webrtc streaming

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

The IIS live Smooth streaming (real-time smooth streaming) is Microsoft's next generation streaming media solution. The technology is to integrate the media transport platform IIS Media Services in the IIS Web to enable the use of standard HTTP Web technology and advanced Silverlight functionality to ensure the best quality and smooth audio and video broadcasts o

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

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

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

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

#!/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

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

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 time. The advantage of IFRAME

There are Licode and kurento in contact.Licode Flaw: Limited documentation support, Licode app client library only JSKurento Advantages: Complete Documentation, demo-ready, Packaging API is more complete. Its main features are: Networked streaming protocols, including HTTP, RTP and WebRTC. Group Communications (MCUs (Multipoint Conferencing Unit) and Sfus (Selective Forwarding unit.) functional

Today let me find a gstreamer of a bull fork of the killer, the mind immediately thought of a general framework and plan, with Gst-inspector first object introspection property detection, and then sacrificed Gst-launcher Broadsword for pipeline test, and finally use C to achieve the pipeline logic source code , you can implement WEBRTC-based video surveillance and live streaming services. Real-time two-pers

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374610.htmlThe first two articles describe the running process of WEBRTC and the use of the framework interface, and then begin to analyze the local audio and video collection process. Due to the large space, video capture and audio capture are divided into two blog posts, where the video capture process is analyzed first. Analysis of the time of the first analysis of the

src\webrtc\examples\peerconnection\client\conductor.cc1, BOOL Conductor::initializepeerconnection ()1.1 WEBRTC::createpeerconnectionfactory();src\talk\app\webrtc\peerconnectionfactory.cc2, BOOL Peerconnectionfactory::initialize ()2.1.1 cricket::mediaengineinterface* peerconnectionfactory::createmediaengine_w() {Return Cricket::webrtcmediaenginefactory::create(Def

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 Videoengine Ultra-Detailed tutorial (iii)--integr

first of all, I admire eguid, whether it is technology or sharing, thanks to his series of Bowen on my first learning process help, so clearly indicated the reprint address: http://blog.csdn.net/eguid_1/article/details/51725970 one, the local push to send the end1, Local: The use of JAVACV (Android and Java Platform recommended JAVACV), FFmpeg, OpenCV or JMF can be very convenient access to the local camera streaming media JAVACV Series arti

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