plan, using the gst-inspector to perform object introspection attribute detection first, then, the gst-launcher tool is used for Pipeline Test. Finally, the channel Logic Source Code is implemented using c to implement webrtc-

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 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: