jitsi webrtc

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

gcc-c++ gcc-g77 Flex Bison autoconf automake bzip2-devel zlib-devel ncurses-devel libjpeg-devel Libpng-dev El libtiff-devel freetype-devel pam-devel openssl-devel libxml2-devel gettext-devel pcre-devel3. Installation dependencies3.1 mac./scripts/installmacdeps.sh3.2 Ubuntu./licode/scripts/installubuntudeps.sh4. Installing Licode./scripts/installerizo.sh. /scripts/installnuve.sh5. Mounting Base Example./scripts/installbasicexample.sh6. Run Licode and examples, run at two terminals, or run in the

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve

First, an overview of the WEBRTC audio processing flow, see:WEBRTC The audio session is abstracted into a channel channels, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. There are three channel, each channel contains codec and rtp/rtcp send function.In the case of a channel, the application will contain three active threads, a recording thread, an audio receive thread, and a playback thread.1) Recor

WEBRTC IntroductionWebRTC (Web real-time Communications) is a protocol that allows us to implement peer-to-peer on the browser. We can use this protocol to transfer text, voice, video and file content. This article has recorded some personal understanding of my learning process. It is highly recommended to read the documentation for MDN for systematic learning.Simple processFirst, we have a bit a and point B want to communicate with each other. At the

WEBRTC's echo Cancellation algorithm (AEC,AECM) has several important modules:1. Echo Delay estimation2.NLMS3.NLP4.CNG5. Double-ended detection (DT)The following are respectively described:(1) Echo delay estimationecho Delay Length: Based on correlated time delay estimation algorithm (including: Based on the speech signal autocorrelation pitch period): Echo cancellation site, time delay search range is large.WEBRTC's echo delay estimation, which is based on the algorithm of Gips chief scientist

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

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