WebRTC as RTMP re-publishing demo is working

Max

Administrator
Staff member
You have to set max bitrate in kbps
Example:
Code:
 var constraints = {
        audio: {
            deviceId: $('#audioInput').val(),
            bitrate: 128 // 128Kbps
        },
        video: {
            deviceId: $('#videoInput').val(),
            width: 1280,
            height: 720,
            bitrate: 3000, // 3Mbps
            frameRate: 30   
        }
If you set bitrate in bps you may crash server.
It was fixed in latest builds. Please do 'service webcallserver update' to fix that.
 
Thank God its working on my server now with bit rate change you mentioned.
But sent bit rate is very slow please see my screen shoots with 3 different internet connections.
Internet Connection 1
upload_2017-6-5_16-28-53.png


Internet connection 2

upload_2017-6-5_16-33-8.png


Internet Connection 3

upload_2017-6-5_16-36-17.png


Also
upload_2017-6-5_16-37-55.png
 

Max

Administrator
Staff member
Try to pass 0
Code:
video: {
deviceId: $('#videoInput').val(),
width: 1280,
height: 720,
bitrate: 0, // Do not limit.
frameRate: 30
}
If you pass 0, you do not limit max bitrate.

Regarding video size, please try to correct this using CSS.

Currently you have two options to increase bitrate
1. Use VP8 as priority codec and add CPU resources.
or
2. We need to check if we can fix webrtc_cc_min_bitrate setting.
It seems the setting does not work properly.
If we fix that, it would work with H.264.
 
Setting bitrate to 0 also not work
Already using VP8 as priority codes and also we have increase the server from small to medium on amazon EC2

@Max can you please fix webrtc_cc_min_bitrate setting.


Thanks,
Faraz
 
@Max we have increased the server on amazon from 4 GB RAM to 16 GB RAM but the bitrate is the same very lo 40K
can you please fix webrtc_cc_min_bitrate setting.
SSH access to EC2 instance is same as previous let me send you again.
 

Max

Administrator
Staff member
We have tested your server with this demo page:
https://35.163.225.113:8888/demo2/webrtc-as-rtmp-re-publishing
It works fine in our tests. Bitrate is 2Mbps.

You can also check combination of settings:
Code:
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
or
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
Code:
use_rtmp_java_client=true
or
use_rtmp_java_client=false
What is your version of browser Google Chrome?
Because we can't reproduce this with latest Chrome 58.

If nothing helps, try to relocate EC2 server or another server to your region.

Regarding webrtc_cc_min_bitrate, it won't be fixed because it is internal Chrome encoding and we can't force this.
We just recommend this to be webrtc_cc_min_bitrate=300000, for example. However final decision is up to Chrome.

So your server works fine for us and the issue is in your network or environment.
Try to test from another location. For example from a remote desktop in US.
Or try to relocate your server.
 
Hi @Max
I have checked in Firefox its working fine with good quality.
But in chrome its quality is very bad i am using chrome Version 59.0.3071.86 (Official Build) (64-bit)
And in IE its not working plz see the error below in console of screen shoot.
upload_2017-6-7_14-15-53.png


Also for opera its not working
Please see below screen shoot.

upload_2017-6-7_15-23-26.png
 

Max

Administrator
Staff member
We have tested:
  • Chrome 59
  • Opera 45
  • IE 11
What is your versions?

Regarding Chrome,
Perhaps you have some plugins or firewall or network restrictions. Please test from another Chrome browser, another network.

Regarding Opera,
It works for me in Opera 45. Try to update to the latest version.

Regarding IE
It works for me. Try to test with IE 11.
 

Max

Administrator
Staff member
Regarding Chrome and Opera it looks like the same network issue.
If you provide permanent TeamViewer access to a testing desktop PC inside your network, we will able to test this and check why Chrome stream has poor quality.
You can send details to logs@flashphoner.com. We need such permanent access for a few hours.
 

Max

Administrator
Staff member
I mean access details to a remote desktop over Teamviewer or AnyDesk.
If you provide permanent TeamViewer access to a testing desktop PC inside your network, we will able to test this and check why Chrome stream has poor quality.
 

Max

Administrator
Staff member
We have checked a stream over teamviewer.
As an assumption you web cam hardware is not fully compatible with Chrome 59, because it provides low FPS.
And this can be a cause why bitrate is low.
1. Try to use another cam, for example USB.
2. Try to use virtual cam, for example manycam and check what FPS do you have on the output.
You can see input FPS on this graph:
input-FPS.jpg
 
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