coturn

问题 I have configured a coturn server which I need to use with DataChannel for a chat app. I am trying to reach a point where I can get the coturn server provide me with a tcp allocation. I am using the standard call of RTCPeerConnection in JS client, providing the turn URI, username and credentials. The transport I am providing in the URI params is TCP ?transport=tcp . With all this, I am always receiving a UDP allocation and the server logs ChannelBind requests (UDP based) Question: How can I

问题 I have configured a coturn server which I need to use with DataChannel for a chat app. I am trying to reach a point where I can get the coturn server provide me with a tcp allocation. I am using the standard call of RTCPeerConnection in JS client, providing the turn URI, username and credentials. The transport I am providing in the URI params is TCP ?transport=tcp . With all this, I am always receiving a UDP allocation and the server logs ChannelBind requests (UDP based) Question: How can I

问题 I have configured a coturn server which I need to use with DataChannel for a chat app. I am trying to reach a point where I can get the coturn server provide me with a tcp allocation. I am using the standard call of RTCPeerConnection in JS client, providing the turn URI, username and credentials. The transport I am providing in the URI params is TCP ?transport=tcp . With all this, I am always receiving a UDP allocation and the server logs ChannelBind requests (UDP based) Question: How can I

问题 I have configured a coturn server which I need to use with DataChannel for a chat app. I am trying to reach a point where I can get the coturn server provide me with a tcp allocation. I am using the standard call of RTCPeerConnection in JS client, providing the turn URI, username and credentials. The transport I am providing in the URI params is TCP ?transport=tcp . With all this, I am always receiving a UDP allocation and the server logs ChannelBind requests (UDP based) Question: How can I

问题 I have my stun / turn server running on local pc (CoTurn). It is tested on "https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/" and works. I have a domain name and configured the modem with public ip. I configured apache2 to make the site visible to the world. I have active and valid letsencript certificates. Everything works in short. But the test application starts the connection (the external pc communicates with the local via socket.io) but then the video is not seen

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

问题 I can connect in any situation when using appr.tc ice servers (google turn servers). but i can't connect with my own turn server. I did config my own turn server by coturn project . I'm using google's libjingle_peerconnection api to create an Android Application that can perform video call . When i run turn server: <pre> RFC 3489/5389/5766/5780/6062/6156 STUN/TURN Server Version Coturn-4.5.0.5 'dan Eider' 0: Max number of open files/sockets allowed for this process: 4096 0: Due to the open

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 WebRTC calls are not reliable in our application. Sometimes we see the black screen, sometimes we don’t see call start at all and sometimes there are seen huge delays or de-sync in audio/video. Setup: Google’s public STUN server stun:stun.l.google.com:19302 ; TURN server is Coturn hosted in azure in docker; Signaling server is a customized https://github.com/andyet/signalmaster that using express as a web server On client JS is used, with simplewebrtc as client; For iOS support used Cordova

问题 I want to host a TURN server for my WebRTC project, but I don't know what the 'realm' attribute does. I've looked at coturn and pion and they both seem to require me to enter this. Can someone explain what it is? 回答1: It is a "key" to access a group of address that have a common "owner". It is an isolating measure: a TURN server can have one "default realms" or more "named realms" in its database. And any realm has its own users and each of them have a proper login/password couple to identify