webrtc to rtmp

A similar article above found the time: Problems in rtmp handshake In FFMPEG, rtmp is implemented. In handshake, C1 is, the time field is filled with 0, and the zero field is filled with client_ver. The generated 1528 is treated as follows: Enter the pseudo-random number first, then, encrypt a key in a certain location. Because we do not pay attention to the pseudo-random number generation algorithm and En

: = $ (LOCAL_PATH)/ IncludeincludE $ (prebuilt_shared_library) #Programinclude $ (clear_vars) #LOCAL_ALLOW_UNDEFINED_SYMBOLS: = Truelocal_module: = Hellolocal_src_files: = Main.clocal_c_includes + = $ (local_path)/includelocal_ldlibs: =-llog-lzlocal_shared_ LIBRARIES: =avcodec avdevice avfilter avformat avutil postproc swresample swscaleinclude $ (build_shared_library)C Language Implementation fileWrite C, C + + files to implement the underlying logic functions, eventually compiled to so file by

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

I tried it in version 1.0, and I can support the MP3,MP4,FLV,F4V format. This will be very powerful, you can do music website, do video website. Regardless of the format of the media files in the background of the website, the front end is played by the RTMP protocol with the Flash player. If it's live, use Adobe's own flash live Media Encoder to do provider Red5 uses Java development, function and performance are very good, personally feel no less

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve

First, an overview of the WEBRTC audio processing flow, see:WEBRTC The audio session is abstracted into a channel channels, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. There are three channel, each channel contains codec and rtp/rtcp send function.In the case of a channel, the application will contain three active threads, a recording thread, an audio receive thread, and a playback thread.1) Recor

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's echo Cancellation algorithm (AEC,AECM) has several important modules:1. Echo Delay estimation2.NLMS3.NLP4.CNG5. Double-ended detection (DT)The following are respectively described:(1) Echo delay estimationecho Delay Length: Based on correlated time delay estimation algorithm (including: Based on the speech signal autocorrelation pitch period): Echo cancellation site, time delay search range is large.WEBRTC's echo delay estimation, which is based on the algorithm of Gips chief scientist

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

1, about WEBRTCWebRTC is a very popular project. The first problem encountered is the WEBRTC compilation problem.Fortunately, a company has helped compile and put it in Maven's repo.Address:Http://mvnrepository.com/artifact/io.pristine/libjingleThe update is very fast, and WEBRTC the official Basic sync update.2,android DemoThe project is also within the pristine project:Https://github.com/pristineio/apprtc

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-