red5 webrtc

from a downhill racing race. Most of the video remains the same, except that the moving parts, i.e. the car and the audience, need to be encoded as P-frames without changing the video. The I frame is generated as a new reference point for P frames. Usually create an I-frame when the image changes very much, such as: panning, scene switching, a large number of actions, sudden disappearance and other scenes. error recovery mechanism:it is suitable for the error recovery mechanism of various packe

The previous article (WEBRTC Audio-related Neteq (a)) is an overview of Neteq, know that it is mainly used to solve the network delay jitter drops and other problems to improve the voice quality, but also know that it has two large units of MCU and DSP components. MCU is mainly received from the network of voice RTP packets into the packet buffer, but also based on the calculated network delay and jitter buffer delay and the feedback from the DSP unit

WebRTC (Web Real time communication) is not Google's original technology, in 2010, Google bought about $68.2 million for VoIP softwareDeveloper Global IP Solutions Company, open source WEBRTC real-time communication project.Voice engine is the gips of voice communication, it is mainly through a series of transmission control to achieve low bandwidth transmission of real-time voice, Gips speech engine hasa w

General Statement In the previous article, we explained how to integrate the OPENH264 codec into the WEBRTC, but OPENH264 can only encode baseline H264 video, and in terms of encoding quality, X264 is the best, This article will explain how to integrate the X264 encoder into the WEBRTC, in order to achieve decoding, at the same time to use the ffmpeg. The overall process, as before, is divided into the re-

The most recent time you've been doing WEBRTC-based Android apps has encountered some problems releasing resources, which are now recorded for memos.The official Apprtcdemo is too simplistic and many questions are not involved.1. Releasing the peerconnection resource problem.Scenario: A and B make a call (Video call)Now stop the call in B.Error: After B terminates the call, the terminal a program will exit unexpectedly.Analysis: When A and b make a ca

This article is mainly their own previous research WEBRTC code structure when some information (including ANDROID,IOS,PC), the article from the blog Garden Rtc.blacker, reproduced please explain the source.1, WEBRTC module: Audio data acquisition, sending, receiving, playback call process:2, WEBRTC module: Video data acquisition, sending, receiving, playback call

. RecordeR.connect (context.destination); }3. Real-time data exchangeWEBRTC's other two api,rtcpeerconnection are used for point-to-point connections between browsers, Rtcdatachannel for Point-to-point data communication.The rtcpeerconnection has a browser prefix and is mozrtcpeerconnection in the Chrome browser for the Webkitrtcpeerconnection,firefox browser. Google maintains a library of adapter.js that is used to pump out differences between browsers.var datachanneloptions = { Ordered:false,

We have introduced WebRTC and read the example of https://apprtc.appspot.com/on the Internet (which may need to be accessed through a wall). This example is an application deployed on the Google App Engine and relies on the Gae environment, the background language is Python and also relies on the Google App Engine channel API. Therefore, it cannot be run locally or be expanded. After studying the Python source code in the example, I decided to impleme

As early as 2014 through the WebRTC realized the PC client real-time video voice, then the establishment of peer-to WEBRTC with the Libjingle library, using the Peerconnection API implementation. Later in the Remote Desktop, file transfer requires point-to-point connection, the Libjingle library for a period of time, found a few problems:The 1.libjingle library is built using the XMPP protocol, but our clie

WEBRTC, also known as "Web Instant Messaging", is a set of API functions that have been certified by the organization to support voice calls, video chats, and peer sharing files between browsers.This protocol mainly includes: Getusermedia,rtcpeerconnection,rtcdatachannels,getstats these modules. The function of Getusermediagon is to allow the browser to use other devices such as a webcam, a mobile phone, and rtcpeerconnection to create a connection be

Transferred from: http://www.cnblogs.com/fangkm/p/4401143.htmlFinally talked about the video data encoding send module, not easy. Overall also looked at a lot of time WEBRTC source, the biggest feeling is that each module in the development of the time is very independent, each module has defined its own set of interfaces, the last string up when adding a variety of adaptation objects to transfer. This gives us those who have just started to read the

SchemaDescription. CAddWhatever isNecessary toEnableStart-up Daemon forthe/usr/Local/bin/turnserver.2)IfYou Do notWant the Turnserver toBe a system service, ThenYou canStart/stop it"Manually",usingThe"Turnserver"Executable withAppropriate options (see the documentation).3) to Create Database Schema, useSchema inchfile/usr/Local/share/turnserver/Schema.SQL.4) forAdditional information, run: $ Man Turnserver $ mans turnadmin $ man turnutilsCreate a user.db file in the root directoryStart with Tu

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Client access to the IMS network, to achieve interoperability with the IMS client, because the standard difference requires a gateway device, Telemedia Media Server to achieve this function, responsible for WEBRTC media layer conversion, including DTLS-SRTP to RTP, and opus to g729/ g711 conversion, Ice-lite support, SBC access to IMS by Telemedia implementation,

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WEBRTC standard customizes how real-time media is transmitted to and from the web, but there is no specification for multiplayer video and is also a challenge for WEBRTC in enterprise solutions, WEBRTC technical VideoThe conference programme can be summed up in several1. Mesh S

Recently flv.js things seem to have ignition, and again to the MSE this thing to bring up.MSE (Media source extensions) is a new function of HTML5, and the general function is to realize streaming media function.If the MSE with WEBRTC and JS binary processing, then you can implement the server to send video to one of the browser users, the browser users will then transfer video streaming to other users of the function. is a web-side in the Peer-to-pee

The newer WEBRTC source has no voiceengine structure corresponding to the vidoeengine, replaced by Meidaengine. Mediaengine contains the Mediaengineinterface interface and its implementation compositemediaengine,compositemediaengine itself is also a template class, two template parameters are the audio engine and video engine respectively. Compositemediaengine derived classes Webrtcmediaengine dependent template parameters are Webrtcvoiceengine and We

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC clients establish a call through interactive candidate to achieve NAT traversal, if these interactions candidate in the Offer/anwser SDP will lead to through the establishment time is very slow,Because the WEBRTC client needs to communicate with the Stun/turn server to get these candidate, the entire process is block, in order to increase the speed of call establ

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WebRTC-Peer part uses the ice framework, the ICE framework includes the Stun,turn, and one problem with the actual use of WebRTC to develop audio and video applications is that calls are built very slowly because the ice process takes too much time for the client to communicate with the stun server before initiating the c

This article mainly introduces the last encounter of some machines do not see the video problem, the article from the blog Park Rtc.blacker, reproduced please indicate the source.Before doing the video chat app has been running well, a few days ago customer feedback said in Samsung 9100. You can't see your own image on the Android4.0.3.After a search to find the WEBRTC is the bottom of the bug, has been repaired and feedback to the community, the foll

Transferred from: http://www.xuebuyuan.com/1248366.htmlThe bandwidth adaptive algorithm in WEBRTC is divided into two types:1, the originator bandwidth control, the principle is the RTCP in the packet loss statistics to dynamically increase or decrease the bandwidth, in the reduction of bandwidth using the TFRC algorithm to increase the smoothness.2, the receiver bandwidth estimation, the principle is and by the receipt of RTP data, the estimated band