You said can about 1000 users right?
We tested 1 publisher and 1000 subscribers.
Publisher is Chrome 62 / WebRTC / Opus+VP8
Subscribers: synthetic subscribers pulling WebRTC traffic from server
This is tuning instructions:
WCS Tuning for build 2556
1. Update to the latest available build
Code:
service webcallserver update 2556
2. Configure WCS heap and GC in WCS_HOME/conf/wcs-core.properties. Make sure you have enough free memory on your server, i.e. 16G RAM.
Code:
-Xms8g -Xmx8g
-XX:+UseConcMarkSweepGC -XX:NewSize=1024m
3. Configure UDP buffers in WCS_HOME/conf/flashphoner.properties
Code:
rtp_receive_buffer_size=13107200
rtp_send_buffer_size =13107200
Values must be not greater than /proc/sys/net/core/rmem_max and /proc/sys/net/core/wmem_max respectively
4. Set VP8 priority and limit incoming bitrate
Code:
webrtc_cc_min_bitrate =400000
webrtc_cc_max_bitrate =500000
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
5. Extend queue on linux
Code:
ip link set txqueuelen 2000 dev eth0
Monitoring of lost UDP packets
However it won't help for your case because you have obvious memory leak.
Please describe in detail how does your case work. How do you stream. Please provide streaming scheme, logs, etc.
Please share WCS_HOME/logs and WCS_HOME/conf for last 24 hours to
logs@flashphoner.com