Server default settings are mostly universal and need to be tuned to certain client case.
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 |
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 |
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 |
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.
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(); |
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 |
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" |
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.
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.
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 ... |
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.