webrtc

问题 I am getting this error on console while refreshing the page.. Everythng else works fine the Chats and everthing.. just the streaming part is not working NotSupportedError: MediaStreamError at module.exports (http://192.168.1.10:9966/index.js:3081:17) at Object.1.getusermedia (http://192.168.1.10:9966/index.js:4:1) at o (http://192.168.1.10:9966/index.js:1:265) at r (http://192.168.1.10:9966/index.js:1:431) at http://192.168.1.10:9966/index.js:1:460 while creating a video chat app This is my

问题 Got very simple code: <script type="text/javascript"> pc = new window.RTCPeerConnection(); pc.onicecandidate = function(event) { console.log("onicecandidate\n", event); } pc.onicegatheringstatechange = function(event) { console.log("onicegatheringstatechange\n", event); } dc = pc.createDataChannel("dataChannel"); errFunc = function(err) { console.log("errFunc\n", err); } successCback = function() { console.log("setLocalDescription is a success\n"); } pc.createOffer() .then(function(offer) {

问题 Got very simple code: <script type="text/javascript"> pc = new window.RTCPeerConnection(); pc.onicecandidate = function(event) { console.log("onicecandidate\n", event); } pc.onicegatheringstatechange = function(event) { console.log("onicegatheringstatechange\n", event); } dc = pc.createDataChannel("dataChannel"); errFunc = function(err) { console.log("errFunc\n", err); } successCback = function() { console.log("setLocalDescription is a success\n"); } pc.createOffer() .then(function(offer) {

问题 I used the the information from the link (here) and got the following code, which helps me to record a video with my webcam. The code allows me to record a video and makes it available to download. However, I want to save the recorded video automatically to a local folder. How can I do that? <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script> <section class="experiment"> <div class="inner"> <button style="float: right;" id="fit-to-screen">Fit to Screen!</button> <label for=

问题 I am doing a webRTC videoCall application . At apoint I need a voice record ( Normal), So I just removed the audio track from peerconnection and after record I need to add audio track to peerconnection . But i cann't do it !! public void removeAudioTrack() { List<RtpSender> senders = new ArrayList<>(); senders.addAll(peerConnection.getSenders()); try { for (RtpSender sender : senders) { if (sender.track() != null) { if (sender.track().id().equals(AUDIO_TRACK_ID)) { boolean flag =

问题 In opentok, with OT.initPublisher, you only can pass a deviceId to the audioSource. Does someone know a method to stream an audio file ? For example, I have done this: navigator.getUserMedia({audio: true, video: false}, function(stream) { var context = new AudioContext(); var microphone = context.createMediaStreamSource(stream); var backgroundMusic = context.createMediaElementSource(document.getElementById("song")); var mixedOutput = context.createMediaStreamDestination(); microphone.connect

问题 I am brand new into PeerJs and WebRTC. I have got a 1:1 NodeJS/PeerJS application working in my remote server and that works great. However now I want to explore extending this to a 1:N model where a host ID can have multiple peers connecting to them and each of the peers can receive every other connected peer's audio/video. I am ok with about 4-5 parties in a call for now so a mesh architecture is fine. In the future I would progress into a Media server based architecture to get more

问题 I'm using a webRTC application with a simple-peer npm package. I want to know what is the purpose of all these topics (SFU., Janus, mediasoup or medooze.) and how can I integrate them to make my application performance greater? PS: I'm using a node.js server the bundle the requesting and signaling between peers on my architecture. are those servers and services required to make my application performance well? Hope I could find an answer here ... 回答1: With regular webrtc every peer needs to

问题 I'm using a webRTC application with a simple-peer npm package. I want to know what is the purpose of all these topics (SFU., Janus, mediasoup or medooze.) and how can I integrate them to make my application performance greater? PS: I'm using a node.js server the bundle the requesting and signaling between peers on my architecture. are those servers and services required to make my application performance well? Hope I could find an answer here ... 回答1: With regular webrtc every peer needs to

问题 I'm using a webRTC application with a simple-peer npm package. I want to know what is the purpose of all these topics (SFU., Janus, mediasoup or medooze.) and how can I integrate them to make my application performance greater? PS: I'm using a node.js server the bundle the requesting and signaling between peers on my architecture. are those servers and services required to make my application performance well? Hope I could find an answer here ... 回答1: With regular webrtc every peer needs to