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Questions tagged [openwebrtcdemo]

10 questions
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Is there a formula for rating WebRTC audio quality as Excellent, Good, Fair, or Poor?

I have been able to get various stats (Jitter, RTT, Packet lost, etc) of WebRTC audio call using RTCPeerConnection.getStats() API. I need to rate the overall call quality as Excellent, Good, Fair, or Poor. Is there a formula that uses WebRTC stats…
Girish MC
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How to calculate Mean Opinion Score (MOS) to measure the quality of WebRTC calls without knowing the Round Trip Time (RTT) of the inbound RTP stream?

WebRTC inbound RTP stream (inbound-rtp) does not return Round Trip Time (RTT), but it is required to to calculate Mean Opinion Score (MOS). Any suggestions? Is is safe to assume asymmetric latency and take the RTT for inbound-rtp same as RTT of…
Girish MC
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Create org.webrtc.JavaI420Buffer from YUV420 ByteBuffer

Hi I am working on a live streaming solution where I need to edit the local video frame and send it to the connected peer, for this task I am…
Anuj J Pandey
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WebRtc Audio transferring stopped after few seconds

From last few days I'm working on WebRtc I setup everything at server side and android side also, but when I call via an Android device, after few seconds audio transferring stopped while it's working fine web to web, so what am I doing wrong at…
Nilesh Rathore
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Add Track to media stream

Is it possible to add or remove audio track to a stream after the stream connect to the pearConnection? I was able to do it before ( in getUserMedia ) But it doesn't work after the stream connect to the peerConnection Thanks you
MoShe
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Web RTC Server is running in localhost only

I've configured Web RTC sever in my PC. It's working with http://localhost:8080 But this server is not working with the IP address in another machines. I've referred Web RTC Server with this URL.
Mr. N.V.Rao
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openwebrtc demo is not working in Chrome

Chrome supports WebRTC. But openwebrtc I am not able to run http://demo.openwebrtc.org:38080/ while apprtc I am able to run https://apprtc.appspot.com/ so why this problem ? so what is the difference between openwebrtc and apprtc ? is there any…
Durai Amuthan.H
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Web RTC between two different web clients not working

As per our web RTC requirements, there are two different client Player (Players the screen shared by capture client) Capture (Share Screen) The two web clients communicate and exchange offers and ICE candidates using WebSocket. In Chrome [Version…
Sowvik Roy
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webRTC Video and Audio Call in Flutter

I am new to Flutter App and I have to create a WEBRTC based video and audio calling app for both IOS and Android without using TURN Server with FCM for creating a connection between 2 users. Could anyone please help me out how can we achieve this…
Himanshu Dhingra
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Uncaught Error: This Firebase project requires import into Google Cloud Platform to resume service

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-…
Nitesh S Chauhan
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