Server default settings are mostly universal and need to be tuned to certain client case.
REST client tuning
When REST hooks are used, on every WCS server action (establishing client connection, publishing and playing a stream, making a SIP call etc) HTTP REST connection to backend server is established. With a large number of simultaneously publishing clients or subscribers, with the default WCS settings it is possible to exhaust the WCS REST client thread pool, that is lead to deadlocks. Then, server stops to publish and play streams.
By default, a maximum number of simultaneous REST connections is set to 200 with the following parameter in flashphoner.properties file
rest_max_connections=200
To escape thred poolexhausting and deadlocks this value should be reduced, for example
rest_max_connections=20
If REST hooks are not used, REST client can be disabled with the following parameter
disable_rest_requests=true
Excessive logging supression
When REST hooks are used, REST client operations, EchoApp default backend operations and REST API server operations are written to WCS core logs. That leads to large number of entries in the log file and, therefore, inceases the server load. The excessive logging may be decreased if necessary using the following parameters in log4j.properties file:
log4j.logger.RestClient=WARN log4j.logger.EchoApp=WARN log4j.logger.RestApiRouter=WARN
UDP tuning
Streaming mediadata are transferred with UDP packets. Those packets can be dropped, for example if server does not have enough time to parse packet queue, that leads to picture quality loss and freezes. To escape this it is necessary to tune UDP sockets buffers with the following settings in flashphoner.properties file
rtp_receive_buffer_size=131072 rtp_send_buffer_size =131072
and to tune system queues with command
ip link set txqueuelen 2000 dev eth0
To diagnose UDP problem, it is necessary to track UDP packets dropping with command
dropwatch -l kas >start
UDP socket buffers tuning on system level
If UDP is used to publish or play high bitrate streams (for instance, as CDN internal transport), UDP socket buffers tuning may be required
sudo sysctl -w net.core.rmem_max=26214400 sudo sysctl -w net.core.rmem_default=26214400
In this case the server performance should be enough to handle all the traffic packets. Otherwise, a translation quality will drop (a freezes will appear), and CPU will become a bottleneck.
Channel load optimization
Users' playback picture quality depends on bitrate: the higher the bitrate, the higher the quality. However, the higher the bitrate, the higher data transfer channel load and, if the bandwidth between the server and clients is limited, there is a possibility that the channel will be fully loaded. This leads the bitrate dropping and a sharp decline in quality.
In this regard, it is necessary to limit the bitrate to ensure sufficient picture quality with an acceptable channel load.
Publisher bitrate limiting
To reduce the load to the channel from publisher to server, maximum and minimum bitrate values in kbps may be set in publisher script with JavaScript API
session.createStream({
name: streamName,
display: localVideo,
constraints: {
video: {
minBitrate: 500
maxBitrate: 1000
}
}
...
}).publish();
Server bitrate limiting
Minimum and maximum bitrate values in bps on server may be set with the following parameters in flashphoner.properties file
webrtc_cc_min_bitrate=500000 webrtc_cc_max_bitrate=1000000
To exclude fast bitrate rise bu=y browser, the following parameter should be set
webrtc_cc2_twcc=false
Stream decoding on demand only must be switched on to reduce server load:
streaming_video_decoder_fast_start=false
Changing dynamic ports range in Linux
Dynamic or ephemeral port is a temporary port that is opened when establishing IP-connection from certain range of TCP/IP stack. Many Linux kernel versions use ports range 32768 — 61000 as dymanic ports. Enter the following command to check what range is used on server
sysctl net.ipv4.ip_local_port_range
If this range overlaps with WCS standard ports, it should be changed with the following command
sysctl -w net.ipv4.ip_local_port_range="59999 63000"
Adjusting the maximum number of opened files
Legacy settings (before build 5.2.762)
In the launch script webcallserver that is in subfolder bin in WCS home folder, for example
/usr/local/FlashphonerWebCallServer/bin/webcallserver
in start() function the maximum number of opened files is set
function start() {
...
echo -n $"$PRODUCT: starting"
ulimit -n 20000
if [[ "$1" == "standalone" ]]; then
...
fi
...
}
By default, this value is set to 20000, but it may be increased if necessary, following the limitations of the operating system used.
Using environment variable (since build 5.2.762)
Since build 5.2.762, maximum opened files limit can be set using the following environment variable
WCS_FD_LIMIT=20000
in setenv.sh file. When updating WCS from previous builds, this variable should be added to setenv.sh manually, for example
export WCS_FD_LIMIT=100000
Unlike the webcallserver startup script, the setenv.sh file is not overwritten on subsequent updates, therefore it is not necessary to restore this setting after every update.
Using service parameter while launching from non-root user (since build 5.2.801)
Since build 5.2.801, WCS is launching from 'flashphoner' user for better security. In this case, maximum opened files limit can be set using service parameters
sudo nano /etc/systemd/system/webcallserver.service
Maximum opened files limit is set with LimitNOFILE parameter, for example
[Service] User=flashphoner Group=flashphoner LimitNOFILE=100000 ...
Internal command to change file descriptors limit
Since build 5.2.1255 the following command can be used to set file descriptors limit:
sudo ./webcallserver set-fd-limit 100000
WCS will be stopped before settings changing and will be automatically started after settings changing to apply them.
If a new value is less than the default one (20000), erroe message will be displayed, and changes will not be applied.
Traffic encryption in a separate thread for each client session
By default, one CPU thread encrypts medai traffic for all the client sessions. This leads to one CPU core overload by such thread, espacyally on low-power servers, for big subscribers amount. Then, server can not send mediapackets to all subscribers, and streams viewed are degrading, FPS lowering and freezing.
To distribute the load evenly across the CPU cores, it is necessary to enable traffic encryption in a separate thread for each client session with the following parameters
rtp_paced_sender=true rtp_paced_sender_initial_rate=200000 rtp_paced_sender_increase_interval=50 rtp_paced_sender_k_up=0.9
and restart WCS.
Stream distribution optimization
A stream playback quality may drop when a number of subscribers are viewing it simutlaneouly (from 100 and more): low FPS, freezes. However, server capacity and channel bandwidth may be enough. In this case it is recommended to enable multithreaded stream distribution to subscribers using the following parameter
streaming_distributor_subgroup_enabled=true
In this case, audio and video client sessions are distributed by groups.
Maximum number of video sessions per group can be set with the following parameter
streaming_distributor_subgroup_size=50
Maximum number of audio sessions per group can be set with the following parameter
streaming_distributor_audio_subgroup_size=500
Frame queue size per group and maximum frame waiting time (in milliseconds) are set by the following parameters
streaming_distributor_subgroup_queue_size=300 streaming_distributor_subgroup_queue_max_waiting_time=5000
for video and
streaming_distributor_audio_subgroup_queue_size=300 streaming_distributor_audio_subgroup_queue_max_waiting_time=5000
for audio sessions respectively.