webrtc multicast

Reliable Multicast Programming (PGM) This section describes the Pragmatic General Multicast (PGM) multicast protocol implementation in Windows, often referred to as reliable multicast. Reliable multicast is implemented through Windows Sockets in Windows Server 2003 and later

When deploying WEBRTC or SIP-to-peer scenarios, you will often encounter environments that are not penetrated by peerThis is where the tunserver comes in.Here we use turnserver-0.7.3Download confuse dependent librarieswget http://savannah.nongnu.org/download/confuse/confuse-2.7.tar.gzTar zxvf confuse-2.7.tar.gzCD confuse*./configureMake make installDownloadwget http://downloads.sourceforge.net/project/turnserver/turnserver-0.7.3.tar.bz2Tar jxvf turns

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version of TELEMCU adds Android phone-side WEBRTC video conferencing capabilities, Android phone installs Chrome browser after loading TELEMCU WEBRTC client TeleWeb can directly participate in video conferencing,At the same time, TeleWeb can support two WEBRTC client-to-

Transferred from: http://www.cnblogs.com/lingyunhu/p/4232348.htmlToss for one weeks to finally set up the environment of Kurento (development stage of the product, some bugs to solve their own), so write a separate document to introduce.The following starts to introduce Kurento, the article from the blog Garden Rtc.blacker, reproduced please explain the source.What is Kurento?Video conferencing involves a one-to-many, many-to-many, broadcast, transcoding, mixing, co-screen, recording, which requ

WebRTC Point-to-point video calling system Main functions:1, based on the WebSocket online user list;2, use WebSocket as signaling channel, build WEBRTC video call.Github:https://github.com/graceup/webrtcDevelopment Ide:myeclipse 8.6 Engineering Code: UTF-8Environmental requirements: 1, TOMCAT requires more than 7.0 of the versionNote: When deploying, you need to change "ws://localhost:8080/" in the Js/con

Google's Turnserver download method:svn checkout http://rfc5766-turn-server.googlecode.com/svn/branches/v3.2/ Rfc5766-turn-server-read-onlyAbout the application of WEBRTC Google gives an example:https://apprtc.appspot.com/(need FQ, sometimes fq may not be able to land, it is estimated that the use of too many people)I was always curious about the way he used turn, and then finally figured out what was going on. Take a look at the following characters:

H264 code Stream parsing, online has a lot of open source files; The general analysis is: Obtain Nalu,sps,pps,nalu type,slice type, obtain QP and so on; The computation can be obtained by the bitwise operation of C + +, but the structure can be obtained directly. Here is the WEBRTC in the H264 parsing Related: In the WEBRTC, about the H264 related source files in: webrtc58\src\

reproduced in the original: http://www.cnblogs.com/mod109/p/5469799.html thank you very much. The WEBRTC audio processing module is divided into noise reduction ns, Echo cancellation AEC(Echo control Acem), Automatic control gain AGC, Mute detection section. In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and

Layer 2 multicast configuration case requirements 1. Three S3600-SI and a S3600-EI composed of two layer multicast network, SwtichA selected S3600-EI, SwitchB, SwitchC, SwitchD selected S3600-SI; 2. multiple Multicast sources can be connected to any of the four S3600 Series Products, and the receiving end can only be connected to SwitchA to receive

the role of multicast and IGMP---------------------------------The so-called multicast, as opposed to unicast and broadcast, refers to the network host will send the data at once to multiple target hosts belonging to the same group. The main use of the IGMP protocol. IGMP is the meaning of Internet Group Management protocol. The protocol is used toThe multicast g

IOS multicast delegate (GCDMulticastDelegate), iosdelegate In IOS, there are several methods to implement callback: Among the above four types, delegate and block are commonly used in my own projects. In reality, callback requirements are also divided into two types One-to-one callback can be implemented using delegate and block in IOS. The one-to-many callback is basically the notification center. If you have a requirement, we will use image downloa

In routing failures, downstream devices cannot learn to troubleshoot multicast routing on the router as follows: Network environment Multicast data is forwarded to the Routera,routera on the vlanif port to terminate the VLAN information of the multicast source through two layers, all three-layer interfaces (both physical and logical) enable

Multicast programming -- Role Summary of setsockopt () Option Parameters So_rcvtimeo so_sndtimeoSets the network transmission and receipt latency. In the send (), Recv () process, sometimes due to network conditions and other reasons, the sending and receiving cannot be performed as expected, but the sending and receiving time limit is set; doubt: if it times out, it cannot be received or cannot be sent, what will happen? So_rcvbuf so_sndbufBy defaul

Definition : Delegate It is a type of method reference stored in an object, and also a type of safe function pointer. One way to understand delegation can regard the role of delegation as specifying a name for the method signature. The delegate definition is similar to the method definition, but there is no method body. The keyword delegate must be added before the defined delegate name. Because the definition delegate basically defines a new class, you can define the delegate anywhere in

, which are called multicast delegates. IfYou can call multiple methods in sequence by calling multicast delegates. Therefore, the delegate signature must return void (otherwise, where should the return value be sent ?) (When a delegate only contains one method, its return type declaration can refer to the encapsulated method, not necessarily void ). In fact, if the compiler finds that a delegate returns vo

Set up Multicast Server Step 1: RunProgramThen select "media-streaming "; Step 2: click Add to select the file to be played (take the WMV file as an example) and click stream "; Step 3: set three parameters for stream output: source, target, and option. The source has been specified just now. Click "Next "; Step 4: Select "display locally", select "RTP/MPEG transport stream", and click "add "; Step 5: if an IPv6

The test code for multicasting is as follows:#include The following error occurred during the Ubuntu compilation run:The answers to the relevant information are as follows:It means that the tool was trying to use multicast and the network interface doesn ' t support it there be, likely causes :· Your machine doesn ' t has multicast support enabled. For example, on Linux and FreeBSD it's possible to compile

to a unique destination. In a conflict domain, all hosts can receive a unicast frame sent by the source host, but other hosts will discard the received frame when the destination address does not match the local MAC address, and only the true destination host will receive and process the received frame.BroadcastingThe second type of transmission is broadcast, which indicates that frames are sent from a single source to all hosts on a shared Ethernet network. The destination MAC address of the b

This article original from Http://blog.csdn.net/voipmaker reprint annotated source. I built a communication learning Exchange Group, 45211986, Welcome to join. WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but

Recently doing a mobile end with mobile, web-side text, video, voice chat features. Text chat using WebSocket, a lot of information on the Internet, there is no difficulty. But in the video, voice chat encountered a small difficulty. have been looking for some of the SDK to quickly develop, such as Opentok, cloud communications, etc., but the project is used in the intranet, these SDKs must be used in an external network, you need to obtain signaling on their servers. Later, I will try to use