2

I tested an audio and video conversation over the webRTC, using Chrome on 2 computers from two different networks. The quality of the video is perfect and does not crash, but the sound hangs all the time, can hear my own voice and does not have quality.

How can I improve?

Code:

var haveGum = navigator.mediaDevices.getUserMedia({fake:true, video:true})
  .then(stream => streams[1] = stream)
  .then(() => navigator.mediaDevices.getUserMedia({ video: true, audio: true }))
  .then(stream => v1.srcObject = streams[0] = stream)

I found this answer from 2014, the audio was very good. But still with cuts and sometimes the voice becomes "robotic" and it is not possible to understand what is spoken.

Community
  • 1
  • 1
luisdemarchi
  • 1,402
  • 19
  • 29
  • just check the bandwidth of audio it will bi higher then the net bandwidth connection – aditya shrivastava Jan 28 '17 at 11:11
  • @adityashrivastava There is no way to know the network capacity of my users. How can I resolve this? In research I found that Opus was considered one of the best codecs because it was considered smart at that point. – luisdemarchi Jan 29 '17 at 12:49
  • I agree with you and even i use opus codec in my project and its working great so far so i am suggesting you to re-tune your bandwidth of audio and video put it on minimum point and then try it will help you – aditya shrivastava Jan 29 '17 at 18:12

0 Answers0