I have created an application using webRTC and tomcat 7. I am able to connect two client for video chat.one client is in sony laptop and other is in HP desktop PC(with Logitech webCam). Video is working fine but audio is generating too much echo. I have google multiple times but i didn't found any solution.
Can this echo problem be resolved at java script level using getUserMedia() or PeerConnection() APIs?
I have checked my browser versions both are chrome 28+.
Please help me out.
May this comment help you. A software recommendation seems the only "simple" solution. Or settings for Windows >= 7 as explained here.
Updated:
Default volume must be set to "0" until remote media stream starts flowing; use "setTimeout" to wait at least "1" second and then set volume back to "1".
Actually, noise occurs out of "huge-audio" bandwidth which happens as soon as "onaddstream" event fires.
If you check "audioInputLevel" via "chrome://webrtc-internals" you'll see its value something like "25000" as soon as "onaddstream" event fires; however in a few milliseconds later, "audioInputLevel" goes down between 1-and-150 and stays lower.
This doc may help: http://www.slideshare.net/MuazKhan/echo-in-webrtc-why
You can even set VoiceActivityDetection:false
as SDP-constraints to minimize peer-level echo.
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