webrtc to rtmp

Recently in window is a platform to do a function to capture audio and video through Obs, and through the RTMP protocol to its encoded compressed data into its own program, since the OBS software with very powerful game recording and desktop recording functions, as well as input, output audio device data acquisition and mixing function , the current fight fish game Live is also used by this software as a recording tool.OBS software because of the use

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

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

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

RTMP is a TCP protocol that Flash player receives video from a video server. Adobe has released most of the RTMP protocol, although some details have not been disclosed, but for our own development of a rtmp server is enough, even if not open, many open-source projects have already made a lot of details of the agreement to clarify. At present, many video surveill

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

to the console. Copy the information and save it as a. SDP text file suffix. It can also be used to receive the RTP stream. After "> test. SDP" is added, you can directly Save the SDP information as text. 2.2. Play the RTP that carries the H.264 bare stream. [Plain]View plaincopy Ffplay test. SDP 3. rtmp3.1. send the H.264 bare stream to the rtmp server (flashmedia server, red5, etc) Run the command to send the "Chunwan. h264" of the H.264 bare str

. SDP" is added, you can directly Save the SDP information as text. 2.2. Play the RTP that carries the H.264 bare stream. ffplay test.sdp 3. rtmp3.1. send the H.264 bare stream to the rtmp server (flashmedia server, red5, etc) Run the command to send the "Chunwan. h264" of the H.264 bare stream to the rtmp URL where the host is localhost, the application is oflademo, and the path is livestream. ffmpeg -re

1. Installation commandRuby-e "$ (curl-fssl https://raw.githubusercontent.com/Homebrew/install/master/install)" If you want to uninstall RUBY-E after installation, "$ (Curl- Fssl https://raw.githubusercontent.com/Homebrew/install/master/uninstall) "2. Install Nginx first clone Nginx project to local brew tap Homebrew/nginx Installation: Brew Install Nginx-full--with-rtmp-module at this point, the Nginx and rtmp