stun

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

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

问题 I have created video chat app using webRTC and Django channel.My app working 90% of time properly when using it in the same network but fail when using it on a different network.I am not able to see a video of remote person when using it on a different network. I used chrome://webrtc-internals/ to trace my webRTC response and I got iceconnectionstate : failed when called using different network Below screenshot is taken when I got success in the same network And after success also it gave

问题 So this is a two part question: Part a: I'm trying to implement a secure connection to the KMS. From the documentation, I've understood that KMS Configuration file would need to be updated with the SSL certificate and then the HTTPS connection from the client can be made. Please let me know if there are any other steps that are involved in achieving SSL security. Part b: From a better understanding now and from comments from a previous question I posted, Kurento Utils does not connect to KMS

问题 This is how I understand the protocol... user on browser get media stream generates own sdp (data on codecs, security, etc) creates an offer with it send this offer to server via signalling method (websocket) signalling server, looks who is this offer for and redirect it to that client client 2's browser gets the offer with sdp and in turn generate its own offer with its own sdp and send it to server server redirect client 2's offer to client 1. now both know each other specs. Meanwhile,

问题 According to: https://en.wikipedia.org/wiki/Network_address_translation there are 4 types of NAT configurations: Full-cone, Address restricted, Port restricted, and Symmetric. Now, suppose we have client A and client B on separate networks and are hidden behind their own separate NATs. What combinations of "client A NAT type" + "client B NAT type" would require a TURN server to be involved (i.e. not solvable by the STUN protocol) if they want to p2p communicate? For example, I would suspect:

问题 index.html ( Offerer ) var socket = io.connect('http://127.0.0.1:80'); //socket.io socket.emit("player 1"); var iceServers = { iceServers: [ {"url":"stun:turn1.xirsys.com"}, {"username":"myusername","url":"turn:turn1.xirsys.com:443?transport=udp","credential":"mycredential"}, {"username":"myusername","url":"turn:turn1.xirsys.com:443?transport=tcp","credential":"mycredential"} ] }; var offererDataChannel, answererDataChannel; var Offerer = { createOffer: function () { var peer = new

问题 I've successfully made a TURN request to coturn server (https://github.com/coturn/coturn), but failed when executing a STUN request. If I try to STUN connect to coturn server from the same machine running the server using turnutils_stunclient myIP, the server responds with RFC 5780 response 1 0: IPv4. Response origin: : IP1:3478 0: IPv4. Other addr: : IP2:3479 0: IPv4. UDP reflexive addr: IP1:36457 ======================================== RFC 5780 response 2 0: IPv4. Response origin: : IP2

问题 Just installed the coturn server (https://github.com/coturn/rfc5766-turn-server/). It looks like I'm getting only STUN candidates, but no TURN. As a client I'm using node's webrtc-native. I tried with public TURN server and did get 'relay' candidates, but I can't get any with my server. Is there something I'm missing? 来源： https://stackoverflow.com/questions/35709804/missing-relay-candidates-from-turn-server