Streaming video quality

Goji

New Member
Hi,

Currently I'm trying to improve the streaming video quality (resolution, fps, bitrate), I tested the demo page media devices using Chrome with setting below:
resolution: 1280 x 720
fps: 30
bitrate: 10Mbps

and the streaming quality is quite okay.
But when I test with my web call server with the same setting, the player video quality is not good at all when there is movement, also the video will become blurry. However, the video quality is slightly better if using Firefox.
FYI, I didn't change any setting after installation, and the version used is 5.0.2209.

Please advise. Thanks.

p/s: I'd tested with different quality setting, but still getting the same result.
 

Max

Administrator
Staff member
1. Try to update to the latest available version.
Code:
service webcallserver update
2. Try to change codec priority in flashphoner.properties
Example: H.264 is in priority h264,vp8
Code:
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
Example: VP8 is in priority vp8,h264
Code:
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
These settings require WCS restart:
Code:
service webcallserver restart
3. Try to check chrome://webrtc-internals charts.
Example:
webrtc-internals-send-video.jpg

4. If you are using Media Devices example, please set playback quality to 0 to avoid server-side transcoding.
webrtc-play-quality.jpg

If you still have issues please share your chrome://webrtc-internals graphs, flashphoner.properties and system CPU, RAM details.
 

Goji

New Member
Hi,
Tested with suggestion given, changed priority to VP8, server restarted, and play back quality set to 0. Video quality is better now, but still having some lagging when movement.
Here is the chrome://webrtc-internals.
upload_2017-5-26_14-51-32.png


Server info:
Window Server 2012 R2 HyperV Virtual Machine
1 Core Processor
2GB RAM (auto extend up to 10GB)
 

Attachments

Max

Administrator
Staff member
Your bitrate 50kbps is too low for 1280x720 stream.
That's why you see lagging and other issues with video.
Normal bitrate for 1280x720 is around of 2Mbps
Please see graphs from https://wcs5-eu.flashphoner.com/demo2/media-devices

bitrate-webrtc.jpg


Here you can see how WCS server limits the bitrate (red line 4Mbps)
bitrate-webrtc-limit-by-wcs.jpg

So you can see if WCS does this limit or it is internal browser limit.
Please check this limit in your test.
It looks like a network performance issue.
Did you update WCS to the latest version?
Could you provide SSH access and WCS Dashboard access, we will check this on your server.
Please email access to logs@flashphoner.com
 

Goji

New Member
Hi,

I already executed the update command, but it show me it's already latest version.
Code:
>>> You have latest version: 5.0.2209
For the bitrate, can provide more information on how should i test it to find out the limit?
Information requested already sent to logs@flashphoner.com.

Thanks.
 

Max

Administrator
Staff member
For the bitrate, can provide more information on how should i test it to find out the limit?
WCS automatically changes bitrate limit on the fly. Please see the red line.

If network is too poor and if packet lost rate is high, WCS decreases this limit.
If network is good and smooth, the limit is automatically increased.
So first of all you have to check network connection and packet lost between your browser and server.
 

Max

Administrator
Staff member
We have checked your server and reproduced the same issue.
As you can see, bitrate fell to minimum value due some network or system reasons.
Recommended actions:
1. Update to latest available build.
2. Do not use desktop machine (we are able to see you have a desktop linux with running processes gnome, firefox, etc).
3. Make sure your network is enough to pass at least 1 Mbps bitrate.
4. If you are sure that your system and network is ok, you can setup minimal bitrate in flashphoner.properties
Code:
webrtc_cc_min_bitrate=1000000
Here you set minimum bitrate to 1 Mbps
If you do that, your bitrate will be constant and if network is really poor you will have a freez because Chrome generates bandwidth greater than network can pass.

As you can see, we have the same graph from your server.
I believe if you move your server to a DC, it will work properly.

webrtc-internals.jpg
 
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