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SCREEN-CAPTURING.JS
You can click any button. Each button explains usage of a unique method: getSourceId » Simply get "chromeMediaSourceId" from chrome extension.; getCustomSourceId » Simply get "chromeMediaSourceId" from chrome extension however pass yoru own array.; getSourceIdWithAudio » Simply get "chromeMediaSourceId" from chrome extension. (system audiois included)
RECORDRTC AUDIO+VIDEO RECORDING & UPLOADING TO PHP SERVER Difference between those two services and RecordRTC is that: 1) obviously RecordRTC is 100% open-sourced and free. 2) RecordRTC only targets javascript API; NO native support. 3) RecordRTC provides fallbacks to plenty of javascript API/recorders. 4) RecordRTC doesn't use Flash or Java or anything similar. see more. RECORDRTC | WEBRTC AUDIO+VIDEO+SCREEN RECORDING MediaRecorder.start: An audio track cannot be recorded: video/webm;codecs=vp8 indicates an unsupported codec. CAUSE: record into vp8 in firefox 76.0.1 GETDISPLAYMEDIA DEMO getDisplayMedia demo. Purpose of this demo is to test all getDisplayMedia API functionalities. aspectRatio: . frameRate: . resolutions: . cursor: . displaySurface WEBRTC SIGNALING CONCEPTS Signaling is used to exchange session control messages known as SDP; network configurations as ICE candidatess; and media capabilities using same session control messages. Signaling can be done either using copy/paste mechanism; or using a gateway like WebSocket, Socket.io, XMPP or most famous one i.e. session initiation protocol(SIP).
WEBRTCPEDIA! THE ENCYCLOPEDIA! MediaStream.stop is obsolete or removed; how to fix it? var MediaStream = window.MediaStream; if (typeof MediaStream === 'undefined' && typeof webkitMediaStream WEBRTC DEMOS, EXPERIMENTS, LIBRARIES, EXAMPLES WebRTC Scalable Broadcasting. This module simply initializes socket.io and configures it in a way that single broadcast can be relayed over unlimited users without any bandwidth/CPU usage issues. DETECTRTC | IS WEBRTC SUPPORTED IN YOUR BROWSER? Supports color adjustment: Brightness, Contrast, Saturation, Hue and Sharpness control. Supports high quality video resolution (NTSC: 720 x 480 @ 30fps / PAL: 720 x 576 @ 25fps) Support For All Formats: record in DVD+/-R/RW, DVD+/-VR, and DVD-Video. Video input: S-video and RCA type composite video. Audio input : Stereo audio (RCA)mm. WEBRTC DOCUMENTS FOR NEWBIES AND BEGINNERS A few documents for newbies and beginners; TURN server installation Guide: WebRTC RTP Usage: STUN or TURN? Which one to prefer; and why? Share files using File.js WEBRTC SCREEN SHARING Advantages. Share full screen with one or more users in HD format! Share screen from chrome and view over all WebRTC compatible browsers/plugins. You can open private rooms and it will be really "totally" private! Use hashes to open private rooms: #private-room.SCREEN-CAPTURING.JS
You can click any button. Each button explains usage of a unique method: getSourceId » Simply get "chromeMediaSourceId" from chrome extension.; getCustomSourceId » Simply get "chromeMediaSourceId" from chrome extension however pass yoru own array.; getSourceIdWithAudio » Simply get "chromeMediaSourceId" from chrome extension. (system audiois included)
RECORDRTC AUDIO+VIDEO RECORDING & UPLOADING TO PHP SERVER Difference between those two services and RecordRTC is that: 1) obviously RecordRTC is 100% open-sourced and free. 2) RecordRTC only targets javascript API; NO native support. 3) RecordRTC provides fallbacks to plenty of javascript API/recorders. 4) RecordRTC doesn't use Flash or Java or anything similar. see more. RECORDRTC | WEBRTC AUDIO+VIDEO+SCREEN RECORDING MediaRecorder.start: An audio track cannot be recorded: video/webm;codecs=vp8 indicates an unsupported codec. CAUSE: record into vp8 in firefox 76.0.1 GETDISPLAYMEDIA DEMO getDisplayMedia demo. Purpose of this demo is to test all getDisplayMedia API functionalities. aspectRatio: . frameRate: . resolutions: . cursor: . displaySurface WEBRTC SIGNALING CONCEPTS Signaling is used to exchange session control messages known as SDP; network configurations as ICE candidatess; and media capabilities using same session control messages. Signaling can be done either using copy/paste mechanism; or using a gateway like WebSocket, Socket.io, XMPP or most famous one i.e. session initiation protocol(SIP).
WEBRTCPEDIA! THE ENCYCLOPEDIA! MediaStream.stop is obsolete or removed; how to fix it? var MediaStream = window.MediaStream; if (typeof MediaStream === 'undefined' && typeof webkitMediaStream WEBRTC DEMOS, EXPERIMENTS, LIBRARIES, EXAMPLES WebRTC Scalable Broadcasting. This module simply initializes socket.io and configures it in a way that single broadcast can be relayed over unlimited users without any bandwidth/CPU usage issues. WEBRTC DOCUMENTS FOR NEWBIES AND BEGINNERS A few documents for newbies and beginners; TURN server installation Guide: WebRTC RTP Usage: STUN or TURN? Which one to prefer; and why? Share files using File.js MULTIRTC! / RTCMULTICONNECTION DEMO ® MUAZ KHAN What is MultiRTC? It is a skype-like demo using WebRTC for realtime connections! It allows you enable/disable webcams; and join with or without webcams! It allows you share screen using existing peer connections! It allows you share files with preview and download links! It allows you auto translate incoming messages in your ownlanguage !
WEBRTC ONE-TO-MANY VIDEO-BROADCASTING Difference between one-way broadcasting and one-to-many broadcasting. For 10 users session, in one-way broadcasting: 10 RTP ports for outgoing audio stream. 10 RTP ports for outgoing video stream. i.e. total 20 outgoing RTP ports will be opened on your browser. On each participant's side; only 2 incoming RTP ports will be opened. WEBRTC SIGNALING CONCEPTS Signaling is used to exchange session control messages known as SDP; network configurations as ICE candidatess; and media capabilities using same session control messages. Signaling can be done either using copy/paste mechanism; or using a gateway like WebSocket, Socket.io, XMPP or most famous one i.e. session initiation protocol(SIP).
RECORDRTC WEBM TO MP4 USING FFMPEG-ASM.JS ® MUAZ KHAN Using ffmpeg-asm.js WebM can be converted in mp4. WAV can be converted in ogg or mp3 (AAC/Vorbis). WAV can be merged in mp4. If WAV or WebM is 5MB in size; ogg, mp3 or mp4 will be 700kb! Ffmpeg.js is MIT licensed on Github! Documentation. HOW TO USE RTCPEERCONNECTION.JS? (V1.5) This tutorial is out-dated (written in 2013).Please check thistutorial instead:
https://codelabs.developers.google.com/codelabs/webrtc-web/#0 Explainshow to. use
TURN SERVER INSTALLATION GUIDE Last Step: "Permanently" run TURN server instance. nohup turnserver -v -r ip:port -a -b turnuserdb.conf -c turnserver.conf -u turn-username -r ip:port -p turn-password &. You can see that the entire TURN execution command is placed between "nohup" and "&". nohup TURN-execution-command &. HOW TO ECHO CANCELLATION / NOISE MANAGEMENT / IN WEBRTC Echo Scenarios: 44.1 kHz and non-44.1 kHz sample rate mismatches cause echo; The "ambient noise reduction" which can be enabled on the built-in mic on Mac appears to cause a very small amount of echo; Solutions: Ensure both capture and render devices are set to either 44.1 or 48 kHz. You can do this through the "Audio MIDI Setup" application; Disable "ambient noise reduction" ECHO IS A DISTORTION OF VOICE THAT OCCURS EITHER WHEN YOU Echo occurs if input output audio devices are placed close together! In old days; usage of the “headsets” was the only “suggested” solution to overcome the echo! WEBRTC DOCUMENTS FOR NEWBIES AND BEGINNERS A few documents for newbies and beginners; TURN server installation Guide: WebRTC RTP Usage: STUN or TURN? Which one to prefer; and why? Share files using File.js DETECTRTC | IS WEBRTC SUPPORTED IN YOUR BROWSER? Supports color adjustment: Brightness, Contrast, Saturation, Hue and Sharpness control. Supports high quality video resolution (NTSC: 720 x 480 @ 30fps / PAL: 720 x 576 @ 25fps) Support For All Formats: record in DVD+/-R/RW, DVD+/-VR, and DVD-Video. Video input: S-video and RCA type composite video. Audio input : Stereo audio (RCA)mm. WEBRTC SCREEN SHARING Advantages. Share full screen with one or more users in HD format! Share screen from chrome and view over all WebRTC compatible browsers/plugins. You can open private rooms and it will be really "totally" private! Use hashes to open private rooms: #private-room.SCREEN-CAPTURING.JS
You can click any button. Each button explains usage of a unique method: getSourceId » Simply get "chromeMediaSourceId" from chrome extension.; getCustomSourceId » Simply get "chromeMediaSourceId" from chrome extension however pass yoru own array.; getSourceIdWithAudio » Simply get "chromeMediaSourceId" from chrome extension. (system audiois included)
GETDISPLAYMEDIA DEMO getDisplayMedia demo. Purpose of this demo is to test all getDisplayMedia API functionalities. aspectRatio: . frameRate: . resolutions: . cursor: . displaySurface MULTIRTC! / RTCMULTICONNECTION DEMO ® MUAZ KHAN What is MultiRTC? It is a skype-like demo using WebRTC for realtime connections! It allows you enable/disable webcams; and join with or without webcams! It allows you share screen using existing peer connections! It allows you share files with preview and download links! It allows you auto translate incoming messages in your ownlanguage !
WEBRTC ONE-TO-MANY VIDEO-BROADCASTING Difference between one-way broadcasting and one-to-many broadcasting. For 10 users session, in one-way broadcasting: 10 RTP ports for outgoing audio stream. 10 RTP ports for outgoing video stream. i.e. total 20 outgoing RTP ports will be opened on your browser. On each participant's side; only 2 incoming RTP ports will be opened. FFMPEG.JS AUDIO/VIDEO ENCODING/TRANSCODING/MUXING It is suggested to download ffmpeg-asm file and try locally! ffmpeg-asm.js file's default size is 18MB; however you can gzip it tomake it about 6MB.
FILE SHARING + TEXT CHAT USING RTCMULTICONNECTIONAdditional:
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