RTSP disabling audio

[Intergon]

Hello
To spare CPU I would like to avoid audio transcoding from RTSP cameras, which have no audio.
Is this options really disables audio in played stream ?

session.createStream({
                name: streamName,
                display: display,
                constraints: { audio:false }
})

REST find_all method shows strange things...

{
....
"name": "rtsp://xxxxxxxxx",
"published": true,
"hasVideo": false,
"hasAudio": true,
"status": "PUBLISHING",
"sdp": "v=0 o=- 2252085561 2252085561 IN IP4 0.0.0.0 s=Media Server c=IN IP4 0.0.0.0 t=0 0 a=control:* a=packetization-supported:DH a=rtppayload-supported:DH a=range:npt=now- m=video 0 RTP/AVP 96 a=control:trackID=0 a=framerate:15.000000 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=64002A;sprop-parameter-sets=Z2QAKq2EAQwgCGEAQwgCGEAQwgCEO1A8ARPywgAAAwACAAADAD0IAA==,aO48sAA= a=sendonly m=application 0 RTP/AVP 107 a=control:trackID=4 a=rtpmap:107 vnd.onvif.metadata/90000 a=sendonly",
"videoCodec": "H264",
"width":1920,
"height":1080,
"bitrate":0,
"fps":0,
....
}

.....

{
....
"name": "rtsp://xxxxxxxxx",
"published": false,
"hasVideo": true,
"hasAudio": true,
"status": "PLAYING",
"audioCodec": "mpeg4-generic",
"videoCodec": "H264",
"width":1920,
"height":1080,
"bitrate":0,
"fps":0,
"constraints": {
    "audio": false
}
...
}

Why hasVideo is false and hasAudio is true, when camera provides only Video?
Why bitrate and fps is 0 ?

Also i would like to check used bandwidth in https://wcs:8444/?action=stat, but it displays zeros:

-----Network Stats (Mbit/s)-----
global_bandwidth_in=0,000
global_bandwidth_out=0,000

 

[Max]

Set global_bandwidth_check_enabled to true to enable bandwidth stats: in WCS_HOME/conf/flashphoner.properties

global_bandwidth_check_enabled=true

 

[Max]

hasVideo is false and hasAudio is true, when camera provides only Video

Created internal issue WCS-1957, will inform when there is an update.

Bitrate and FPS are changing in real-time. The functionality for monitoring is under development.
Currently, WebRTC statistics via Web SDK can be used as described here.