install webrtc

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-

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

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

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

WEBRTC IntroductionWEBRTC provides three types of APIs: MediaStream, namely Getusermedia Rtcpeerconnection Rtcdatachannel Getusermedia has been supported by Chrome, opera and Firefox.rtcpeerconnection is currently supported by Chrome, opera and Firefox. Chrome and opera provide an interface named Webkitrtcpeerconnection,firefox with the name Mozrtcpeerconnection.Rtcdatachannel is only available in Chrome, Opera 18 and Firefox 22

WEBRTC is a set of new standards for media data transmission based on the browser side, introducing a number of new concepts, including Dtls, SDEs, DTLS-SRT, ice, turn, Rtp-mux, BWE, FEC jsep, Tricle-ice and other terms,This article first said Dtls, DTLS-SRTPDTLS: Full name Datagram Transport Layer Security, which is UDP + secure, the datagram layer is safe, DTLS employs TLS security mechanism, but is more lightweight,

WEBRTC Technology Group: 234795279 1. Voiceengine CODEC data structure WEBRTC, a struct struct codecinst is used to represent a specific audio codec object: struct Codecinst { int pltype; Payload Type Payload char plname[32];//payload name payload, 32 characters representing int plfreq; Payload frequence Load Frequency int pacsize; Packet size package int channels; Chan