webrtc to rtmp

Very interesting website http://io13webrtc.appspot.com/#1HTML5 about using WEBRTC http://www.html5rocks.com/en/tutorials/getusermedia/intro/HTML5 example of setting resolution https://simpl.info/getusermedia/constraints/HTML5 examples of various special effects http://webcamtoy.com/zh/app/Examples of HTML5 recordings http://www.webaudiodemos.appspot.com/AudioRecorder/index.htmlGoogle's latest HTML5-based code HTTPS://GITHUB.COM/GOOGLECHROME/WEBRTCSIPM

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

The specific parameters in the code meaning, you can go here: https://wiki.videolan.org/Documentation:WebPlugin/ Looking for half a day, HTML5 can support rtmp but can't play Rtsp,flash also stop in Rtmp, finally colleagues recommended a good East open source VLC, ask Google great God, this goods can be used to make each browser (IE ActiveX mode, Google, FF) to achieve playback RTSP video stream, it is very

A background: Users need to watch the video through flash or mobile phone, and most cameras are RTSP protocol, so need to do a relay. Resources: http://blog.csdn.net/firehood_/article/details/8813587 http://blog.csdn.net/firehood_/article/details/8783589 These two articles detail the Mp4v2 save the file and the 264 files through the rtmp live, the two aspects of interest can be directly read the two articles. This article is only read the two articles

It is not recommended to compile individual modules in the WEBRTC separately for use. Yesterday, I was fortunate enough to ask the Google forum about the delay in computing the AECM module, and Project member said churn this delay actually didn't help the AECM effect, which only sped up the convergence of the built-in latency estimator when the AECM started, and if the delay in the update was incorrect, it would even make AE The CM built-in delay

-ndk-r9d/toolchains/arm-linux-androideabi-4.8/prebuilt/linux-x86/bin/ ARM-LINUX-ANDROIDEABI-GCC--cross-prefix=/home/wangrui/java/android-ndk-r9d/toolchains/ arm-linux-androideabi-4.8/prebuilt/linux-x86/bin/arm-linux-androideabi---nm=/home/wangrui/java/android-ndk-r9d /TOOLCHAINS/ARM-LINUX-ANDROIDEABI-4.8/PREBUILT/LINUX-X86/BIN/ARM-LINUX-ANDROIDEABI-NM--sysroot=/home/wangrui/ java/android-ndk-r9d/platforms/android-9/arch-arm/--extra-cflags= '-o3-fpic-dandroid-dhave_sys_uio_h=1- Mfloat-abi=softfp-

are also modified. 6. Connect the phone with the data cable, turn on the USB mode, install the ADB tool sudo apt-get install android-tools-adb 7, go to the SDK directory, my is/home/wangrui/java/sdk/platform-tools Execute command: ADB shellWill go into the phone and execute the command: CD SDcardGo to the SD card and locate the Arm-nginx directory 8. The complete command is: Perform ./nginxPrompt error: Nginx: [alert] could not open error log file:open () "/home/wangrui/arm-nginx/logs/error.

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

Article Link: http://justcoding.iteye.com/blog/2110275Ckplayer, which is all called super cool FLV player, it is a software for playing video on the Web page, supported by the format of: HTTP protocol on the FLV,F4V,MP4 format, while supporting RTMP video streaming format playback, this player is characterized in that the user can define the player's style, such as Play/pause button, mute button, full screen button is called external image interface,

to real-time requirements are very high, such as 0.5s or less, this is a good choice. The former mimic Spydroid wrote a proposed RTSP server, in fact, is options,describe,setup,play,pause,teardown these steps, the agreement with the most extensive, on-line introduction is more. To really understand the RTSP protocol, the C + + language is good to see live555.RTMP protocol, own recent research, if interested, can look at my other articles.RELATED Link

protocol, server, to HTTP. and gradually adapt to the development and demand of the network, complex and changeable network environment, only to generate the HTTP video protocol.----Invited to update.The application scenarios for different protocols have been explained separately above.Pure protocol, HTTP is very simple, the RTSP family is more complex, rtmp did not know in depth.If you just want to do an application, or use it, then HTTP is enough,

You have uploaded the file to csdn at http://download.csdn.net/detail/avsuper/7421647... This program can record the camera video and microphone audio as FLV files. Video Compression uses H.264 encoding, and audio compression uses AAC encoding. rtmp live video can be synchronized (the server end must be FMS/wowza/red5 ). Information such as bit rate, resolution, frame rate, and key frame interval can be selected. No. 1 classroom network (http://www

Adboe's red5 Streaming Media Server is free and open-source. It can be used with flash, but the use of Java and Android as client calls is a trigger. Adobe's red5 Source Code contains an rtmpclient class, which is not complex in use, but cannot be called successfully. Observe the log and find that the connection is closed when the stream is created after the connection is successful. What I can think of is that the current version of the red5 server used by the company may be incompatible with t