turn

What may be the expected percentage of connections that will fallback to TURN?

由 倖福魔咒の 提交于 2019-12-24 08:33:04
问题 Say I have built the WebRTC video chat website, some connections after the handshake (ICE Candidates) will go directly p2p, some will use the STUN server, and some will use the "last resort" the TURN server to establish the connection. TURN server based connection is very expensive compared to the direct connection and the STUN connection (which are free) because all traffic must actually go through the TURN server. How can we estimate the percentage of connections of random users that will

rfc5766-turn-server - how to enable TLS and HTTP CONNECT method with it?

由 我的未来我决定 提交于 2019-12-23 10:53:09
问题 I have this following setup for rfc5766-turn-server but i am not sure yet how to enable the TLS in turnserver.conf? Any idea what is missing to make sure TLS is activated and what else related sources are missing? # cat turnserver.conf user=root:root realm=x.x.x.x #no-tls #no-dtls syslog aux-server=x.x.x.x:80 aux-server=x.x.x.x:443 Problem: When TURN client connects with following primitives, to that above TURN server then there is auto TURN session close issue. config: '{"iceServers":[{"urls

Coturn WebRTC issue on AZURE VM (Ubuntu Server 16.04 LTS)

由 强颜欢笑 提交于 2019-12-20 05:37:10
问题 I use webRTC to establish a web browser real-time communication with other browsers. To make sure this connection is always solid I want to set up my own STUN/TURN Server. I decided to use coturn on Azure VM (Ubuntu Server). I followed this documentation to set up coturn on Azure https://www.microsoft.com/developerblog/2018/01/29/orchestrating-turn-servers-cloud-deployment/ . According to [Trickele ICE][1] [1]: https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/ my STUN

螺旋矩阵算法

由 泪湿孤枕 提交于 2019-12-19 05:35:19
最近看到博客里有人介绍螺旋矩阵，忍不住啊～ 以下面的4*4螺旋矩阵为例 1　　2　　3　 4 12　 13　 14　 5 11　 16　 15　 6 10　 9 　　8　 7 我们可以理解为一个二维数组，从四个方向（从左至右，从上至下，从右至左，从下至上）依次递增赋值（超出数组边界或遇到已赋值的数组元素则改变赋值方向，直至完成全部数组元素的赋值）。 初始时，赋值方向为从左至右： /* * 0 turn right * 1 turn down * 2 turn left * 3 turn up */ int dir = 0; 改变赋值方向： dir = (dir + 1)%4; 全部代码如下： /////////////////////////////////////////////////////////// // Copyright (c) 2013, ShangHai xxxx Inc. // // FileName: spiralmatrix-2.cpp // // Description: // // Created: 2013年10月13日 星期日 19时42分47秒 // Revision: Revision: 1.0 // Compiler: g++ // ////////////////////////////////////////////////////////

why the turn server does not support for the tcp connection?

由 ≯℡__Kan透↙ 提交于 2019-12-12 12:22:52
问题 I am new to WebRTC. I need to configure my own turn server for my webrtc application. I was installed my turn server using the below command. apt-get install coturn I need to run the turn server over tcp only. It don't have to use the UDP for anything for the media transfer also it have to use the TCP only. So, I was run the turn server using the below command. turnserver -n -a -v --no-udp --min-port 9002 --max-port 9008 --no-dtls -r "bksystems.org" --listening-port 9001 -X 13.126.11.252 -u

NAT traversal using free STUN and TURN servers in C#

由 馋奶兔 提交于 2019-12-12 09:16:48
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

WebRTC video/audio calling failed 90% of time on different network but got success 90% on same network

由 醉酒当歌 提交于 2019-12-12 05:28:02
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

What type of NAT combinations requires a TURN server?

由 ☆樱花仙子☆ 提交于 2019-12-11 02:19:52
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:

WebRTC, STUN/TURN not working outside LAN

由 旧时模样 提交于 2019-12-10 21:43:55
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

coturn STUN requests work locally, but not for remote connections

由 耗尽温柔 提交于 2019-12-10 18:22:48
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
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