webrtc android

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-

. 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

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

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

based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each client needs to establish peerconnection with other N o

Recently the major live sites are compared to fire, want to explore how to play. But read a few Daniel's answer, feel there are too many unfamiliar things, try to get up a little higher cost. Found that there is a thing called WEBRTC, someone has analyzed he is not suitable for the flow of large numbers of live. But I'm just playing with it and feeling the video connectivity.The first thing I saw on GitHub was a demo of all the APIs, and an example of

WEBRTC FEC (forward error-correcting code) is an important part of its QoS, which can be used to recover original data packets when packet loss is lost, reduce retransmission times, reduce latency and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its rationale. Redundant coding To understand the FEC in WEBRTC, you need to first understand red Packet. The

Quality Scaler is WEBRTC in accordance with the video quality, adaptive adjustment resolution of the scheme, the idea is generally: to observe the video encoding loss frame rate and QP changes, determine whether the capturer in the adaptor to adjust the encoding resolution. Its code is located at:$ (ROOT)/src/webrtc/modules/video_coding/utilityThe implementation is fairly simple,Observe the average QP and d

The main reference in this document is [1], which takes the code from the reference article. But [1] did not upload the complete code. Environment configuration can refer to the previous article [2] The main realization of the video in the local transmission. It took a day. As for WEBRTC video capture, codec, please refer to the online blog, here is not mentioned. Compile using CMake to generate makefile, step cmake. Note the points that follow, an

Resources:Http://bucephalus.org/text/CanvasHandbook/CanvasHandbook.html#getcontext2dHttps://developer.mozilla.org/zh-CN/docs/Web/HTML/CanvasHttp://www.w3school.com.cn/html5/html5_canvas.aspHttps://developer.mozilla.org/zh-CN/docs/Web/API/HTMLCanvasElementis a new element of HTML5, you can use JavaScript scripts to draw graphics. For example: paint, synthesize photos, create animations and even real-time video processing and rendering.Mozilla programs are supported from Gecko 1.8 (Firefox 1.5) .

poll is a way to persist after a connection is opened, waiting for the server to push the data back down. IFrame Stream The IFRAME stream is to insert a hidden iframe in the page, using its SRC attribute to create a long link between the server and the client, and the server transmits the data to the IFRAME (usually HTML, the JavaScript that is responsible for inserting the information) to update the page in real time. The advantage of IFRAME streaming