openwebrtc

WebRTC error while creating video chat app

会有一股神秘感。 提交于 2021-02-11 15:01:28
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

How do peers involved in a p2p communication authenticate each other?

自闭症网瘾萝莉.ら 提交于 2021-02-05 12:22:04
问题 How do peers in WebRTC authenticate each other? 回答1: DTLS in WebRTC uses self-signed certificates. RFC 5763 has the details, in a nutshell the certificate fingerprint is matched against the one provided in the a=fingerprint line of the SDP. 回答2: As the comment from Patrick Mevzek already mentioned: It doesn't depend on DTLS or TLS, if a self-signed certificate is trusted/accepted. This depends only from the peer's trusted certificates. If the client's or server's certificate path/chain

how to do webRTC in reactJs

喜欢而已 提交于 2020-08-10 19:29:48
问题 Am trying to include a webRTC tech to my already existing reactJs App the problem is that react is not recognizing the webRTC API's Line 185:19: 'webkitRTCPeerConnection' is not defined no-undef Line 191:1: 'rtcPeerConn' is not defined no-undef Line 212:3: 'rtcPeerConn' is not defined no-undef Line 214:62: 'rtcPeerConn' is not defined no-undef this function is inside a functional react component function startSignaling(){ displayMessage("start signaling..."); rtcPeerConn = new

how to do webRTC in reactJs

与世无争的帅哥 提交于 2020-08-10 19:29:14
问题 Am trying to include a webRTC tech to my already existing reactJs App the problem is that react is not recognizing the webRTC API's Line 185:19: 'webkitRTCPeerConnection' is not defined no-undef Line 191:1: 'rtcPeerConn' is not defined no-undef Line 212:3: 'rtcPeerConn' is not defined no-undef Line 214:62: 'rtcPeerConn' is not defined no-undef this function is inside a functional react component function startSignaling(){ displayMessage("start signaling..."); rtcPeerConn = new

how ( stop,exit ) video in webrtc navigator user media JavaScript

佐手、 提交于 2019-12-17 12:37:11
问题 how i stop and exit in pure js, stream webcam in WEBRTC api js , i have in my code the following script : <script type="text/javascript"> $(document).ready(function() { $("#abrirModal").click(function() { navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia; var constraints = { audio: false, video: true }; var live = document.getElementById("live"); function successCallback(stream) { window.stream = stream; // stream available to console

Uncaught Error: This Firebase project requires import into Google Cloud Platform to resume service

不打扰是莪最后的温柔 提交于 2019-12-13 04:00:44
问题 Getting this Uncaught Error: This Firebase project requires import into Google Cloud Platform to resume service. When I am trying to conference video call through Webrtc. I got this url- https://firebase.google.com/support/guides/firebase-web#import_your_project_to_the_firebase_console but I don't want to upload my project on cloud. 回答1: I got the solution. Go to this url https://console.firebase.google.com/ and login as gmail account. Click on "Add Project" You will get page like this See

Ways to capture incoming WebRTC video streams (client side)

為{幸葍}努か 提交于 2019-12-10 12:31:36
问题 I am currently looking to find a best way to store a incoming webrtc video streams. I am joining the videocall using webrtc (via chrome) and I would like to record every incoming video stream to from each participant to the browser. The solutions I am researching are: Intercept network packets coming to the browsers e.g. using Whireshark and then decode. Following this article: https://webrtchacks.com/video_replay/ Modifying a browser to store recording as a file e.g. by modifying Chromium

Native Android WebRTC application development

半腔热情 提交于 2019-12-03 18:34:27
问题 I am trying to create an android application for video chat and messaging by using WebRtc Native apis. I have been through several links and found out that most of the documentation for android is vague, specially if you dont know where to start from. I followed the following links, https://webrtc.org/native-code/android/# https://www.chromium.org/developers/how-tos/android-build-instructions But the above mentioned links doesnt make any sense because I want to develop an android application

how to use adapter.js of webRTC-adapter?

风格不统一 提交于 2019-12-02 06:06:18
问题 I am writing a WebRTC application and have the following problem: I want to use the adapter.js library. I have the following index.html: <html> <header> <script src='../out/adapter.js'></script> <script src='../out/main.js'></script> </header> <body> <video id="localVideo" width='500' autoplay></video> </body> </html> and my main.js looks like: var adapter=require('webrtc-adapter'); var localVideo=document.querySelector('video#localVideo'); navigator.getUserMedia(media_constraints,

how ( stop,exit ) video in webrtc navigator user media JavaScript

核能气质少年 提交于 2019-11-27 14:51:51
how i stop and exit in pure js, stream webcam in WEBRTC api js , i have in my code the following script : <script type="text/javascript"> $(document).ready(function() { $("#abrirModal").click(function() { navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia; var constraints = { audio: false, video: true }; var live = document.getElementById("live"); function successCallback(stream) { window.stream = stream; // stream available to console if (window.URL) { live.src = window.URL.createObjectURL(stream); } else { live.src = stream; } $("