safari webrtc

016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very much

When deploying WEBRTC or SIP-to-peer scenarios, you will often encounter environments that are not penetrated by peerThis is where the tunserver comes in.Here we use turnserver-0.7.3Download confuse dependent librarieswget http://savannah.nongnu.org/download/confuse/confuse-2.7.tar.gzTar zxvf confuse-2.7.tar.gzCD confuse*./configureMake make installDownloadwget http://downloads.sourceforge.net/project/turnserver/turnserver-0.7.3.tar.bz2Tar jxvf turns

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version of TELEMCU adds Android phone-side WEBRTC video conferencing capabilities, Android phone installs Chrome browser after loading TELEMCU WEBRTC client TeleWeb can directly participate in video conferencing,At the same time, TeleWeb can support two WEBRTC client-to-

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 Point-to-point video calling system Main functions:1, based on the WebSocket online user list;2, use WebSocket as signaling channel, build WEBRTC video call.Github:https://github.com/graceup/webrtcDevelopment Ide:myeclipse 8.6 Engineering Code: UTF-8Environmental requirements: 1, TOMCAT requires more than 7.0 of the versionNote: When deploying, you need to change "ws://localhost:8080/" in the Js/con

WebRTC really is not very good to get, currently only the PC-side web page and mobile phone-side web page video. But there are still some problems. 1, both must use Firefox 2, feel pc-side camera shot out of the screen can also, the phone side a little bit of spending 3, enter the room after a period of time to show two video ~~~~APPRTC demo has not been tuned, the problem in Turnserver , and then sent the article. There are a lot of APPRTC on the Int

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 on. Take a look at the following characters:

H264 code Stream parsing, online has a lot of open source files; The general analysis is: Obtain Nalu,sps,pps,nalu type,slice type, obtain QP and so on; The computation can be obtained by the bitwise operation of C + +, but the structure can be obtained directly. Here is the WEBRTC in the H264 parsing Related: In the WEBRTC, about the H264 related source files in: webrtc58\src\

reproduced in the original: http://www.cnblogs.com/mod109/p/5469799.html thank you very much. The WEBRTC audio processing module is divided into noise reduction ns, Echo cancellation AEC(Echo control Acem), Automatic control gain AGC, Mute detection section. In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and

Recently doing a mobile end with mobile, web-side text, video, voice chat features. Text chat using WebSocket, a lot of information on the Internet, there is no difficulty. But in the video, voice chat encountered a small difficulty. have been looking for some of the SDK to quickly develop, such as Opentok, cloud communications, etc., but the project is used in the intranet, these SDKs must be used in an external network, you need to obtain signaling on their servers. Later, I will try to use

Crosswalk QuickStart, using WEBRTC (HTML) to start developing video callsInstall PythonDownload the installer from http://www.python.org/downloads/After the installation is complete, add the environment variable again.Installing Oracle JDK Download page:http://www.oracle.com/technetwork/java/javase/downloads/Select the Java version to download (recommended Java 7). Select a JDK to download and accept the license agreement. Once downlo

Some time ago in the audio and video version of iOS, so the title changed to Android IOS WebRTC audio and Video development summary, the following summarizes some of the experience in the development process:1. iOS WEBRTC audio and video compilation and download: have android WebRTC compile download experience and then go to get IOS, you will find a lot easier, a

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

Recently, the tutor asked to study WebRTC, hoping to use our ICT2 system in the future.But never did the foundation of the web, whether front-end or back-end, HTML, JS all learn from the beginning. HTML is good to say, not too complicated things.JS is a bit difficult, roughly turned over the JS authoritative guide book, understand the basic grammar, also is enough to deal with. But it's completely out of the picture of the various objects built into t

In the next is WEBRTC development novice, at present encountered a problem, turned over to have not understood. Maybe English is not good, look at the document to see blindfolded, so has not found a solution.Development environment:node. JS Server builtI'm using Socket.io to do communications now.Development Purpose:A classmate to B students to initiate a request, B received after the two sides live video.If there is a clear classmate trouble tell me

The development of video conferencing based on the third party WEBRTC open source platform is not very difficult, mainly the business aspects. However, once involved in the core of the underlying issues need to read the source code, to find out the bug, the difficulty is not small.The project needs to analyze the creation process of peerconnection.assuming clienta,clientb is divided into offer and answer. Offer end PC =new rtcpeerconnec

write on the frontA: The purpose of writing a blog1. Self-study of the hard self-evident.2. All kinds of information on the Internet is a mixed bag, many are outdated.3. Based on the latest WEBRTC source to share some experience in their work.4. If you write a good people clap, write bad don't spray. Money to hold a field, no money ...Two: Compile compile or compile1. It is best to prepare a VPN, do not think of someone to copy the code to upload to t

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4401075.htmlThe first two blog posts complete the WEBRTC audio and video collection module, and the next step is to introduce the key audio and video coding modules. However, before introducing the audio and video coding module, we need to introduce the channel concept, and the transmission flow of each WEBRTC data is encapsulated into a channel

, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen

This article mainly introduces to help a programmer solve WEBRTC doubt process, the article from the blog Garden Rtc.blacker, support original, reprint please explain the source (www.rtc.help)This article mainly comes from the mail, why I will be specially organized into essays, mainly based on the following reasons:1, the author email me The purpose is to ask questions, but he asked questions in a way worthy of praise, asked very specific (if asked t