Hi there,
I been using the audio bridge for some time on its default sampling rate of 16000. I rtp_forward the audio to a janus streaming plugin endpoint (audio only), that is created like this:
var body = {
"request": "create",
"type": 'rtp',
"id": 1234,
"is_private": true,
"audio": true,
"video": false,
"audioport": 5001,
"audiopt": 96,
"audiortpmap": 'OPUS/48000/2'
}
Note the opus/48000/2.
I've heard complains that the quality is not as good as my other system (flash speex codec, 22k).
What can I do to improve the audio quality? Would setting the audio bridge to sample at 48k help? is the issue with the miss match of rates? i.e. 48k on the opus rtpmap vs 16k on the audio bridge?
Will lowering the audiortpmap to OPUS/16000/1 work ??
Most of the people connect to the streaming endpoint, only a few connect to the audio bridge (presenters). So as far as bandwidth utilization, I think I am already using 48k (stereo) !
Ideas? thoughts?
any help appreciated!