getting Uncaught DOMException: Failed to construct 'RTCPeerConnection' username and password is required
来源： https://stackoverflow.com/questions/64154550/getting-uncaught-domexception-failed-to-construct-rtcpeerconnection-username
rfc5766turnserver
getting Uncaught DOMException: Failed to construct 'RTCPeerConnection' username and password is required
来源： https://stackoverflow.com/questions/64154550/getting-uncaught-domexception-failed-to-construct-rtcpeerconnection-username
rfc5766-turn-server - how to enable TLS and HTTP CONNECT method with it?
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
ApprtcDemo with local server works between browsers but not Android native to browser
问题 I am developing a chat application and done with it. Now I want to implement video chat also. After research a lot I decided to go with "WebRTC" library. What I have done? 1) Able to run AppRtcDemo at local server and Its working fine between browsers. Reference : http://www.webrtc.org/reference/getting-started 2) Able to build Android AppRtcDemo.But when I run it say "Cross origin does not support". After research I found in webrtc discussion that to resolve this issue I need to set-up own
Apprtc with coturn STUN/TURN server
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
rfc5766-turn-server as TURN and STUN for webrtc application
问题 I have implemented rfc5766-turn-server and have it running on my own server. in the app, I set : pc_config = {"iceServers": [{"url":"turn:username@<turn_server_address>", "credential":"password"}]}; it seems to be working, but I have couple of questions: is having the app set this way , with rfc5766-turn-server is enough to act as a TURN and also STUN server or do I also need to run a stun server. a turn server is also stun so will the rfc5766-turn-server function as a stun server (on most
Why my turn server doesn't work?
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
missing relay candidates from TURN server
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
How to secure a TURN server for WebRTC?
问题 I've just installed rfc5766-turn-server (https://code.google.com/p/rfc5766-turn-server/) on an Amazon server in order to relay my WebRTC calls. Since authentication username and password will be distributed to every client in WebRTC iceServers , how can I ensure that only my clients use my TURN to relay their call? 回答1: You can use this instead https://code.google.com/p/coturn/. It's evolved from rfc5766-turn-server project. Supported TURN authentication mechanisms: 'classic' long-term
Installing a TURN Server on Ubuntu for WebRTC
How can I install a TURN server on my ubuntu 12.04? Can you share tutorial? I read this tutorial: Implementing our own STUN/TURN server for WebRTC Application . But what I don't understand is how I can I install my own TURN server on my ubuntu 12.04? I am using currently using something like the following code to create the RTCPeerConnection const pc_config = {"iceServers": [{"url": "stun:stun.l.google.com:19302"}, {"url":"turn:my_username@<turn_server_ip_address>", "credential":"my_password"}]}; const pc_new = new webkitRTCPeerConnection(pc_config); And I want to fill the above code's