stun

I'm trying to make a peer to peer app in php to work in the same kind of way skype works. It works with internal addresses, but when I try to send data to my external address the message never gets there. Obviously I'm missing some address translation, but I'm at a loss for how to do that with PHP sockets. Is it possible? This is the code i have at the moment to receive the message, which I run first to wait for some message: <?php $socket = socket_create(AF_INET, SOCK_DGRAM, SOL_UDP) or die("Could not create socket: " . socket_strerror(socket_last_error($socket))); socket_bind($socket, "0.0.0

I try to make a P2P Program and need help with getting through the NAT of the clients. I have read many questions here on stackoverflow, but i never got what the drawbacks and benefits of all the Methods to get through a NAT are. How many routers support which methods? Which methods are commonly used by the big companies? (BitTorrent, TeamViewer,...) What are the drawbacks/benefits of the three methods i listed and which else exists? UPnP requires software support in your router. Even when routers have it, it's sometimes turned off due to security concerns. STUN uses a simple public server to

WebRTC是HTML5支持的重要特性之一，有了它，不再需要借助音视频相关的客户端，直接通过浏览器的Web页面就可以实现音视频对聊功能。而且WebRTC项目是开源的，我们可以借助WebRTC源码快速构建自己的音视频对聊功能。无论是使用前端JS的WebRTC API接口，还是在WebRTC源码上构建自己的对聊框架，都需要遵循以下执行流程： 上述序列中，WebRTC并不提供Stun服务器和Signal服务器，服务器端需要自己实现。Stun服务器可以用google提供的实现stun协议的测试服务器（stun:stun.l.google.com:19302），Signal服务器则完全需要自己实现了，它需要在ClientA和ClientB之间传送彼此的SDP信息和candidate信息，ClientA和ClientB通过这些信息建立P2P连接来传送音视频数据。由于网络环境的复杂性，并不是所有的客户端之间都能够建立P2P连接，这种情况下就需要有个relay服务器做音视频数据的中转，本文本着源码剖析的态度，这种情况就不考虑了。这里说明一下， stun/turn、relay服务器的实现在WebRTC源码中都有示例，真是个名副其实的大宝库。 上述序列中，标注的场景是ClientA向ClientB发起对聊请求，调用描述如下： ClientA首先创建PeerConnection对象

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: This is my previous post: NAT Traversal using only free STUN and TURN servers in C# I am making an online application that communicates with each other peer-to-peer. Most users are expected to be in their private network behind a NAT, and I need to traverse it. I can't afford an external server, so the only things that I can depend on is free servers like Numb (which means I cannot provide my own implementation or extension of it.) However, after some research, I couldn't figure out how to use such servers. I didn't find a library that could

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: Simply, I am going run locally popular example of WEBRTC app: github.com/webrtc/apprtc The apprtc installed, and even works locally without turn server ( "Same origin policy" don't allow use Google TURN server, which works only from apprtc.appspot.com: access-control-allow-origin:" https://apprtc.appspot.com "). But I know that in real internet world (nats and firewalls) I need turn server. So I have decided to use own STUN/TURN server: code.google.com/p/coturn/ I am trying integrate my apprtc with coturn: +apprtc: http://localhost:8080/

I'm trying to make an (apache) server behind multi-layered NAT to be accessible from Internet. Restrictions: Avoid relay. There is a public server (we call him the OldMan) for login / stun, but its bandwidth is too poor to carry relay data. I have no right to configure the NATs manually. 3.User don't have to change their browser or device.(that is, works on android, ios, and any PC) I tried UPnP , but it only works on 1-layered NAT . I tried to search NAT traversal solutions for several months, but in vain. I study STUN/TURN/ICE ,and tried pjsip (icedemo.exe + numb), however, its stun cannot

if I am in a room with other 7 users, I am wondering if WebRTC force every user to establish a connection to each one of other participants. Obviously it would consume something like 7kb/s*7 download and even upload, and many connection cannot handle this if their connection is already busy. Instead with some kind of media relay the bandwidth usage would be only 7kb/s but you would lose bandwidth adaptation between peers. Do you know any media relay, or way to solve this problem? is TURN server ( like https://code.google.com/p/rfc5766-turn-server/ ) suitable for this kind of job ( multicast