TURN it’s an extension of STUN, so TURN server has also STUN features. https://code.google.com/p/rfc5766-turn-server/ works also as a STUN, so you can try to write something like this: var pc_config = { “iceServers”: [{ “url”:”turn:my_username@<turn_server_ip_address>”, “credential”:”my_password” }] }; pc_new = new webkitRTCPeerConnection(pc_config);
I came along the same problem and stumbled upon these two gems: https://github.com/helloIAmPau/node-rtc Sadly it is lacking any documentation. However https://github.com/js-platform/node-webrtc seems more reliable to me.
the answer from @Ichigo is correct, but it is a litte bit bigger. Every ICE contains ‘a node’ of your network, until it has reached the outside. By this you send these ICE’s to the other peer, so they know through what connection points they can reach you. See it as a large building: one … Read more
As it was pretty much covered here, what you are trying to do here is not possible with plain, old-fashionned WebRTC (strictly peer-to-peer). Because as it was said earlier, WebRTC connections renegotiate encryption keys to encrypt data, for each session. So your broadcaster (B) will indeed need to upload its stream as many times as … Read more
It’s outdated code. It contains 6 problems that track the evolution of the WebRTC API. TL;DR: It doesn’t work because you’re not checking for errors and you’ve only tested one browser. 1) Old vendor prefixes (remove them): yourConnection = new RTCPeerConnection(configuration); theirConnection = new webkitRTCPeerConnection(configuration); // <– wrong webkit-names won’t work in Firefox or Edge. … Read more
Edit: A nice implementation in github.io taken from comment to another answer( choose “relay” in IceTransports value): Test TURN Server following Benjamin Trent’s advice, I wrote the below code to test TURN server’s connectivity, works on both firefox n chrome: function checkTURNServer(turnConfig, timeout){ return new Promise(function(resolve, reject){ setTimeout(function(){ if(promiseResolved) return; resolve(false); promiseResolved = true; }, … Read more
For RTSP<->WebRTC / RTMP<->WebRTC conversions, you need to run some kind of WebRTC gateway / media server software that works with all these formats/protocols and can transmux between all of them. Try Wowza / Unreal Media Server / Flashphoner. https://en.wikipedia.org/wiki/Comparison_of_streaming_media_systems So in your case you want to publish the screen from browser to media server … Read more
STUN/TURN is different from signaling. STUN/TURN in WebRTC are used to gather ICE candidates. Signaling is used to transmit between these two PCs the session description (offer and answer). You can use free STUN server (like stun.l.google.com or stun.services.mozilla.org). There are also free TURN servers, but not too many (these are resource expensive). One is … Read more
Webrtc is considered as comparatively much better pre-processing tool for Audio and Video. Webrtc native development includes fully optimized native C and C++ classes, In order to maintain wonderful Speech Quality and Intelligibility of audio and video which is quite interesting. Visit Reference Link: https://github.com/jitsi/webrtc/tree/master/examples regularly. As Problem states; I want to record but smaller … Read more
You can’t send a UDP packet directly with WebRTC. That would violate the basic security constraints required by the browser. You can send SRTP to an ICE-enabled host. That is probably not what you are looking for. If a browser permitted sending arbitrary UDP packets, then malicious applications could send packets to any host. This … Read more