flashphoner.properties config file¶
Main server settings
aac_bitrate¶
Default:
128000
Type:
Integer
Need restart: No
Description:
AAC encoding bitrate
aac_encoder_sync_drop_threshold¶
Default:
1000
Type:
Long
Need restart: Yes
Description:
JitterBuffer will be reset upon reaching this number of dropped sync packets
aac_test_start_codec¶
Default:
20
Type:
Integer
Need restart: Yes
Description:
AAC test codecs count
aac_test_transcode_iterations¶
Default:
1000
Type:
Integer
Need restart: Yes
Description:
AAC test interval
add_register_auth_headers¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then add Authorization header in REGISTER request when first registering.
Some SIP servers are configured so that they do not accept such requests. In that case this setting should be set to false''
agent_set_local_session_debug¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable local agent session debug
agent_use_subscriber_listener¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, does the agent have to wait for the subscriber
allow_domains¶
Default:
null
Type:
String
Need restart: No
Description:
If set, then WebSocket connections from these domains only will be allowed
allow_domains_allow_empty_origin¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If set, then WebSocket connections with empty origin will be allowed
allow_outside_codecs¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If false, dont add outside (browser) codecs to SDP'
allow_reinvite_in_hold_state¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, process re-INVITE requests within the session even if the call is in hold state
allow_stream_names¶
Default:
*
Type:
String
Need restart: No
Description:
If set, then client stream name from these stream names only will be allowed
answer_with_one_codec_in_sdp¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, answer with one codec only in SDP.
It can be useful in cases of improper operation of SIP equipment from some vendors, which incorrectly interpret two or more codecs in SDP during a connection establishment in Offer-Answer model
audio_force_sync_timeout¶
Default:
100
Type:
Integer
Need restart: No
Description:
Waiting for RTCP sync packet on this interval in ms, for audio
audio_frames_per_packet¶
Default:
6
Type:
Integer
Need restart: No
Description:
RTMFP. Audio will be flushed after this number of audio frames in the packet is reached
audio_incoming_min_buffer_size¶
Default:
2
Type:
Integer
Need restart: No
Description:
Waiting for RTCP sync packet at least on this interval in packets, for audio
audio_mixer_max_delay¶
Default:
300
Type:
Integer
Need restart: No
Description:
Audio mixer max delay in milliseconds
audio_mixer_output_channels¶
Default:
1
Type:
Integer
Need restart: No
Description:
Audio mixer output channels, default is 1 channel, mono
audio_mixer_output_codec¶
Default:
opus
Type:
String
Need restart: No
Description:
Audio mixer output codec (multiple codecs not allowed)
audio_mixer_output_sample_rate¶
Default:
48000
Type:
Integer
Need restart: No
Description:
Audio mixer output samle rate in Hz
audio_reliable¶
Default:
partial
Type:
on
partial
off
Need restart: No
Description:
RTMFP, reliability for audio
audio_stream_mode_udp¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Not in use
auto_login_url¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Not in use
av_paced_sender¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable paced sender for output stream. EXPERIMENTAL
av_paced_sender_max_buffer_size¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Max size of audio or video buffer. Once size is reached buffers are cleared
avatar_dir¶
Default:
avatar
Type:
String
Need restart: Yes
Description:
Avatar base folder
avcc_buffer_wait_frames_count¶
Default:
5
Type:
Integer
Need restart: No
Description:
Wait until the buffer is filled with frames
avcc_send_buffer_size¶
Default:
500000
Type:
Integer
Need restart: No
Description:
Avcc send buffer size in bytes
aws_s3_credentials¶
Default:
null
Type:
String
Need restart: Yes
Description:
AWS s3 credentials: region;accessKey;secretKey
balance_header¶
Default:
balance
Type:
String
Need restart: No
Description:
This SIP header will be sent to client as a balance
burst_avoidance_count¶
Default:
100
Type:
String
Need restart: No
Description:
Burst avoidance count
busy_state¶
Default:
null
Type:
String
Need restart: No
Description:
Used if send_busy_when_on_call=true, and an incoming call comes during another established call. Caller will receive this status.
If true, the value of ip_local= setting will be used in SIP and SDP. If false, then the value of ip= will be used
call_record_listener¶
Default:
com.flashphoner.server.client.DefaultCallRecordListener
Type:
String
Need restart: No
Description:
Full name of Java class that implements interface ICallRecordListener
public interface ICallRecordListener {
void onRecordReport(RecordReport recordReport);
}
case_sensitive_auth_match¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If false, ignore case on url auth
cdn_advertise_pulled¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, pulls CDN advertise
cdn_advertise_streams_by_kframe¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Advertise stream to CDN by key frame
cdn_allowed_ips¶
Default:
Type:
ArrayList
Need restart: Yes
Description:
Comma-separated list of allowed IPs or networks for CDN.
Example: 88.198.98.1/24, 88.198.99.219
cdn_auto_pull¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Pull CDN stream once it becomes available
cdn_connection_quality_calculation_timeout_ms¶
Default:
10000
Type:
Integer
Need restart: Yes
Description:
Connection quality calculation update timeout ms
cdn_connection_tcp_no_delay¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Turns on tcp no delay for CDN signalling connections
cdn_controller_request_timeout¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Timeout for requests sent to CDN controller
cdn_controller_response_cache_expire¶
Default:
10000
Type:
Integer
Need restart: Yes
Description:
TTL for cached records received from CDN controller
cdn_dtls_force_version_0¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Force DTLS version 1.0 inside cdn
cdn_enabled¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, enables CDN
cdn_force_version¶
Default:
2.0
Type:
String
Need restart: Yes
Description:
Force to set CDN version
cdn_group_origin_to_transcoder_relation¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Use CDN group indications to relate origin to transcoder rather than transcoder to edge
cdn_groups¶
Default:
Type:
ArrayList
Need restart: Yes
Description:
CDN groups for this node
cdn_inbound_auditor_interval¶
Default:
1000
Type:
Integer
Need restart: Yes
Description:
Time interval to check inbound connections, in milliseconds
cdn_inbound_connection_unanswered_pings¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
Inbound connection unanswered pings number.
Connection considered to be lost when this number is reached
cdn_inbound_ws_read_socket_timeout¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Enable WebSocket read timeout for inbound cdn connactions
cdn_inbound_ws_read_socket_timeout_sec¶
Default:
60
Type:
Integer
Need restart: Yes
Description:
WebSocket read timeout value (if enabled) for inbound cdn connections
cdn_ip¶
Default:
null
Type:
String
Need restart: Yes
Description:
CDN node IP address (or domain name when cdn_nodes_resolve_ip=true)
cdn_load_interval¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
load interval
cdn_load_node¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Turn on cdn load behaviour
cdn_load_pool_size¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
load pool
cdn_load_pool_size_change_interval¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
Change pool size every interval
cdn_load_pool_size_lower_threshold¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
Lower threshold for pool size change
cdn_load_pool_size_upper_threshold¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
Upper threshold for pool size change
cdn_load_proto_pull¶
Default:
websocket
Type:
String
Need restart: Yes
Description:
CDN load protocol stream
cdn_load_reserved_stream¶
Default:
Type:
String
Need restart: Yes
Description:
CDN load reserved stream
cdn_load_step¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
load step
cdn_load_unique_streams¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Pull only unique streams
cdn_load_use_profile_name¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Put profile name in stream name. Use if entry point is edge
cdn_load_use_profiles¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Pull with profiles
cdn_node_load_average_threshold¶
Default:
1.0
Type:
Double
Need restart: Yes
Description:
If threshold reached node will advertise it's state as GROUP_CONNECTIONS
cdn_nodes_acl_refresh_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN node acl list, in milliseconds
cdn_nodes_auditor_interval¶
Default:
1000
Type:
Integer
Need restart: Yes
Description:
Time interval to check available CDN nodes, in milliseconds
cdn_nodes_group_refresh_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN node group, in milliseconds
cdn_nodes_on_single_server¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, cdn nodes can be located on single server
cdn_nodes_resolve_ip¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, resolve CDN node domain names to IP addresses
cdn_nodes_role_refresh_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN node role, in milliseconds
cdn_nodes_route_refresh_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN routes, in milliseconds
cdn_nodes_state_refresh_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN node state, in milliseconds
cdn_nodes_timeout¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
CDN nodes timeout in seconds. -1 means nodeTimeout disabled
cdn_nodes_version_refresh_interval¶
Default:
90000
Type:
Integer
Need restart: Yes
Description:
Time interval to refresh CDN node version, in milliseconds
cdn_origin_allowed_to_transcode¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
In case no transcoders left node will request transcoding profile from origin
cdn_origin_to_origin_route_propagation¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, origin sends routes to other origins
cdn_outbound_auditor_interval¶
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Time interval to check outbound connections, in milliseconds
cdn_outbound_connection_timeout¶
Default:
6000
Type:
Integer
Need restart: Yes
Description:
Outbound connection timeout, in milliseconds
cdn_outbound_ws_read_socket_timeout¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Enable WebSocket read timeout for outbound cdn connactions
cdn_outbound_ws_read_socket_timeout_sec¶
Default:
60
Type:
Integer
Need restart: Yes
Description:
WebSocket read timeout value (if enabled) for outbound cdn connections
cdn_point_of_entry¶
Default:
Type:
String
Need restart: Yes
Description:
CDN point of entry node IP address (or domain name when cdn_nodes_resolve_ip=true)
cdn_port¶
Default:
8084
Type:
Integer
Need restart: Yes
Description:
CDN server port
cdn_role¶
Default:
EDGE
Type:
ORIGIN
EDGE
TRANSCODER
CONTROLLER
Need restart: Yes
Description:
CDN role:
origin - the source of media streams for other CDN nodes
edge (default) pulls media streams from origin CDN node(s)
cdn_role_strict¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, sets restrictions on publishing streams by role
cdn_role_strict_stream_name¶
Default:
null
Type:
String
Need restart: No
Description:
Reserved name for setting cdn_role_strict bypass
cdn_skip_pulled_streams¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, skip pulled streams
cdn_ssl¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, enables SSL
cdn_standalone¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, start server in CDN standalone mode, streaming will not available
cdn_strict_transcoding_boundaries¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Prevent transcoding to the same or higher resolution of original stream by placing resolution boundary
cdn_strict_transcoding_throws_exception¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Whether to fail play or substitute requested profile with original stream if profile hit the strict transcoding boundary
cdn_test_enabled¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Turn on cdn tests
cdn_test_interval¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
test interval
cdn_test_max_subscribers_for_stream¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
Max subscribers for each CDN stream. Edge-only setting
cdn_test_pool_size¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
test pool
cdn_test_step¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
test step
cdn_transcoder_degraded_streams_threshold¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
If threshold reached node will advertise it's state as GROUP_CONNECTIONS. Threshold is the percent of degraded streams
cdn_transcoder_for_new_connects_expire¶
Default:
10000
Type:
Integer
Need restart: Yes
Description:
CDN transcoder cache expire for new stream requests
cdn_transcoder_threshold_state¶
Default:
GROUP_CONNECTIONS_ALLOWED
Type:
UNKNOWN
PASSIVE
GROUP_CONNECTIONS_ALLOWED
CONNECTIONS_ALLOWED
NEW_STREAMS_ALLOWED
Need restart: Yes
Description:
If threshold reached node will change state to provided value
cdn_transcoder_video_decoders_load_threshold¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
If decoders load threshold reached node will advertise it's state as GROUP_CONNECTIONS. Threshold is the sum of decoderWidthdecoderHeightdecoderFPS
cdn_transcoder_video_encoders_load_threshold¶
Default:
-1
Type:
Integer
Need restart: Yes
Description:
If encoders load threshold reached node will advertise it's state as GROUP_CONNECTIONS. Threshold is the sum of encoderWidthencoderHeightencoderFPS
cdn_transcoder_video_encoders_threshold¶
Default:
10000
Type:
Integer
Need restart: Yes
Description:
If threshold reached node will advertise it's state as GROUP_CONNECTIONS
cdn_transport¶
Default:
udp
Type:
String
Need restart: Yes
Description:
CDN internal transport (udp, tcp, srt)
chat_listener¶
Default:
null
Type:
String
Need restart: No
Description:
Full name of Java class that implements interface IChatListener
public interface IChatListener {
void onMessage(InstantMessage message);
}
check_receiver_origin¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, check origin of RTCP packet and discard if unknown
cli.address¶
Default:
localhost
Type:
String
Need restart: Yes
Description:
Listening address for CLI SSH server
cli_auth_keys¶
Default:
/usr/local/FlashphonerWebCallServer/.ssh/authorized_keys
Type:
String
Need restart: Yes
Description:
CLI client auth keys file path
cli_auth_server_keys¶
Default:
/usr/local/FlashphonerWebCallServer/conf/cli-hostkey.pem
Type:
String
Need restart: Yes
Description:
CLI host identification key file path
cli_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enables CLI
cli_port¶
Default:
2001
Type:
Integer
Need restart: Yes
Description:
CLI server port
cli_v2.address¶
Default:
localhost
Type:
String
Need restart: Yes
Description:
Listening address for CLI_V2 SSH server
cli_v2_port¶
Default:
2002
Type:
Integer
Need restart: Yes
Description:
CLI V2 server port
client_acl_property_name¶
Default:
aclAuth
Type:
String
Need restart: Yes
Description:
Access list identifier property key that server should look for in custom config when client connects
client_appkey_property_name¶
Default:
appKey
Type:
String
Need restart: Yes
Description:
Property name to get application name used in authentication
client_dump_level¶
Default:
0
Type:
Integer
Need restart: No
Description:
If tcpdump is installed in the system, it will be started and will capture client session traffic:
0 - do not capture traffic
1 - capture SIP traffic only
2 - capture SIP and media traffic: ICE, RTP, SRTP, RTCP, WebRTC
client_handler¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: Yes
Description:
Not in use
client_log_exclude¶
Default:
Type:
String
Need restart: No
Description:
Do not log events listed
client_log_force_debug¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable client logs for every newly connected client for a period of time specified by client_log_force_debug_timeout regardless of other settings
client_log_force_debug_timeout¶
Default:
60
Type:
Integer
Need restart: No
Description:
Timeout after which client logs will be turned off
client_log_level¶
Default:
INFO
Type:
String
Need restart: No
Description:
Log4j level.
Logs related to client sessions will be recorded on the server in /usr/local/FlashphonerWebCallServer/logs/client_logs directory with the set logging level.
Will work only if enable_extended_logging=true
client_mode¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, the value of ip_local= setting will be used in SIP and SDP. If false, then the value of ip= will be used
client_subscribe_streams_max¶
Default:
10
Type:
Integer
Need restart: No
Description:
Max subscribe streams allowed for client
client_timeout¶
Default:
3600000
Type:
Integer
Need restart: No
Description:
Client timeout value in milliseconds
client_timeout_check_interval¶
Default:
300000
Type:
Integer
Need restart: No
Description:
Client timeout interval value in milliseconds
codec_terminator_timeout¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Codec release timeout, in seconds.
Default: If codec has been marked as terminated, and if no new packets went through this codec in 5 seconds, the codec will be released
codecs¶
Default:
null
Type:
String
Need restart: No
Description:
List of supported codecs ordered by priority
codecs_exclude_cdn¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of codecs which will not be used for CDN
codecs_exclude_sfu¶
Default:
alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h265,flv,mpv,vp9,h264
Type:
String
Need restart: No
Description:
Comma-separated list of codecs which will not be used for SIP as RTMP case
codecs_exclude_sip¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of codecs which will not be used for SIP phone cases
codecs_exclude_sip_rtmp¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of codecs which will not be used for SIP as RTMP case
codecs_exclude_streaming¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of codecs which will not be used for streaming
compact_media_port_usage¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Use odd media ports for transferring data (requires rtcpMux)
complex_test_config¶
Default:
Type:
String
Need restart: No
Description:
Complex transcoder test configuration
complex_test_decode¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable decoding during complex transcoding test
complex_test_fps¶
Default:
15
Type:
Integer
Need restart: Yes
Description:
Complex transcoder test FPS
complex_test_replay¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
Complex transcoder test repeats count
complex_test_thread¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
Complex transcoder test threads count
core_standalone_web_dir¶
Default:
null
Type:
String
Need restart: No
Description:
Web directory for standalone mode
cost_header¶
Default:
cost
Type:
String
Need restart: No
Description:
This SIP header will be sent to client as a call cost
cps_client¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of IPs or networks with corresponding CPS limits.
Example: 192.168.88.2:10,192.168.88.0/16:15
cps_interval¶
Default:
1000
Type:
Long
Need restart: No
Description:
Time window for measuring CPS, in milliseconds
cps_node¶
Default:
2147483647
Type:
Integer
Need restart: No
Description:
Global CPS limitation for node
cpu_load_avg_size¶
Default:
20
Type:
Integer
Need restart: Yes
Description:
CPU load average size
cpu_load_refresh¶
Default:
50
Type:
Integer
Need restart: Yes
Description:
CPU load refresh rate
cpu_load_reject¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, reject streams when CPU load exceeds treshold
cpu_load_threshold¶
Default:
80
Type:
Integer
Need restart: Yes
Description:
CPU load treshold
cpu_load_window¶
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Timeslice to estimate CPU load
custom_ice_agent¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use custom ICE agent
custom_stats_script¶
Default:
Type:
String
Need restart: No
Description:
Script can be used to provide custom stat params with action=stat request
custom_watermark_filename¶
Default:
null
Type:
String
Need restart: No
Description:
Sets custom PNG file for watermark. The file should be placed in /usr/local/FlashphonerWebCallServer/conf directory. The feature is not available for Trial license
custom_watermark_mix¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enables watermark mixing for alpha-layer
data_packet_decoder_fire_null_messages¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, pass special data packet up the RTP process chain when original received data failed to decode
decoded_frame_interceptor¶
Default:
null
Type:
String
Need restart: No
Description:
Full name of Java class that implements interface IDecodedFrameInterceptor. This class should be wrapped to .jar file placed in /usr/local/FlashphonerWebCallServer/lib directory
decoded_pcm_interceptor¶
Default:
null
Type:
String
Need restart: No
Description:
Full name of Java class that implements interface IDecodedPcmInterceptor. This class should be wrapped to .jar file placed in /usr/local/FlashphonerWebCallServer/lib directory
decoder_binary_log_enable¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Binary log decoder
decoder_binary_log_size¶
Default:
5
Type:
Integer
Need restart: Yes
Description:
Binary log decoder size
decoder_buffer_pool¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Enable buffer pool usage during video decoding
decoder_buffer_pool_stats¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable buffer pool stats, might slow down video transcoding
decoder_mode¶
Default:
JNI
Type:
QUEUE
JNI
Need restart: No
Description:
Decoder mode
decoder_priority¶
Default:
FF,OPENH264
Type:
String
Need restart: No
Description:
Decoder priority
decoder_stat_log¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable decoder statistics logging
default_packetization_mode¶
Default:
0
Type:
Integer
Need restart: No
Description:
Packetization mode default value if incoming SDP doesn't contains packetization_mode parameter.
default_sdp_state¶
Default:
sendrecv
Type:
String
Need restart: No
Description:
If SDP from SIP side comes without sendrecv, recvonly, or sendonly attribute, then it is assumed that the attribute defined in this setting was received
degraded_streams_threshold¶
Default:
20
Type:
Integer
Need restart: Yes
Description:
Degraded streams threshold
degraded_streams_window¶
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Timeslice to estimate stream degradation
delta_threshold¶
Default:
100
Type:
Integer
Need restart: No
Description:
RTMFP. If delta between UDP media packets is greater than the threshold, it will be reported
depacketizer_dump_dir¶
Default:
depacketizer_dump_dir
Type:
String
Need restart: Yes
Description:
H264 RTP stream dump base folder
detect_flash_2_flash_calls¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, WCS server will use an RTP extension header in RTP packets, which can be used for designation of WCSs own streams, even if they are traced through third-party PBX, e.g. Asterisk'
disable_drop_aac_frame¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, disables dropping AAC frames
disable_manager_rmi¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, disables RMI communications between WCS Core and WCS Manager
disable_rest_auth¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, disables authorization in rest api
disable_rest_requests¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, disables Rest requests to application
disable_rtc_ata¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
By default WCS server will try to avoid transcoding and send its supported codec to the other side, even if codecs will be chosen asymmetrically. This behaviour is called Avoid Transcoding Algorithm (ATA).
This option defines comma-separated list of SIP User Agents, for which the algorithm will be disabled. It means that if codecs are asymmetrical, then for these User Agents transcoding will proceed
disable_rtc_avoid_transcoding_alg¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, disables RTC ATA (see above)
disable_streaming_proxy¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, disable proxy and enable transcoding for all streams. For debug only
disable_streaming_proxy_aac¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If false, enable AAC proxying
dns_test_enable¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Enable dns test statistic
dns_test_name¶
Default:
Type:
String
Need restart: Yes
Description:
DNS name for time resolved statistic
domain¶
Default:
null
Type:
String
Need restart: No
Description:
SIP domain. If this parameter is set, it will redefine values that were transmitted during connection
dtls0_ua_match_substring¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, DTLS User-Agent matching will be by substring. Ex: Chrome/70.0
dtls_close_socket_after_tries¶
Default:
10
Type:
Integer
Need restart: No
Description:
Disable / enable DTLS session termination after the specified number of connection attempts.
By default, DTLS session will not be terminated: dtls_close_socket_after_tries=0
dtls_force_version_0¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Force DTLS version 1.0
dtls_message_timeout¶
Default:
15
Type:
Integer
Need restart: No
Description:
DTLS handshake timeout in seconds, must be set to a non-zero value
dtls_socket_timeout_ms¶
Default:
1000
Type:
Integer
Need restart: No
Description:
DTLS socket SO_TIMEOUT in milliseconds. With this option set to a non-zero value, a read() call on the InputStream associated with this Socket will block for only this amount of time
dtls_use_socket_timeout¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable DTLS socket SO_TIMEOUT
dtmf¶
Default:
null
Type:
String
Need restart: No
Description:
This type will be used if DTMF type (INFO, INFO_RELAY, RFC2833) was not specified when DTMF was sent
dump_avcc_relay¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, write outgoing MSE packets to file. That file can afterwards be processed as VoD at client side. Used for MSE development tests
enable_candidate_harvester¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, gather ICE candidates using external STUN server
enable_empty_shift_writer¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable empty shift writer for conference
enable_extended_logging¶
Default:
true
Type:
Boolean
Need restart: No
Description:
When extended logging is enabled, these settings are used:
- client_log_level
- client_dump_level
Then logs for all client sessions are saved in /usr/local/FlashphonerWebCallServer/logs/client_logs directory
enable_flight_recorder¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable flight recorder
enable_flight_recorder_test¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable flight recorder test
enable_local_videochat¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Not in use
enable_network_address_cache¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Enable cache for resolved addresses.
enable_new_client_logger¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable new client logger
enable_rtc_video_generator¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Designed to avoid video negotiation issue in SIP cases. If true, generated video will be sent once session is established. It is a workaround and should not be used in normal situation
enable_sip_stack_thread_audit¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable audit of SIP stack
enable_sync_time_normalizer¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, then enable sync time normalizer
encode_record_name¶
Default:
null
Type:
String
Need restart: Yes
Description:
Encode record name setting
encoder_buffer_length_sec¶
Default:
1
Type:
Integer
Need restart: No
Description:
Encoding buffer for audio, in seconds
encoder_default_video_resolution¶
Default:
640x480
Type:
String
Need restart: No
Description:
encoder_default_video_resolution
encoder_mode¶
Default:
JNI
Type:
QUEUE
JNI
Need restart: No
Description:
Encoder mode
encoder_priority¶
Default:
FF,OPENH264
Type:
String
Need restart: No
Description:
Encoder priority
encoder_stat_log¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable encoder statistics logging
event_scanner_cached_pool¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use event scanner cached pool
event_scanner_pool_size¶
Default:
10
Type:
Integer
Need restart: No
Description:
Event scanner pool size
exclude_record_name_characters¶
Default:
null
Type:
String
Need restart: Yes
Description:
Exclude characters from record name
fetch_caller_from_pai¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then for an incoming call the caller should be taken from PAI (P-Asserted-Identity) header. If that header is empty, the caller will be displayed as Unknown/Anonymous
fetch_caller_from_pai_set_from_if_empty¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, fetch caller from PAI from' when caller is empty'
file_recorder_error_interval¶
Default:
60
Type:
Integer
Need restart: No
Description:
Error counter's interval in minutes
file_recorder_max_errors_per_interval¶
Default:
3
Type:
Integer
Need restart: No
Description:
Max errors per interval
file_recorder_min_space¶
Default:
1g
Type:
String
Need restart: No
Description:
Minimum available disk space for recording in GiB(G|g), MiB(M|m) or KiB(K|k). By default, GiB is used if no suffix specified
file_recorder_thread_pool_max_size¶
Default:
4
Type:
Integer
Need restart: Yes
Description:
Maximum core threads count in record thread pool
file_recorder_thread_queue_initial_size¶
Default:
50
Type:
Integer
Need restart: Yes
Description:
Initial size of queue of samples in record thread pool
flash_codecs¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
alaw,ulaw,speex16,h264,vp8
Type:
String
Need restart: No
Description:
This set of codecs (if it is not empty) will be used if either party of a call is Flash
flash_detect_metadata_by_traffic¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true try to detect flash SDP by incoming traffic
flash_detect_metadata_by_traffic_timeout¶
Default:
1000
Type:
Integer
Need restart: No
Description:
Traffic metadata waiting time (ms), if no metadata has been received after this time, the media (video or audio) will be excluded from the SDP.
flash_handler_play_sdp_filename¶
Default:
flash_handler_play.sdp
Type:
String
Need restart: Yes
Description:
Filename of RTMP subscriber sdp
flash_handler_publish_sdp_filename¶
Default:
flash_handler_publish.sdp
Type:
String
Need restart: Yes
Description:
Filename of RTMP publisher sdp
flash_policy.port¶
Default:
843
Type:
Integer
Need restart: Yes
Description:
Listening port for flash policy requests to crossdomain.xml file
flash_rtp_activity_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable RTP activity for Flash streams
flash_streaming_enable¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
Not in use
flight_recorder_capacity¶
Default:
500
Type:
Integer
Need restart: No
Description:
Flight recorder's buffer capacity in records
flight_recorder_categories¶
Default:
NONE
Type:
NONE
WCS1438
Need restart: Yes
Description:
Flight recorder categories
flush_audio_interval¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
80
Type:
Integer
Need restart: Yes
Description:
RTMFP flush interval in milliseconds for flash-audio data from server
flush_video_interval¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
0
Type:
Integer
Need restart: Yes
Description:
RTMFP flush interval in milliseconds for flash-video data from server
force_client_requested_video_resolution¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use client-specified resolution passed in Stream object
force_expires¶
Default:
-1
Type:
Integer
Need restart: No
Description:
If this parameter is set, WCS server will assume that Expires header had this value in 200 OK received in response to SIP REGISTER request
force_local_audio_codec¶
Default:
null
Type:
String
Need restart: No
Description:
This setting is used for Flash SIP calls. You can enforce audio codec, e.g. ulaw, and Flash client should switch to that audio codec
force_periodic_fir_request_for_sip_as_rtmp¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, FIR request will be sent to SIP endpoint every 5 seconds
force_profile_level¶
Default:
null
Type:
String
Need restart: No
Description:
If set, this profile will be used regardless of profiles which figured in H.264 codec negotiation.
Example: force_profile_level=420020
force_rtmp_audio_codec¶
Default:
null
Type:
String
Need restart: No
Description:
Forced codec for old as-RTMP cases using RTMPOutputWriter and for the latest HLS writer
force_sendrecv_for_outgoing_calls¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, force sendrecv' for audio and video for outgoing SIP calls'
frame_cnt_to_determine_their_type¶
Default:
10
Type:
Integer
Need restart: No
Description:
How long to wait for frames to determine their type
frep_database_address¶
Default:
jdbc:mysql://localhost/wcs?user=wcs&password=wcs
Type:
String
Need restart: No
Description:
Address of database that will be used for FREP data storing
frep_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enables Flashphoner remote event protocol
frep_filter_events¶
Default:
CONNECT,CONNECTION_STATUS_EVENT,STREAM,CONNECTION_STATUS_EVENT
Type:
ArrayList
Need restart: Yes
Description:
List of allowed events, which client can send and server can handle
frep_port¶
Default:
8085
Type:
Integer
Need restart: No
Description:
FREP port
frep_role¶
Default:
CLIENT
Type:
CLIENT
SERVER
Need restart: No
Description:
Role of the frep stack, client or server
frep_secret_key¶
Default:
dsjfoiewqhriywqtrfewfiuewqiufh
Type:
String
Need restart: No
Description:
Secret key for FREP authentication
frep_server_ip¶
Default:
null
Type:
String
Need restart: No
Description:
Address of FREP server. Has no effect in server mode.
generate_av_for_ua¶
Default:
null
Type:
String
Need restart: Yes
Description:
WCS server generates RTP traffic (inaudible audio and video with Flashphoner logo) when SIP session is established if detected that the other party's SIP User Agent name is specified in the setting.
Required in case of 'SIP as RTMP' stream with Zoom or Twilio SIP Domain as the SIP endpoint.
Example:
generate_av_for_ua = Twilio Media Gateway
generate_av_start_delay¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
Generator start delay in ms, 0 - no delay
get_callee_url¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Not in use
global_bandwidth_check_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable global bitrate in and out in statistics
h264_allowed_nal_types¶
Default:
1,5,7,8,12
Type:
String
Need restart: No
Description:
List of NAL unit types allowed for decoding
h264_b_frames_force_transcoding¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, force transcoding by higher profile
h264_buffer_nack_list_threshold¶
Default:
30
Type:
Integer
Need restart: No
Description:
JitterBuffer will be reset upon reaching this number of NACK packets
h264_buffer_reset_on_flush_indicator¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Clear h264 buffer state upon flush indication
h264_encoder_filler_data_padding¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Fill frames with Filler Data NAL Units to always maintain max bitrate
h264_encoder_rc_buffer_size¶
Default:
2
Type:
Integer
Need restart: No
Description:
Coefficient for rc buffer
h264_max_nalu_size¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
1346
Type:
Integer
Need restart: Yes
Description:
Maximum size of outgoing NALU while H.264 is encoded. The option is used to prevent MTU excess while encoding high resolution video
h264_new_buffer¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Not in use
h264_remove_filler_data¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enables condition to remove FILLER_DATA from h264 bitstream
h264_sps_buff_scale¶
Default:
1.6
Type:
Double
Need restart: No
Description:
Buffer scale for H264 SPS
h264_sps_default_size¶
Default:
100
Type:
Integer
Need restart: No
Description:
Default size of H264 sps buffer
h264_sps_max_dec_frame_buffering¶
Default:
-1
Type:
Integer
Need restart: No
Description:
SPS VUI decoder buffer
h264_sps_rbsp_scale¶
Default:
1.5
Type:
Double
Need restart: No
Description:
Buffer scale for H264 SPS RBSP
h264_strict_kframe_detect¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, set frame as keyframe only if contains SPS and PPS NAL units or IDR NAL
h265_buffer_nack_list_threshold¶
Default:
30
Type:
Integer
Need restart: No
Description:
JitterBuffer will be reset upon reaching this number of NACK packets
h265_buffer_reset_on_flush_indicator¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Clear h265 buffer state upon flush indication
h265_max_rtp_packet_size¶
Default:
1400
Type:
Integer
Need restart: No
Description:
Maximum size of H265 carrying packet
handler_async_disconnect¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable asynchronous disconnect handler
hangup_incoming_call_state¶
Default:
null
Type:
String
Need restart: No
Description:
Send BUSY_HERE by default.
It is also possible to set custom status that should be returned as BUSY response.
This can be used for IMS use cases.
If true, do not send SIP messages to browser
hide_all¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, do not send SIP messages to browser
hls.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for HLS server
hls.http.port¶
Default:
8082
Type:
Integer
Need restart: Yes
Description:
HLS server HTTP port
hls.https.port¶
Default:
8445
Type:
Integer
Need restart: Yes
Description:
HLS server HTTPS port
hls_abr_auto_start¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable HLS ABR autostart
hls_abr_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable ABR and master playlist for HLS
hls_abr_path_template¶
Default:
{streamName}{abrSuffix}/{streamName}{abrSuffix}.m3u8
Type:
String
Need restart: Yes
Description:
Template for HLS ABR streams path
hls_abr_stream_name_suffix¶
Default:
-HLS-ABR-STREAM
Type:
String
Need restart: Yes
Description:
This is a suffix for HLS ABR stream names, client that wants to get ABR version instead of ordinary version should append this suffix to original stream name'
hls_abr_with_cdn¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Use HLS ABR with CDN or use current node for transcoding
hls_acao_header_domain_mask¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable origin replacement in HLS Access-Control-Allow-Origin header
hls_access_control_headers¶
Default:
null
Type:
String
Need restart: Yes
Description:
HLS response headers
hls_auth_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable check auth tokens for hls
hls_auth_token_cache¶
Default:
10
Type:
Integer
Need restart: No
Description:
Timeout for cache auth tokens in seconds
hls_auto_start¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable HLS autostart
hls_debug_dir¶
Default:
hls-debug
Type:
String
Need restart: Yes
Description:
Folder for debug HLS stream dumps
hls_debug_stream_name_suffix¶
Default:
-DEBUG
Type:
String
Need restart: Yes
Description:
This is a suffix for recorded stream names, used only for debug purposes
hls_delayed_shutdown¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true waits some time before removing HLS provider
hls_dir¶
Default:
hls
Type:
String
Need restart: Yes
Description:
HLS base folder
hls_disable_cleanup¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Do not remove inactive hls files from hdd
hls_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable HLS support
hls_fps_discontinuity¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If false, disable discontinuity tag on fps change
hls_fps_threshold¶
Default:
10
Type:
Integer
Need restart: No
Description:
Value of the threshold in percent change in fps, at which the segment is marked as discontinuity when the setting hls_fps_discontinuity is enabled
hls_hold_segments_before_delete¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, hold segments on disk before delete
hls_hold_segments_size¶
Default:
5
Type:
Integer
Need restart: No
Description:
How many segments to hold, before delete. May be useful for high-latency HLS subscribers.
hls_list_size¶
Default:
8
Type:
Integer
Need restart: No
Description:
Maximum number of segments in playlist
hls_ll_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable Low Latency HLS
hls_manager_provider_timeout¶
Default:
300
Type:
Long
Need restart: No
Description:
HLS manager provider timeout
hls_manifest_file¶
Default:
index.m3u8
Type:
String
Need restart: Yes
Description:
HLS master playlist file name. Default is 'index.m3u8'
hls_metrics_log_size¶
Default:
50
Type:
Integer
Need restart: No
Description:
Number of HLS log lines in hls/find_all response
hls_min_list_size¶
Default:
1
Type:
Integer
Need restart: No
Description:
Minimum number of segments in playlist (should be less than 11)
hls_path_template¶
Default:
{streamName}/{streamName}.m3u8
Type:
String
Need restart: Yes
Description:
Template for HLS non ABR streams path
hls_player_height¶
Default:
480
Type:
Integer
Need restart: No
Description:
HLS player height
hls_player_width¶
Default:
640
Type:
Integer
Need restart: No
Description:
HLS player width
hls_preloader_dir¶
Default:
hls/.preloader
Type:
String
Need restart: No
Description:
HLS preloader dir
hls_preloader_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enables HLS preloader
hls_preloader_time_min¶
Default:
2000
Type:
Long
Need restart: No
Description:
Minimal size of preloader's HLS segment in milliseconds
hls_provider_traffic_waiting_time¶
Default:
6000
Type:
Integer
Need restart: No
Description:
Time in milliseconds that hls provider waits traffic
hls_sdp_filename¶
Default:
hls.sdp
Type:
String
Need restart: Yes
Description:
Filename of HLS sdp
hls_segment_name_suffix_randomizer_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
HLS segment name suffix randomizer
hls_server_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, activate HLS server
hls_session_log_level¶
Default:
INFO
Type:
String
Need restart: No
Description:
Level (INFO, DEBUG, ERROR) for HLS session
hls_static_dir¶
Default:
client2/examples/demo/streaming/hls_static
Type:
String
Need restart: No
Description:
HLS static dir
hls_static_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enables HLS static content
hls_store_segment_in_memory¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Store HLS segments in memory
hls_subscriber_active_timeout¶
Default:
2000
Type:
Long
Need restart: No
Description:
Timeout for active state for hls subscriber
hls_test_interval¶
Default:
182000
Type:
Integer
Need restart: Yes
Description:
HLS test interval
hls_test_run_for¶
Default:
180
Type:
Integer
Need restart: Yes
Description:
HLS test duration in seconds
hls_test_start_streams¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
HLS test streams count
hls_test_start_writers¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
HLS test writers count
hls_time¶
Default:
4
Type:
Integer
Need restart: No
Description:
Size of one HLS segment in seconds
hls_time_min¶
Default:
2000
Type:
Long
Need restart: No
Description:
Minimal size of one HLS segment in milliseconds
hls_version¶
Default:
9
Type:
Integer
Need restart: No
Description:
HLS version
hls_wrap¶
Default:
20
Type:
Integer
Need restart: No
Description:
Maximum number of ts-files. The option is necessary to prevent disc overflow
http.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for HTTP server (statistics)
http.port¶
Default:
8081
Type:
Integer
Need restart: Yes
Description:
WCS server HTTP port
http_client_connection_read_timeout¶
Default:
2000
Type:
Integer
Need restart: No
Description:
HTTP client connection read timeout in milliseconds
http_client_connection_timeout¶
Default:
2000
Type:
Integer
Need restart: No
Description:
HTTP client connection timeout in milliseconds
http_enable_paths¶
Default:
rest,action,admin,shared,client,client_records,embed_player,empty,health-check,zclient-invite,zclient-join,verify,rest-api-spec
Type:
String
Need restart: No
Description:
List of permitted access to the web interface
http_enable_root_redirect¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable root redirect to /admin
https.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for HTTPS server (statistics)
https.port¶
Default:
8444
Type:
Integer
Need restart: Yes
Description:
WCS server HTTPS port
https_server_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, activate HTTPS server
ice_add_ipv6_candidate¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, server will try to add IPv6 ICE candidates
ice_authorize_by_address¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, authorize ICE by IP address only. So, if we receive packets from authorized address but another port, the packets will be accepted even though the port was not authorized
ice_consent_freshness¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, send binding request instead of binding indication for consent freshness
ice_keep_alive_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enables ICE keep-alive
ice_keep_alive_timeout¶
Default:
15
Type:
Integer
Need restart: No
Description:
ICE establishing timeout in seconds. By default, if ICE is in running (waiting COMPLETE) state after 15 seconds, the session will be terminated
ice_tcp_channel_high_water_mark¶
Default:
104857600
Type:
Integer
Need restart: Yes
Description:
High watermark for ICE tcp channels
ice_tcp_channel_low_water_mark¶
Default:
10485760
Type:
Integer
Need restart: Yes
Description:
Low watermark for ICE tcp channels
ice_tcp_receive_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: Yes
Description:
Receive buffer size for ice tcp channels
ice_tcp_send_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: Yes
Description:
Send buffer size for ice tcp channels
ice_tcp_transport¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use tcp transport only
ice_tcp_transport_force¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use tcp transport regardless of client config
ice_timeout¶
Default:
15
Type:
Integer
Need restart: No
Description:
ICE keep-alive timeout in seconds. By default, ICE session will be terminated if no refresh packets from browser in 15 seconds
ice_transport_new¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use new udp transport
ice_udp_channel_high_water_mark¶
Default:
104857600
Type:
Integer
Need restart: No
Description:
High watermark for ice udp channels
ice_udp_channel_low_water_mark¶
Default:
10485760
Type:
Integer
Need restart: No
Description:
Low watermark for ice udp channels
ice_udp_receive_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: No
Description:
Receive buffer size for ice udp channels
ice_udp_send_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: No
Description:
Send buffer size for ice udp channels
ice_udp_transport_new¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use new udp transport
ignore_incoming_call_if_sip_login_port_does_not_match_request_uri¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, terminate incoming call if the SIP port does not correspond to the user indicated in Request-URI
ignore_incoming_rtp¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Discard incoming rtp before decoding/decryption. For test purposes only
in_jitter_buffer_enabled¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, switch on intermediary buffer on server side, which will reset downstream packets according to reset algorithm and min_drop_rate=, max_drop_rate=, min_queue_size=, max_queue_size= and in_jitter_buffer_enabled= settings
inbound_video_rate_stat_rels_throttle¶
Default:
1800
Type:
Integer
Need restart: No
Description:
Inbound video rate stats interval throttle for RELS. 0 - disabled
inbound_video_rate_stat_send_interval¶
Default:
0
Type:
Integer
Need restart: No
Description:
Inbound video rate stat send interval in sec needed to calculate channel quality. 0 - disabled
increase_equals_timestamp¶
Default:
100
Type:
Integer
Need restart: No
Description:
Timestamps are equal within this interval in milliseconds
inject_wait_keyframe_ms¶
Default:
1000
Type:
Long
Need restart: No
Description:
Time server should wait for the injected stream to produce keyframe. Once elapsed server will start to generate video stream with the watermark. Use -1 to turn it off.
injector_pli_request_interval¶
Default:
1000
Type:
Long
Need restart: Yes
Description:
PLI interval at changeover inject to target stream
injector_queue_threshold¶
Default:
1000
Type:
Long
Need restart: Yes
Description:
Minimum injector's queue size for overload identification
ip¶
Default:
0.0.0.0
Type:
String
Need restart: Yes
Description:
External IPv4 address. This IP address will differ from specified with ip_local option when WCS server is behind NAT
ip_local¶
Default:
0.0.0.0
Type:
String
Need restart: Yes
Description:
WCS server will create sockets and listen on this interface
ip_v6¶
Default:
Type:
String
Need restart: Yes
Description:
External IPv6 address. This IP address will differ from specified with ip_local option when WCS server is behind NAT
jitter_buffer_always_detect_frame_type¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enables mandatory detecting frame type from rtp packets
jitter_buffer_attempt_to_correct_broken_timestamp¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If enabled, jitter buffer adding +1 to broken rtp timestamp
jitter_buffer_capacity¶
Default:
0
Type:
Integer
Need restart: No
Description:
JitterBuffer will drop frames when value exceeded
jitter_threshold¶
Default:
50
Type:
Integer
Need restart: No
Description:
RTMFP. If jitter between UDP media packets is greater than the threshold, it will be reported
jni_cache_class¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, cache JNI Class object
jni_debug_enable¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enables jni logs in stdout
keep_alive.algorithm¶
Default:
HIGH_LEVEL
Type:
INTERNAL
NONE
HIGH_LEVEL
Need restart: Yes
Description:
Keep-alive algorithm: INTERNAL, NONE, or HIGH_LEVEL
keep_alive.enabled¶
Default:
websocket,rtmfp
Type:
String
Need restart: Yes
Description:
Enable keep-alive for the listed protocols
keep_alive.peer_interval¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Keep-alive peer interval (Not in use)
keep_alive.probes¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
Number of unsuccessfull attempts to ping connected client (WebSocket, RTMP, RTMFP).
If reached, server will consider the client as disconnected and will release the associated resources.
keep_alive.server_interval¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Interval in milliseconds between attempts to ping connected client (WebSocket, RTMP, RTMFP)
keep_alive_streaming_sessions_enabled¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, server sends keep-alive REST requests to check if stream playback is allowed to continue / resume
kill_event_scanner¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Debug option, for development only
ll_hls_can_skip_segments_for_delta_list¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
6
Type:
Integer
Need restart: No
Description:
Number of segments in playlist can skipped for delta playlist
ll_hls_create_preloader¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, enables LL-HLS Preloader creator
ll_hls_custom_preloader_dir¶
Default:
custom-preloader
Type:
String
Need restart: Yes
Description:
LL-HLS custom preloader base folder
ll_hls_delta_list_size¶
Default:
6
Type:
Integer
Need restart: No
Description:
Number of segments in playlist delta
ll_hls_dir¶
Default:
ll-hls
Type:
String
Need restart: Yes
Description:
LL-HLS base folder
ll_hls_fragmented_mp4¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enables FragmentedMP4 container for low-latency hls media files
ll_hls_keep_min_segment_duration¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true keep minimum duration of parent segments, so duration will be >= hls_time_min
ll_hls_list_size¶
Default:
12
Type:
Integer
Need restart: No
Description:
Maximum number of segments in playlist for LL-HLS
ll_hls_max_number_of_parent_segments_containing_partials¶
Default:
5
Type:
Integer
Need restart: No
Description:
Max number of parent segments containing partials
ll_hls_max_size_queue¶
Default:
50
Type:
Integer
Need restart: No
Description:
Maximum size of buffer for hls media data
ll_hls_min_size_queue¶
Default:
10
Type:
Integer
Need restart: No
Description:
Minimal size of buffer for hls media data
ll_hls_part_hold_back_count¶
Default:
6
Type:
Integer
Need restart: No
Description:
PART-HOLD-BACK attribute value in Part Target Duration
ll_hls_partial_time_max¶
Default:
400
Type:
Long
Need restart: No
Description:
Maximum size of one partial HLS segment in milliseconds
ll_hls_playlist_delta_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
LL-HLS playlist delta
ll_hls_preloader_dir¶
Default:
ll-hls/.preloader
Type:
String
Need restart: No
Description:
LL-HLS preloader dir
ll_hls_preloader_segment_count¶
Default:
5
Type:
Integer
Need restart: No
Description:
LL-HLS Preloader segment count
ll_hls_preloader_segment_duration¶
Default:
400
Type:
Long
Need restart: No
Description:
Duration of preloader LL-HLS segment in milliseconds
ll_hls_time_min¶
Default:
2000
Type:
Long
Need restart: No
Description:
Minimal size of one LL-HLS segment in milliseconds
load_balancing_acao_header¶
Default:
Type:
String
Need restart: Yes
Description:
Use this value for Access-Control-Allow-Origin (ACAO) header in the response when cross-domain HTTP request to the loadbalancer received
load_balancing_enabled¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, activate loadbalancer
log_metrics_stats¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enables/Disables log metrics statistic
log_metrics_time_buffer¶
Default:
10000
Type:
Long
Need restart: Yes
Description:
Setting for time buffer(lesser value = faster result, higher value = slower and more accurate). min 1 second
mail.password¶
Default:
null
Type:
String
Need restart: Yes
Description:
Password for the mail server
mail.username¶
Default:
null
Type:
String
Need restart: Yes
Description:
Username for the mail server
mail.verification.ttl¶
Default:
86400000
Type:
Long
Need restart: Yes
Description:
Verification transaction TTL
mail.verification.ttl.active¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Remove invalid verification transactions
mail.verification.ttl.interval¶
Default:
60000
Type:
Long
Need restart: Yes
Description:
Verification transaction cleanup interval
mail.verification.url¶
Default:
null
Type:
String
Need restart: Yes
Description:
Base url for verification service, e.g. http://my.wcs.ip:8081/
manager_http_ports_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, bind old manager http(s) ports 9091 and 8888
matroska_unknown_segment_size¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Unknown segment atom size
max_callid_length¶
Default:
32
Type:
Integer
Need restart: No
Description:
Maximum length of SIP callID. If the length of generated callID exceeds this value, it will be cut to this length
max_drop_rate¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Queue size will be increased if loss raises up to this value.
Is used only if out_jitter_buffer_enabled=true or in_jitter_buffer_enabled=true
max_queue_size¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Packets will be reset if queue size exceeds this maximum value.
Is used only if out_jitter_buffer_enabled=true or in_jitter_buffer_enabled=true
media_dir¶
Default:
media
Type:
String
Need restart: Yes
Description:
Media base folder
media_port_from¶
Default:
31001
Type:
Integer
Need restart: Yes
Description:
Beginning of media ports range for ICE, RTP, SRTP, RTCP
media_port_stress_test_iterations¶
Default:
1
Type:
Integer
Need restart: No
Description:
Media port stress test iterations
media_port_stress_test_thread_sleep¶
Default:
5
Type:
Integer
Need restart: No
Description:
Media port stress test thread sleeping interval
media_port_stress_test_threads¶
Default:
5
Type:
Integer
Need restart: No
Description:
Media port stress test threads count
media_port_to¶
Default:
32000
Type:
Integer
Need restart: Yes
Description:
End of media ports range for ICE, RTP, SRTP, RTCP
media_ports_auditor_interval¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Audit interval for busy and free ports, in milliseconds
media_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
Number of audits to make sure freed port is not bound.
Freed port will be returned to the pool of free ports if this number of successfull audits is reached
media_processor_input_data_stat_window¶
Default:
30000
Type:
Integer
Need restart: No
Description:
Window for gathering min and max incoming data arrival time, in ms
media_transponder_sdp_filename¶
Default:
media_transponder.sdp
Type:
String
Need restart: Yes
Description:
Filename of transponder sdp
min_drop_rate¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Queue size will be decreased if loss reduces to this value.
Is used only if out_jitter_buffer_enabled=true or in_jitter_buffer_enabled=true
min_queue_size¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Queue size will not be decreased lower that this minimum value.
Is used only if out_jitter_buffer_enabled=true or in_jitter_buffer_enabled=true
mixer_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
Mixer will be terminated after {mixer_activity_timer_cool_off_period * mixer_activity_timer_timeout} since last stream activity for the corresponding mixer
mixer_activity_timer_timeout¶
Default:
-1
Type:
Integer
Need restart: No
Description:
If there is no streams added to mixer within this timeout in milliseconds, corresponding mixer will be terminated
mixer_app_name¶
Default:
defaultApp
Type:
String
Need restart: No
Description:
AppName for mixer streams
mixer_audio_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
When false, mixer stream has video-only
mixer_audio_only_height¶
Default:
360
Type:
Integer
Need restart: No
Description:
Height constraint for mixer audio only frame
mixer_audio_only_width¶
Default:
640
Type:
Integer
Need restart: No
Description:
Width constraint for mixer audio only frame
mixer_audio_opus_float_coding¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Use float optimisations for opus audio coding.
mixer_audio_silence_threshold¶
Default:
-50.0
Type:
Double
Need restart: No
Description:
Audio silence threshold in db
mixer_audio_threads¶
Default:
4
Type:
Integer
Need restart: No
Description:
How many threads should multithreaded audio mixer use
mixer_auto_create_delimiter¶
Default:
#
Type:
String
Need restart: No
Description:
Mixer auto create stream/room delimiter
mixer_auto_start¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable mixer autostart
mixer_autoscale_desktop¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Separate screen share font size from other frames
mixer_debug_mode¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Turns on debug mode, this will output debug information directly onto mixers canvas'
mixer_decode_stream_name¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Decode stream name to mixer's canvas
mixer_desktop_align¶
Default:
TOP
Type:
TOP
BOTTOM
LEFT
RIGHT
CENTER
Need restart: No
Description:
Alignment of screen sharing stream
mixer_display_stream_name¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Output stream name to mixer's canvas
mixer_font_size¶
Default:
20
Type:
Integer
Need restart: No
Description:
Font size for stream name and debug info
mixer_font_size_audio_only¶
Default:
40
Type:
Integer
Need restart: No
Description:
Font size for stream name and debug info for audio only streams
mixer_frame_background_colour¶
Default:
0x2B2A2B
Type:
String
Need restart: No
Description:
Hex value of frame background colour
mixer_idle_timeout¶
Default:
60000
Type:
Long
Need restart: No
Description:
Mixer idle timeout in milliseconds
mixer_in_buffering_ms¶
Default:
200
Type:
Integer
Need restart: No
Description:
How much stream should be buffered before it gets into mix
mixer_incoming_time_rate_lower_threshold¶
Default:
0.95
Type:
Double
Need restart: No
Description:
Relation between incoming stream time and actual machine mixing time, 0.9 means that incoming time rate can be 10% lower then actual stream playback rate
mixer_incoming_time_rate_upper_threshold¶
Default:
1.05
Type:
Double
Need restart: No
Description:
Relation between incoming stream time and actual machine mixing time, 1.2 means that incoming time rate can be 20% bigger then actual stream playback rate
mixer_layout_class¶
Default:
com.flashphoner.media.mixer.video.presentation.GridLayout
Type:
String
Need restart: Yes
Description:
Name of class for custom mixer layout
mixer_layout_dir¶
Default:
Type:
String
Need restart: No
Description:
Directory name for custom mixer descriptors
mixer_linear_smoothing_audio¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Smoothly fade-in and fade-out audio in mixer. 20ms of audio is lost during fade-in
mixer_lossless_video_processor_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable custom video processor for mixer incoming streams, setting this to true may degrade realtime part
mixer_lossless_video_processor_max_mixer_buffer_size_ms¶
Default:
200
Type:
Integer
Need restart: No
Description:
Max size that is allowed for mixers incoming buffer, after reaching this point processor will use own buffer instead'
mixer_lossless_video_processor_wait_time_ms¶
Default:
20
Type:
Integer
Need restart: No
Description:
How long to wait before checking mixer's incoming buffer again in case it was full
mixer_maintain_streams_delay_while_buffered¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If false, mixer does not drop audio video frames to maintain minimum set delay mixer_in_buffering_ms
mixer_mcu_audio¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable mcu like audio mixing, each added stream will have dedicated audio mix available as a separate stream
mixer_mcu_multithreaded_delivery¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Use separate threads for mcu streams injest into engine.
mixer_mcu_multithreaded_mix¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Mix audio/video in separate threads.
mixer_mcu_video¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Works only with mcu audio, send video to each audio mcu stream. Video stays the same as in the root mixer.
mixer_minimal_font_size¶
Default:
1
Type:
Integer
Need restart: No
Description:
Minimal font size for stream name if autoscaling is on
mixer_out_buffer_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable buffer for out mixer streams
mixer_out_buffer_initial_size¶
Default:
2000
Type:
Long
Need restart: No
Description:
Initial size of output mixer buffer in milliseconds
mixer_out_buffer_max_bufferings_allowed¶
Default:
-1
Type:
Integer
Need restart: No
Description:
mixer_out_buffer_max_bufferings_allowed
mixer_out_buffer_overflow_allowed_deviation¶
Default:
1000
Type:
Long
Need restart: No
Description:
max allowed difference between min(buffer) and max(buffer). if this constraint met over the rtmp_in_buffer_overflow_deviation_window overflow state will be set leading to clock acceleration
mixer_out_buffer_overflow_deviation_window¶
Default:
30000
Type:
Integer
Need restart: No
Description:
window for gathering min and max buffer sizes over time, in ms
mixer_out_buffer_overflow_rate¶
Default:
0.15
Type:
Double
Need restart: No
Description:
buffer clock acceleration rate. To calculate increase in output speed use (1 + rate) / buffer = 1/x
mixer_out_buffer_polling_time¶
Default:
100
Type:
Long
Need restart: No
Description:
Output mixer buffer polling time in milliseconds
mixer_out_buffer_start_size¶
Default:
150
Type:
Long
Need restart: No
Description:
Start size of output mixer buffer in milliseconds
mixer_prune_streams¶
Default:
false
Type:
Boolean
Need restart: No
Description:
When true, prune mixer stream
mixer_realtime¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Turns on realtime version of mixer
mixer_separate_buffering_audio_video¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, separate audio video buffering in mixer
mixer_show_separate_audio_frame¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Show audio frame for audio+video stream if added with hasVideo: false
mixer_text_align¶
Default:
BOTTOM_LEFT
Type:
TOP_LEFT
TOP_CENTER
TOP_RIGHT
CENTER
BOTTOM_LEFT
BOTTOM_CENTER
BOTTOM_RIGHT
EXTERNAL_TOP_CENTER
EXTERNAL_BOTTOM_CENTER
Need restart: Yes
Description:
Text position relative to frame
mixer_text_autoscale¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable stream name autoscaling
mixer_text_background_colour¶
Default:
0x2B2A2B
Type:
String
Need restart: No
Description:
Hex value of stream names background colour
mixer_text_background_opacity¶
Default:
100
Type:
Integer
Need restart: No
Description:
Opacity of text background percentage
mixer_text_bulk_write¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Use bulk write with DirectByteBuffer for text
mixer_text_bulk_write_with_buffer¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Use bulk write with DirectByteBuffer for text, cache whole text as a frame
mixer_text_colour¶
Default:
0xFFFFFF
Type:
String
Need restart: No
Description:
Hex value of stream names colour
mixer_text_cut_top¶
Default:
3
Type:
Integer
Need restart: No
Description:
Clip top part of the text
mixer_text_display_room¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Display room name in participant stream names
mixer_text_font¶
Default:
Serif
Type:
String
Need restart: No
Description:
Font of mixer text
mixer_text_outside_frame¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
NO
Type:
String
Need restart: Yes
Description:
Text position relative to frame
mixer_text_outside_frame_padding¶
Default:
50
Type:
Integer
Need restart: Yes
Description:
External padding for outside frame text
mixer_text_padding_bottom¶
Default:
5
Type:
Integer
Need restart: No
Description:
Padding for the bottom side of text in pixels
mixer_text_padding_left¶
Default:
5
Type:
Integer
Need restart: No
Description:
Padding for the left side of text in pixels
mixer_text_padding_right¶
Default:
4
Type:
Integer
Need restart: No
Description:
Padding for the right side of text in pixels
mixer_text_padding_top¶
Default:
5
Type:
Integer
Need restart: No
Description:
Padding for the top side of text in pixels
mixer_thread_priority¶
Default:
5
Type:
Integer
Need restart: No
Description:
Mixer thread priority, min 1 max 10
mixer_type¶
Default:
NATIVE
Type:
JAVA
NATIVE
MULTI_THREADED_NATIVE
Need restart: No
Description:
Mixer implementation, can be JAVA, NATIVE or MULTI_THREADED_NATIVE
mixer_use_sdp_state¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable audio/video only stream detection via sdp state
mixer_video_background_filename¶
Default:
null
Type:
String
Need restart: No
Description:
Mixer video background. Example: background.png
mixer_video_bitrate_kbps¶
Default:
2000
Type:
Integer
Need restart: No
Description:
Encoded video bitrate kbps
mixer_video_buffer_length¶
Default:
1000
Type:
Integer
Need restart: No
Description:
Video buffer length for decoded frames
mixer_video_desktop_fullscreen¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Display desktop stream in fullscreen mode
mixer_video_desktop_layout_inline_padding¶
Default:
10
Type:
Integer
Need restart: No
Description:
Padding between video streams in bottom row (under screen sharing stream)
mixer_video_desktop_layout_padding¶
Default:
30
Type:
Integer
Need restart: No
Description:
Padding between top row (screen sharing stream) and bottom row (other streams)
mixer_video_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
When false, mixer stream has audio-only
mixer_video_fps¶
Default:
30
Type:
Integer
Need restart: No
Description:
Fps constraint for mixer stream
mixer_video_grid_layout_middle_padding¶
Default:
10
Type:
Integer
Need restart: No
Description:
Padding between video streams in one row (when there is no screen sharing stream)
mixer_video_grid_layout_padding¶
Default:
30
Type:
Integer
Need restart: No
Description:
Padding between rows of video streams (when there is no screen sharing stream)
mixer_video_height¶
Default:
720
Type:
Integer
Need restart: No
Description:
Height constraint for mixer stream
mixer_video_layout_desktop_key_word¶
Default:
desktop
Type:
String
Need restart: No
Description:
Keyword for screen sharing streams
mixer_video_profile_level¶
Default:
42c02a
Type:
String
Need restart: No
Description:
Mixer video profile and level in hex. Example: 42c02a
mixer_video_quality¶
Default:
24
Type:
Integer
Need restart: No
Description:
Encoded video quality (CRF)
mixer_video_stable_fps_threshold¶
Default:
15
Type:
Integer
Need restart: No
Description:
Streams with fps lower then threshold won't trigger buffering of the stream if video buffer was exhausted
mixer_video_threads¶
Default:
4
Type:
Integer
Need restart: No
Description:
How many threads should multithreaded video mixer use
mixer_video_width¶
Default:
1280
Type:
Integer
Need restart: No
Description:
Width constraint for mixer stream
mixer_voice_activity¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable/disable voice activity frame
mixer_voice_activity_colour¶
Default:
0x00CC66
Type:
String
Need restart: No
Description:
Hex value of voice activity colour
mixer_voice_activity_frame_position_inner¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Draw voice activity frame inside the frame. If false - draw around the frame
mixer_voice_activity_frame_thickness¶
Default:
6
Type:
Integer
Need restart: No
Description:
Thickness of voice activity frame
mixer_voice_activity_switch_delay¶
Default:
0
Type:
Integer
Need restart: No
Description:
Voice activity indicator switch off delay in milliseconds
mp4_container_moov_first¶
Default:
true
Type:
Boolean
Need restart: No
Description:
When recording mp4 write moov atom first so recording can be played/downloaded progressively
mp4_container_moov_first_reserve_space¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Turn on space reservation for moov atom to avoid additional filesystem copy
mp4_container_moov_reserved_space_size¶
Default:
2048
Type:
Integer
Need restart: No
Description:
When writing moov first how much space should be reserved for moov atom in kilobytes
mp4_cutter_dir¶
Default:
records
Type:
String
Need restart: Yes
Description:
Folder to place MP4 fragments while playing recording files in browser
mp4_cutter_manager_cache_expire¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Mp4 cutter manager cache expire
mpeg1.gop_size¶
Default:
60
Type:
Integer
Need restart: No
Description:
GOP size or k-frame interval
mpeg1.qmax¶
Default:
24
Type:
Integer
Need restart: No
Description:
Maximum value of quality parameter. The lower the value, the better is quality, and the higher is bitrate. If it is too low (e.g. 1), bitrate is too high and vice versa
mpeg1.qmin¶
Default:
4
Type:
Integer
Need restart: No
Description:
Minimum value of quality parameter. The lower the value, the better is quality, and the higher is bitrate. If it is too low (e.g. 1), bitrate is too high and vice versa
mpeg1.trellis¶
Default:
0
Type:
Integer
Need restart: No
Description:
Trellis quantization
mpegts_agent_sdp_filename¶
Default:
mpegts_agent.sdp
Type:
String
Need restart: Yes
Description:
Filename of MPEG-TS publisher sdp
mpegts_max_pts_diff¶
Default:
1
Type:
Integer
Need restart: No
Description:
if server receives a packet whose PTS differs by more than this value in seconds, sessions of subscribers will be terminated
mpegts_stream_timeout¶
Default:
90000
Type:
Long
Need restart: No
Description:
MpegTS stream with no data will be terminated after this timeout in milliseconds
mpegts_udp_constant_socket¶
Default:
true
Type:
Boolean
Need restart: No
Description:
if true, the server accepts packets only from the first client socket. Packets from other sockets will be ignored
mse_sdp_filename¶
Default:
mse.sdp
Type:
String
Need restart: Yes
Description:
Filename of MSE sdp
msrp_port¶
Default:
2855
Type:
Integer
Need restart: No
Description:
Port for receiving MSRP / TCP connections
multi_record_dir¶
Default:
records
Type:
String
Need restart: Yes
Description:
MultiRecord base folder
multi_recorder_mkv_fill_gaps¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Fill gaps in matroska tracks
multi_recorder_type¶
Default:
MP4
Type:
MKV
MP4
Need restart: No
Description:
MP4 or MKV multiRecorder format
multipart_message_service_uri¶
Default:
null
Type:
String
Need restart: No
Description:
SIP URI for sending message to multiple destinations.
A message is sent from client with Content-Type:multipart/mixed and then sent by SIP server to multiple destinations
multiple_pull_test_server_url¶
Default:
null
Type:
String
Need restart: Yes
Description:
Server url for mpt test
multiple_pull_test_stream_name¶
Default:
null
Type:
String
Need restart: Yes
Description:
Stream name for mpt test
multiple_pull_test_subscribers¶
Default:
100
Type:
Integer
Need restart: No
Description:
multiple_pull_test_subscribers
native_test_aac¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable AAC native test
native_test_decoder¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable decoder native test
native_test_encoder¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable encoder native test
native_test_opus¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable Opus native test
native_test_resampler¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable native test resampler
native_test_run_for¶
Default:
180
Type:
Integer
Need restart: Yes
Description:
Native test duration
native_test_start_threads¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
Native test threads count
native_test_thread_interval¶
Default:
200
Type:
Integer
Need restart: Yes
Description:
Native test interval
netty_deadlock_aware_worker_timeout¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Timeout to detect SSL connection with Netty deadlock
no_media_dump_interval¶
Default:
15000
Type:
Long
Need restart: No
Description:
Period in milliseconds, within which media traffic should be captured by tcpdump when client sends bug report with no_media type
notification_apns_key_id¶
Default:
null
Type:
String
Need restart: No
Description:
Key Id for ios apns
notification_apns_key_path¶
Default:
/usr/local/FlashphonerWebCallServer/conf/apns_auth_key.p8
Type:
String
Need restart: No
Description:
Full path to p8 key file for ios apns
notification_apns_team_id¶
Default:
null
Type:
String
Need restart: No
Description:
Team Id for ios apns
notify_message_call_timeout¶
Default:
null
Type:
String
Need restart: No
Description:
Timeout in milliseconds to wait for client confimation of receiving an incoming message.
When an incoming message is received, it is sent to the destination client, and the confirmation timeout is started. If the client does not confirm receiving the message within the timeout, WCS server responds to the sender that the message was not received and delivered (in cases when delivery report is required)
on_multiple_record_hook_script¶
Default:
on_multiple_record_hook.sh
Type:
String
Need restart: No
Description:
This option points to shell script located in /usr/local/FlashphonerWebCallServer/bin directory, which is started when multipleRecorder is terminated. By default, the script run offline_mixer_tool.sh script located in /usr/local/FlashphonerWebCallServer/tools with default offlineMixer config located in /usr/local/FlashphonerWebCallServer/conf.
on_record_hook_script¶
Default:
on_record_hook.sh
Type:
String
Need restart: No
Description:
This option points to shell script located in /usr/local/FlashphonerWebCallServer/bin directory, which is started when stream is unpublished, if a recording of the stream has been created. Two parameters will be passed to the script:
$1 - the stream name
$2 - absolute name of the file with recording of audio and video of the stream
This script can be used to copy or move a stream record from /usr/local/FlashphonerWebCallServer/records directory to another location as soon as the recording is completed. By default, the script does not contain such commands and should be edited as required.
Example:
STREAM_NAME=$1
SRC_FILE=$2
SRC_DIR=/usr/local/FlashphonerWebCallServer/records/
REPLACE_STR=/var/www/html/stream_records/$STREAM_NAME-
DST_FILE=${SRC_FILE/$SRC_DIR/$REPLACE_STR}
cp $SRC_FILE $DST_FILE
Make sure the script works correctly: start it manually, e.g.
./on_record_hook.sh streamName /usr/local/FlashphonerWebCallServer/records/stream-a58aea39-6333-4cb2-8jtn93gtmgr6mrq0nilk6l958j.mp4
options2flash_delegate¶
Default:
null
Type:
String
Need restart: No
Description:
If true, then wait for a client response prior to responding with 200 OK to an OPTIONS request
opus.encoder.bitrate¶
Default:
-1
Type:
Integer
Need restart: No
Description:
Target bitrate for Opus encoder, in bps
opus.encoder.complexity¶
Default:
-1
Type:
Integer
Need restart: No
Description:
Target complexity for Opus encoder
opus_formats¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of Opus formats (name=value).
Example: maxaveragebitrate=20000.
These formats will be listed in SDP
order_threads_by_seq¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, order incoming SIP messages by sequence number and wait if number is out of order
out_jitter_buffer_enabled¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
If true, switch on intermediary buffer on server side, which will reset upstream packets according to reset algorithm and min_drop_rate=, max_drop_rate=, min_queue_size=, max_queue_size= and in_jitter_buffer_enabled= settings
outbound_port¶
Default:
null
Type:
String
Need restart: No
Description:
SIP port. If this parameter is set, it will redefine values that were transmitted during connection
outbound_proxy¶
Default:
null
Type:
String
Need restart: No
Description:
SIP outbound proxy. If this parameter is set, it will redefine values that were transmitted during connection
outbound_video_rate_stat_rels_throttle¶
Default:
1800
Type:
Integer
Need restart: No
Description:
Outbound video rate stats interval throttle for RELS. 0 - disabled
outbound_video_rate_stat_send_interval¶
Default:
0
Type:
Integer
Need restart: No
Description:
Outbound video rate stat send interval in sec needed to calculate channel quality. 0 - disabled
parse_system_stats¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, gather system level statistics such as netstat, lsof, etc. The parsing may take a lot of time
periodic_fir_request¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then every 5 seconds WCS server sends an RTCP Full Intra Request (FIR) message to the input stream source and then forwards its response to the RTMP CDN.
Required in case of SIP as RTMP' stream with Zoom as the SIP Endpoint and the input stream source, so that every new subscriber receives video keyframe (otherwise, stream video may be not played)'
periodic_fir_request_interval¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Interval to send RTCP FIR in milliseconds
play_stream_force_video_orientation¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Force negotiation of 3gpp video orientation extension for play stream requests
port_from¶
Default:
30000
Type:
Integer
Need restart: No
Description:
Beginning of range of ports for SIP signaling
port_to¶
Default:
31000
Type:
Integer
Need restart: No
Description:
End of range of ports for SIP signaling
preserve_non_mixed_recorded_files¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Two files are created when recording: one for incoming sound, and another for outgoing. Then those files are mixed in one resulting recording.
If this setting is false, the temporary files will be deleted after mixing.
If true, the files will be saved
print_publication_tables¶
Default:
false
Type:
Boolean
Need restart: No
Description:
RTMFP. If true, print statistics of streams in logs
print_rtcp_stats¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, print RTCP report on end of session
priority_outside_codecs¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then outside (browser) codecs will be in first place
process_remote_sdp_candidates¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, process candidates from SDP
profiles¶
Default:
640028
Type:
String
Need restart: No
Description:
Comma-separated list of H.264 profiles. These profiles will be used in SDP for video calls
proxy_propagate_fir¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Propagate FIR requests through proxy
proxy_use_h264_packetization_mode_1_only¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use H.264 packetization mode 1
ptime¶
Default:
20
Type:
Integer
Need restart: Yes
Description:
Packetization time. Use carefully
ptime_corrector_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enabling corrector by required packetization time
publication_report_format¶
Default:
null
Type:
String
Need restart: No
Description:
RTMFP. Sets format for statistics.
Possible value: csv
pull_streams¶
Default:
null
Type:
String
Need restart: Yes
Description:
Comma separated list of urls to pull from at server startup
queue_ping_period¶
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Queue ping interval in ms
queue_stat_log¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable queue statistics logging
queue_transcoder_core_router_uri¶
Default:
tcp://127.0.0.1:5555
Type:
String
Need restart: No
Description:
Queue transcoder core router URI
queue_transcoder_receive_timeout¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
Queue transcoder receive timeout
queue_transcoder_shm_path¶
Default:
/dev/shm/
Type:
String
Need restart: No
Description:
Path to shared memory objects for queue transcoder
queue_transcoder_shm_size¶
Default:
5
Type:
Integer
Need restart: Yes
Description:
Shared memory object size for queue transcoder
queue_transcoder_transmit_timeout¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
Queue transcoder transmit timeout
queue_transcoder_worker_router_uri¶
Default:
ipc:///tmp/flashphoner.pipe
Type:
String
Need restart: No
Description:
Queue transcoder core router URI
record¶
Default:
null
Type:
String
Need restart: No
Description:
Path to the directory for audio call recordings. If this path is designated, then audio call recordings will be saved to that directory in WAV Track format.
Also, this is used for recording PCM audio on streams for debug needs (see record_audio_processor_pcm= setting)
record_audio_buffer_max_size¶
Default:
100
Type:
Integer
Need restart: No
Description:
Record audio buffer size
record_audio_codec_channels¶
Default:
2
Type:
Integer
Need restart: No
Description:
Codec channel count used for recording streams
record_audio_codec_sample_rate¶
Default:
44100
Type:
Integer
Need restart: No
Description:
Codec sample rate used for recording streams
record_audio_processor_pcm¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, record audio on stream as PCM16. (Then record= option should point to a valid path, e.g. record=/tmp/)
record_close_scheduling_period¶
Default:
20
Type:
Integer
Need restart: Yes
Description:
Buffer check period for closing a record in milliseconds
record_dir¶
Default:
records
Type:
String
Need restart: Yes
Description:
Record base folder
record_fdk_aac_bitrate_mode¶
Default:
5
Type:
Integer
Need restart: No
Description:
Record FDK bitrate mode. 0 - CBR, 1-5 - VBR
record_filename_template¶
Default:
null
Type:
String
Need restart: No
Description:
Filename template for an audio call recording. Besides the default fields, {date} field can also be used
record_flash_published_streams¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, record streams published from native Flash clients and RTMP live encoders such as Wirecast, FFmpeg, FMLE, etc.
record_formats¶
Default:
h264-mp4,vp8-webm
Type:
RecordFormats
Need restart: No
Description:
H264 and VP8 recorder type
record_h264_to_ts¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If set, record to TS instead of mp4
record_mixer_streams¶
Default:
false
Type:
Boolean
Need restart: No
Description:
When true, mixer streams are recorded
record_response_content_disposition_header_value¶
Default:
null
Type:
String
Need restart: No
Description:
/client/records/ path content-disposition header
record_rotation¶
Default:
null
Type:
String
Need restart: No
Description:
If set, rotation for stream recording files is enabled, in seconds or in Megabytes.
Example: 3600 - rotate every hour
Example: 10M - rotate after every 10 Megabytes
record_rotation_index_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, rotation for stream recording files is enabled
record_rtsp_streams¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, record RTSP streams
record_stop_timeout¶
Default:
15
Type:
Integer
Need restart: No
Description:
Record stop timeout in seconds
record_streams¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, WebRTC and RTMFP streams published will be recorded if stream recording is enabled for the publishing client as well: session.createStream({record:true,...}).
The records will be saved to /usr/local/FlashphonerWebCallServer/records directory
record_tmp_dir¶
Default:
records
Type:
String
Need restart: Yes
Description:
Recording temporary files base folder
recording_by_user¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, call is recorded for the initiator of the call only
reg_expires¶
Default:
3600
Type:
Integer
Need restart: No
Description:
Value in seconds, which will be used in Expires header when SIP REGISTER request is sent
rels_client_type¶
Default:
HTTP
Type:
JDBC
HTTP
Need restart: Yes
Description:
ClickHouse client implementation
rels_consumer_interval¶
Default:
1000
Type:
Integer
Need restart: Yes
Description:
Buffer consumer interval
rels_database_address¶
Default:
http://localhost:8123/wcs?user=wcs&password=wcs
Type:
String
Need restart: No
Description:
Address of ClickHouse database that will be used for events storing
rels_enabled¶
Default:
Type:
EnumSet
Need restart: No
Description:
Enables logging of the listed event types to database
rels_event_buffer_initial_size¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Initial buffer size
rels_event_buffer_interval¶
Default:
60000
Type:
Integer
Need restart: No
Description:
Buffer interval
rels_event_buffer_max_size¶
Default:
40000
Type:
Integer
Need restart: No
Description:
Max buffer size
rels_media_session_events_default_frequency¶
Default:
1000
Type:
Integer
Need restart: Yes
Description:
Default frequency (ms) of RELS media session events if not specified by REST API request
rels_test_start_threads¶
Default:
2
Type:
Integer
Need restart: Yes
Description:
RELS test threads count
rels_test_thread_events¶
Default:
2
Type:
Integer
Need restart: Yes
Description:
RELS test thread events
rels_test_thread_interval¶
Default:
2
Type:
Integer
Need restart: Yes
Description:
RELS test thread interval
remove_ssrc_attr¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, remove ssrc attribute
replace_cached_pool_with_default_pool¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, replaces cached thread pool with default
resample_video¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable video rescaling.
Example:
1. Publish video as 640x480 (4:3)
2. Play video as 400x225 (16:9)
If resample_video=true, WCS server will rescale video from 640x480 to 400x225 and it will be flattened vertically.
If resample_video=false, video will be cut down to 400x225, and part of the video will be lost.
So, when setting playback width and height, you should specify appropriate ratio (e.g., 320x240 for 640x480 published stream); then, if resample_video=true, video will be rescaled properly
rest_access_control_allow_credentials¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Rest-api response access_control_allow_credentials header
rest_access_control_allow_headers¶
Default:
content-type,x-requested-with
Type:
String
Need restart: No
Description:
Rest-api response access_control_allow_headers header
rest_access_control_allow_methods¶
Default:
POST
Type:
String
Need restart: No
Description:
Rest-api response access_control_allow_methods header
rest_access_control_allow_origin¶
Default:
*
Type:
String
Need restart: No
Description:
Rest-api response access_control_allow_origin header
rest_access_control_headers¶
Default:
null
Type:
String
Need restart: Yes
Description:
REST response headers
rest_client_request_retry_count¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
How many times to retry sent request; 0 means no retries.
rest_client_request_sent_retry_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
True if it's OK to retry non-idempotent requests that have been sent.
rest_hook_secret_key¶
Default:
null
Type:
String
Need restart: No
Description:
Rest hook secret key
rest_hook_send_hash¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Rest hook send hash
rest_max_connections¶
Default:
200
Type:
Integer
Need restart: Yes
Description:
Rest max connextions
rest_request_timeout¶
Default:
15
Type:
Integer
Need restart: Yes
Description:
Rest request timeout in seconds
rfc2833_packets_count¶
Default:
null
Type:
String
Need restart: No
Description:
Number of RTP packets for sending one DTMF
rmi.port¶
Default:
1098
Type:
Integer
Need restart: Yes
Description:
Internal RMI port for communications with WCS Manager
room_idle_timeout¶
Default:
60000
Type:
Long
Need restart: Yes
Description:
Room idle timeout in milliseconds
rtc_ice_add_local_component¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, add local component for ICE candidates
rtc_ice_add_local_interface¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, ip_local= address will be added to ICE candidates as another candidate. (External IP address specified in ip= setting is added to ICE candidates by default)
rtc_ip¶
Default:
null
Type:
String
Need restart: No
Description:
External IP address for WebRTC. Can be used for WebRTC deployment on particular network interface having external address different from the one specified with ip= setting
rtc_ip_local¶
Default:
null
Type:
String
Need restart: No
Description:
Local IP address for WebRTC. Can be used for WebRTC deployment on particular network interface having local address different from the one specified with ip_local= setting
rtcp_pli_request_interval¶
Default:
1000
Type:
Long
Need restart: No
Description:
Minimal waiting time to send PLI after receiving K-frame
rtcp_sender_interval¶
Default:
0.1
Type:
Double
Need restart: No
Description:
Guard RTCP interval based on the specified fraction of RTCP bitrate
rtmfp.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for RTMFP server
rtmfp.port¶
Default:
1935
Type:
Integer
Need restart: Yes
Description:
RTMFP server port, UDP
rtmfp_server_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Enable/disable rtmfp server
rtmp.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for RTMP server
rtmp.port¶
Default:
1935
Type:
Integer
Need restart: Yes
Description:
RTMP server port, TCP
rtmp.server_buffer_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable/disable buffering rtmp data on java's heap if socket buffer is full
rtmp.server_channel_high_water_mark¶
Default:
52428800
Type:
Integer
Need restart: Yes
Description:
High watermark for connected rtmp channels
rtmp.server_channel_low_water_mark¶
Default:
5242880
Type:
Integer
Need restart: Yes
Description:
Low watermark for connected rtmp channels
rtmp.server_channel_send_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: Yes
Description:
Send buffer size for rtmp channels
rtmp.server_read_socket_timeout¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
TCP socket write timeout for RTMP server, in seconds
rtmp.server_socket_timeout¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
TCP socket write and read timeout for RTMP server for, in seconds
rtmp.server_write_socket_timeout¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
TCP socket write timeout for RTMP server, in seconds
rtmp.use_server_socket_timeout¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
DEPRECATED (use rtmp.server_socket_timeout, rtmp.server_read_socket_timeout, rtmp.server_write_socket_timeout). If true, use for RTMP server TCP socket timeout set with rtmp.server_socket_timeout option
rtmp_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
RTMP agent will be terminated after {rtmp_activity_timer_cool_off_period * rtmp_activity_timer_timeout} since last subscriber activity for the corresponding RTMP stream
rtmp_activity_timer_timeout¶
Default:
60000
Type:
Integer
Need restart: No
Description:
If there is no subscribers for an RTMP stream within this timeout in milliseconds, corresponding RTMP session will be terminated
rtmp_agent_sdp_filename¶
Default:
rtmp_agent.sdp
Type:
String
Need restart: Yes
Description:
Filename of RTMP agent sdp
rtmp_appkey_source¶
Default:
default
Type:
String
Need restart: No
Description:
RTMP appkey source: default/app
rtmp_command_amf3¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
rtmp_command_amf3
rtmp_detect_h264_frame_type¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, for H264 frames, frame type will be determined by the NAL units instead of RTMP control field
rtmp_dump_dir¶
Default:
rtmp_dump_dir
Type:
String
Need restart: Yes
Description:
Rtmp dump base folder
rtmp_dump_incoming_video¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, write incoming rtmp video (4 bytes length | flv packet)
rtmp_dump_republished_video¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, write republished outgoing rtmp video (4 bytes length | flv packet)
rtmp_flash_ver_publisher¶
Default:
FMLE/3.0
Type:
String
Need restart: No
Description:
RTMP publisher Flash version
rtmp_flash_ver_subscriber¶
Default:
LNX 9,0,124,2
Type:
String
Need restart: No
Description:
RTMP subscriber Flash version
rtmp_in_buffer_clear_threshold¶
Default:
30000
Type:
Long
Need restart: No
Description:
once reached in overflow state buffer will clear up to overflow lower bound
rtmp_in_buffer_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable buffer for incoming RTMP streams
rtmp_in_buffer_increase_rate¶
Default:
0.25
Type:
Double
Need restart: No
Description:
buffer increase rate
rtmp_in_buffer_initial_size¶
Default:
2000
Type:
Long
Need restart: No
Description:
Initial size of incoming RTMP buffer in milliseconds
rtmp_in_buffer_input_delay_threshold¶
Default:
0
Type:
Long
Need restart: No
Description:
Once stream delay reached the threshold buffer will passthrough stream without buffering, 0 - turned off
rtmp_in_buffer_max_bufferings_allowed¶
Default:
-1
Type:
Integer
Need restart: No
Description:
rtmp_in_buffer_max_bufferings_allowed
rtmp_in_buffer_overflow_allowed_deviation¶
Default:
1000
Type:
Long
Need restart: No
Description:
max allowed difference between min(buffer) and max(buffer). if this constraint met over the rtmp_in_buffer_overflow_deviation_window overflow state will be set leading to clock acceleration
rtmp_in_buffer_overflow_deviation_window¶
Default:
30000
Type:
Integer
Need restart: No
Description:
window for gathering min and max buffer sizes over time, in ms
rtmp_in_buffer_overflow_lower_bound¶
Default:
1000
Type:
Long
Need restart: No
Description:
Lower bound for switching from overflow to hold state
rtmp_in_buffer_overflow_rate¶
Default:
0.15
Type:
Double
Need restart: No
Description:
buffer clock acceleration rate. To calculate increase in output speed use (1 + rate) / buffer = 1/x
rtmp_in_buffer_overflow_state_change_delay¶
Default:
10000
Type:
Long
Need restart: No
Description:
Enforces delay between two consecutive overflow states
rtmp_in_buffer_polling_time¶
Default:
100
Type:
Long
Need restart: No
Description:
Incoming RTMP buffer polling time in milliseconds
rtmp_in_buffer_start_size¶
Default:
300
Type:
Long
Need restart: No
Description:
Start size of incoming RTMP buffer in milliseconds
rtmp_metadata_to_sdp_state¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Translate publishers metadata into sdp state, this is used in conjunction with mixer_use_sdp_state'
rtmp_out_buffer_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable buffer for outgoing RTMP streams
rtmp_out_buffer_initial_size¶
Default:
2000
Type:
Long
Need restart: No
Description:
Initial size of outgoing RTMP buffer in milliseconds
rtmp_out_buffer_max_bufferings_allowed¶
Default:
-1
Type:
Integer
Need restart: No
Description:
rtmp_out_buffer_max_bufferings_allowed
rtmp_out_buffer_overflow_allowed_deviation¶
Default:
1000
Type:
Long
Need restart: No
Description:
max allowed difference between min(buffer) and max(buffer). if this constraint met over the rtmp_in_buffer_overflow_deviation_window overflow state will be set leading to clock acceleration
rtmp_out_buffer_overflow_deviation_window¶
Default:
30000
Type:
Integer
Need restart: No
Description:
window for gathering min and max buffer sizes over time, in ms
rtmp_out_buffer_overflow_rate¶
Default:
0.15
Type:
Double
Need restart: No
Description:
buffer clock acceleration rate. To calculate increase in output speed use (1 + rate) / buffer = 1/x
rtmp_out_buffer_polling_time¶
Default:
50
Type:
Long
Need restart: No
Description:
Outgoing RTMP buffer polling time in milliseconds
rtmp_out_buffer_start_size¶
Default:
300
Type:
Long
Need restart: No
Description:
Start size of outgoing RTMP buffer in milliseconds
rtmp_output_writer_bufsize¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
0
Type:
Integer
Need restart: No
Description:
Buffer time for FFRtmpOutputWriter old outbound buffer for as-RTMP cases
rtmp_port_from¶
Default:
33001
Type:
Integer
Need restart: No
Description:
First port in RTMP ports range, for RTMP republisher
rtmp_port_to¶
Default:
34000
Type:
Integer
Need restart: No
Description:
Last port in RTMP ports range, for RTMP republisher
rtmp_ports_auditor_interval¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Audit interval for RTMP ports, in milliseconds
rtmp_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: No
Description:
Number of audits to make sure freed port is not bound.
Freed port will be returned to the pool of free ports if this number of successfull audits is reached
rtmp_publisher_ip¶
Default:
Type:
String
Need restart: Yes
Description:
IPv4 address for outgoing RTMP publishing
rtmp_publisher_start_time_ts¶
Default:
1000
Type:
Long
Need restart: No
Description:
RTMP publisher start time
rtmp_pull_agent_account_for_lost_audio¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable RTMP pull agent account for lost audio
rtmp_pull_allow_to_reuse_uri¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, allow to multiple pulling with the same URI
rtmp_pull_rtp_activity_detection¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable RTP activity detection while RTMP pulling
rtmp_push_auto_start¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable RTMP push autostart for newly published streams
rtmp_push_auto_start_url¶
Default:
null
Type:
String
Need restart: No
Description:
RTMP server address to auto start pushing to
rtmp_push_restore¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then reconnect after connection reset by peer
rtmp_push_restore_attempts¶
Default:
3
Type:
Integer
Need restart: No
Description:
RTMP push reconnect attempts
rtmp_push_restore_interval_ms¶
Default:
5000
Type:
Integer
Need restart: No
Description:
RTMP push reconnect interval in ms
rtmp_receive_buffer_size_predictor_factory¶
Default:
2053
Type:
Integer
Need restart: Yes
Description:
RTMP receive buffer size predictor factory in bytes
rtmp_send_video_first¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Send video first in RTMP
rtmp_server_channel_receive_buffer_size¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
RTMP receive buffer size in bytes
rtmp_server_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Enable/disable rtmp server
rtmp_transponder_force_kframe_interval¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, force k-frame interval for transponder in latest cases as-RTMP'. It is implemented sending RTCP PLI, if that is supported'
rtmp_transponder_full_url¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, ignore streamName and use full rtmpUrl in transponders and as RTMP' cases.
If false, streamName will be used as RTMP stream name and rtmpUrl will be treated as URL to RTMP application, e.g. rtmp://host:1935/live'
rtmp_transponder_kframe_interval¶
Default:
60
Type:
Integer
Need restart: No
Description:
Forced k-frame interval in frames. See also rtmp_transponder_force_kframe_interval= setting.
rtmp_transponder_metadata¶
Default:
null
Type:
String
Need restart: No
Description:
RTMP transponder metadata
rtmp_transponder_send_metadata¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, RTMP transponder will send metadata
rtmp_transponder_stream_name_prefix¶
Default:
rtmp_
Type:
String
Need restart: No
Description:
The specified prefix is added for all as-RTMP stream names. By default, stream named stream1 will be republished as RTMP stream with name rtmp_stream1
rtmp_use_stream_params_as_connection¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Use stream params as connection
rtp_activity_audio¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, RTP activity check is enabled for audio.
rtp_activity_audio_exclude¶
Default:
Type:
String
Need restart: No
Description:
Disable RTP activity for audio stream with name matches this regex pattern
rtp_activity_detecting¶
Default:
null
Type:
String
Need restart: No
Description:
Disables / enables and sets value of RTP activity timeout, in seconds.
By default, RTP session will be closed if there is no media traffic in 60 seconds period (rtp_activity_detecting=true,60)
rtp_activity_timeout¶
Default:
60
Type:
Long
Need restart: No
Description:
RTP activity timer in seconds
rtp_activity_video¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, RTP activity check is enabled for video.
If false, this check is enabled for audio only
rtp_activity_video_exclude¶
Default:
Type:
String
Need restart: No
Description:
Disable RTP activity for video stream with name matches this regex pattern
rtp_bundle¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable rtp bundle
rtp_elapsed_time_threshold¶
Default:
10000
Type:
Long
Need restart: No
Description:
RTP elapsed time threshold, in milliseconds
rtp_generator_start_timeout¶
Default:
2000
Type:
Integer
Need restart: Yes
Description:
Time in ms for enable generator if no rtp in call
rtp_in_buffer¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use RTP in buffer
rtp_in_buffer_initial_size¶
Default:
2000
Type:
Integer
Need restart: No
Description:
Initial size of incoming RTP buffer in milliseconds
rtp_in_buffer_polling_time¶
Default:
100
Type:
Long
Need restart: No
Description:
Incoming RTP buffer polling time in milliseconds
rtp_in_reset_marker¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use RTP in reset marker
rtp_paced_sender¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable paced sender for WebRTC video session. EXPERIMENTAL
rtp_paced_sender_capacity¶
Default:
200000000
Type:
Long
Need restart: No
Description:
RTP paced sender capacity
rtp_paced_sender_increase_interval¶
Default:
50
Type:
Integer
Need restart: No
Description:
Paced sender increase interval
rtp_paced_sender_initial_rate¶
Default:
200000
Type:
Integer
Need restart: No
Description:
Paced sender initial rate
rtp_paced_sender_k_deviation¶
Default:
0.02
Type:
Double
Need restart: No
Description:
Paced sender K deviation
rtp_paced_sender_k_down¶
Default:
0.02
Type:
Double
Need restart: No
Description:
Paced sender K down
rtp_paced_sender_k_up¶
Default:
0.04
Type:
Double
Need restart: No
Description:
Paced sender K up
rtp_paced_sender_period¶
Default:
1000
Type:
Long
Need restart: No
Description:
RTP paced sender period
rtp_paced_sender_queue_size¶
Default:
2000
Type:
Integer
Need restart: No
Description:
Outgoing queue maximum size
rtp_paced_sender_refill¶
Default:
200000000
Type:
Long
Need restart: No
Description:
RTP paced sender refill
rtp_packet_cache_size¶
Default:
250
Type:
Integer
Need restart: No
Description:
Cache size for sent packets. This is used only on video sessions to provide response to NACK requests
rtp_receive_buffer_predicator_size¶
Default:
1500
Type:
Integer
Need restart: No
Description:
DatagramSocket constructing: receiveBufferSizePredictorFactory size
rtp_receive_buffer_size¶
Default:
65536
Type:
Integer
Need restart: No
Description:
Buffer size for incoming RTP and SRTP (WebRTC).
DatagramSocket constructing: receiveBufferSize
rtp_send_buffer_size¶
Default:
65536
Type:
Integer
Need restart: No
Description:
Buffer size for outgoing RTP and SRTP (WebRTC).
DatagramSocket constructing: sendBufferSize
rtp_session_init_always¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true init rtp session for all media providers
rtsp.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for RTSP server
rtsp.port¶
Default:
554
Type:
Integer
Need restart: Yes
Description:
RTSP server port
rtsp_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
RTSP agent will be terminated after {rtsp_activity_timer_cool_off_period * rtsp_activity_timer_timeout} since last subscriber activity for the corresponding RTSP stream
rtsp_activity_timer_timeout¶
Default:
60000
Type:
Integer
Need restart: No
Description:
If there is no subscribers for an RTSP stream within this timeout in milliseconds, corresponding RTSP session will be terminated
rtsp_auth_cnonce¶
Default:
1234567890
Type:
String
Need restart: Yes
Description:
RTSP server port
rtsp_client_address¶
Default:
0.0.0.0
Type:
InetAddress
Need restart: Yes
Description:
RTSP client address
rtsp_client_strip_audio_codecs¶
Default:
null
Type:
String
Need restart: No
Description:
Comma-separated list of audio codecs which will not be used for RTSP
rtsp_fail_on_error_track¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, RTSP pulling fails on error in any track
rtsp_interleaved_channels¶
Default:
null
Type:
String
Need restart: No
Description:
Interleaved mode channels: audio channels;video channels. Default: dynamic channels
rtsp_interleaved_enable_rtcp¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable replying to RTCP packets on the RTSP interleaved channel
rtsp_interleaved_mode¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, interleaved mode for RTSP (RTP over RTSP/TCP) is enabled
rtsp_pcap_server_custom_stream_name¶
Default:
null
Type:
String
Need restart: No
Description:
User custom stream name for all pcap sources with incrementing #N at the end
rtsp_pcap_server_handler_redirect_url¶
Default:
null
Type:
String
Need restart: Yes
Description:
Rtsp pcap server redirect URL
rtsp_pcap_server_redirect_method¶
Default:
OPTIONS
Type:
String
Need restart: Yes
Description:
Rtsp pcap server redirect method: OPTIONS/DESCRIBE
rtsp_port_from¶
Default:
32001
Type:
Integer
Need restart: No
Description:
First TCP port in the port range for RTSP pooling agent
rtsp_port_to¶
Default:
33000
Type:
Integer
Need restart: No
Description:
Last TCP port in the port range for RTSP pooling agent
rtsp_ports_auditor_interval¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Audit interval for RTSP ports, in milliseconds
rtsp_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: No
Description:
Number of audits to make sure freed port is not bound.
Freed port will be returned to the pool of free ports if this number of successfull audits is reached
rtsp_refresh_requests_limit¶
Default:
5
Type:
Integer
Need restart: No
Description:
Maximum number of non-answered GET_PARAMETER refresh requests. Stop sending refresh requests if the limit has been reached
rtsp_server_auth_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable RTSP server authentication
rtsp_server_enabled¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, activate RTSP server
rtsp_server_forse_interleave¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, force interleaved mode for RTSP server and answer with interleaved mode SDP
rtsp_server_packetization_mode¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
H.264 packetization mode for RTSP server. FU-A by default
rtsp_server_profile_level_id¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
H.264 profile-level-id for RTSP server
rtsp_server_sdp_filename¶
Default:
rtsp_server.sdp
Type:
String
Need restart: No
Description:
Filename of RTSP server sdp
rtsp_user_agent¶
Default:
Type:
String
Need restart: No
Description:
User agent indicated in RTSP packets
rvg_timer_activity¶
Default:
500
Type:
Integer
Need restart: No
Description:
RVG timer interval in milliseconds
rvg_timer_delay¶
Default:
500
Type:
Integer
Need restart: No
Description:
RVG timer initial delay in milliseconds
scheduling_service_core_threads¶
Default:
5
Type:
Integer
Need restart: Yes
Description:
Core threads count for scheduling service
sdp_origin_username¶
Default:
Flashphoner
Type:
String
Need restart: No
Description:
Sdp Origin value
send_receive_buffer_size¶
Default:
1600
Type:
Integer
Need restart: Yes
Description:
RTMFP buffer size in bytes
send_receive_on_incoming_re_invite¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, send receive' on incoming re-INVITE'
session_idle_timeout¶
Default:
300000
Type:
Integer
Need restart: Yes
Description:
RTMFP server-side timeout in milliseconds if no UDP messages received over RTMFP/UDP session
sessions_auditor_interval¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Audit interval for pending media sessions
sessions_auditor_session_timeout¶
Default:
60000
Type:
Integer
Need restart: Yes
Description:
Audit timeout for pending media sessions
set_sync_time_from_ts¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Workaround for SIP audio only
sfu_bridge_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
With the bridge SFU will wrap tracks into streams
sfu_mongo_storage_database¶
Default:
wcs_sfu
Type:
String
Need restart: Yes
Description:
MongoDB database name
sfu_mongo_storage_url¶
Default:
mongodb://localhost/?w=majority
Type:
String
Need restart: Yes
Description:
MongoDB connection url
sfu_periodic_fir_request¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, then every sfu_periodic_fir_request_interval' seconds WCS server sends an RTCP Full Intra Request (FIR) message to the input stream source'
sfu_periodic_fir_request_interval¶
Default:
30000
Type:
Integer
Need restart: No
Description:
Interval to send RTCP FIR in milliseconds to SFU participant
sfu_proxy_pli_requests¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, then every PLI request from viewer will be forwarded to source
sfu_storage_type¶
Default:
YAML
Type:
YAML
MONGO
Need restart: Yes
Description:
SFU storage type, YAML or MONGO
sip.pre_init¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, use SIP pre-initiation
sip_add_contact_id¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, record SIP as RTMP stream and SIP as stream
sip_as_rtmp_java_client¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, then the latest RTMP transponder implementation will be used for as-RTMP cases. See also use_rtmp_java_client option
sip_as_rtmp_stream_type¶
Default:
live
Type:
String
Need restart: No
Description:
Sets RTMP AMF stream type for as-RTMP cases
sip_auditor_dialog_timeout¶
Default:
10000
Type:
Integer
Need restart: No
Description:
SIP auditor dialog timeout
sip_auditor_transaction_timeout¶
Default:
50000
Type:
Integer
Need restart: No
Description:
SIP auditor transaction timeout
sip_dns_failover¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable DNS failover.
See also sip_srv_lookup= option
sip_force_rtcp_feedback¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, force rtcp feedback to sip provider
sip_force_session_expires¶
Default:
1800
Type:
Integer
Need restart: No
Description:
Forced session expiration timeout in seconds. WCS server will send refresh request before the timeout is reached
sip_force_tcp¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, force TCP usage for SIP messaging
sip_invite_params_to_headers¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, place SIP INVITE parameters to headers
sip_msg_listener¶
Default:
com.flashphoner.sdk.sip.ChangeCallIdListener
Type:
String
Need restart: No
Description:
Full name of Java class that implements interface ISipMessageListener
public interface ISipMessageListener {
void processMessage(SIPMessage sipMessage);
}
sip_ports_auditor_interval¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Audit interval for SIP ports, in milliseconds
sip_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: No
Description:
Number of audits to make sure freed port is not bound.
Freed SIP port will be returned to the pool of free ports if this number of successfull audits is reached
sip_record_stream¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, record SIP as RTMP stream and SIP as stream
sip_remove_video_sdp_section_instead_of_adding_inactive_with_zero_port¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, fully remove video part of SDP. If false, just set video part to inactive
sip_sdp_unsupported_protocols¶
Default:
String
Type:
UDP/DTLS/SCTP
Need restart: No
Description:
null
sip_session_expires_header¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use Expires header
sip_single_route_only¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then traffic is passed only to the streaming engine, and is not passed to the SIP caller
sip_srv_lookup¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable DNS SRV lookup.
See also sip_dns_failover= option
sip_thread_pool_size¶
Default:
null
Type:
String
Need restart: No
Description:
Size of SIP thread pool
sip_timer¶
Default:
null
Type:
String
Need restart: No
Description:
Value of timer T1 according to RFC 3261, in milliseconds
sip_traffic_class¶
Default:
null
Type:
String
Need restart: No
Description:
QoS class for SIP traffic
sip_use_netty¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, use Netty
sip_use_reentrant_listener¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable SIP reentrant listener
sip_use_tls¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, TLS used for SIP connections
sip_user_agent_shutdown_timeout¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Timeout for remove sip user agent for unregister in sip provider. Default is 5000 ms
snapshot_auto_dir¶
Default:
/usr/local/FlashphonerWebCallServer/snapshots
Type:
String
Need restart: No
Description:
Snapshots dir
snapshot_auto_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then enable snapshot auto cut
snapshot_auto_naming¶
Default:
mediaSessionId
Type:
String
Need restart: No
Description:
Snapshot auto naming
snapshot_auto_rate¶
Default:
60
Type:
Integer
Need restart: No
Description:
Snapshot rate. By default save every 60 frame
snapshot_auto_retention¶
Default:
20
Type:
Integer
Need restart: No
Description:
Snapshot retention. By default keep last 20 frames
snapshot_taking_attempts¶
Default:
30
Type:
Integer
Need restart: Yes
Description:
The number of attempts to take a snapshot. By default 30
snapshot_taking_interval_ms¶
Default:
3000
Type:
Integer
Need restart: Yes
Description:
Snapshot taking interval. By default 3000 milliseconds
speex_g711_speex_transcoding¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then Speex16 codec is forcedly deleted from the list of supported codecs, which leads to double transcoding. The option was used for debugging
speex_in_policy¶
Default:
null
Type:
String
Need restart: No
Description:
Speex encoding settings used in transcoding featuring the codec.
Default:
8 - Quality
false - VBR encoding
8 - Quality of VBR
4 - Algorithmic complexity
start_test¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, tests listed in streaming_tests= setting will be launched after WCS server startup
stats¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, enable sampling for streams. The sampling is used for charts
stats_average_calculation_window¶
Default:
30
Type:
Integer
Need restart: Yes
Description:
Window size for general average stats calculation
stats_bitrate_window¶
Default:
1000
Type:
Integer
Need restart: No
Description:
Window size to collect bitrate statistics
stats_fps_window¶
Default:
1000
Type:
Integer
Need restart: No
Description:
Window size to collect FPS statistics
stats_incoming_stream_monitor_deviation_threshold¶
Default:
500
Type:
Integer
Need restart: No
Description:
If deviation between audio and video is greater than the threshold in milliseconds, it will be logged
stats_sampling_frequency¶
Default:
1000
Type:
Long
Need restart: Yes
Description:
Interval in milliseconds. Stream sampling will be taken with the specified frequency
stream_idle_bitrate_monitoring¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable monitoring of published streams based on bitrate
stream_idle_bitrate_monitoring_threshold_bps¶
Default:
10000
Type:
Long
Need restart: No
Description:
Lowest bitrate possible for the active stream
stream_idle_bitrate_monitoring_window_sec¶
Default:
120
Type:
Integer
Need restart: No
Description:
Mean stream bitrate calculation window in seconds
stream_record_policy¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
Type:
String
Need restart: No
Description:
Available values: streamName, template.
By default, WCS server generates filename based on mediaSessionId and login.
If set to streamName', recorded file will have the exact name of stream with extension .mp4 or .webM (depending on the video codec).
If set to 'template', filename will be built using template.
See also stream_record_policy_template= option'
stream_record_policy_template¶
Default:
Type:
String
Need restart: No
Description:
If set, name of recorded file will be built using the specified template.
Example: {streamName}-{startTime}-{sessionId}-{mediaSessionId}-{login}-{audioCodec}-{videoCodec}-{duration}
Note that if filename length exceeds system limit, recording may be not created.
See also stream_record_policy= option
streaming_custom_stream_stress_test_encoding_subscriber_groups¶
Default:
1
Type:
String
Need restart: No
Description:
StreamingCustomStreamStressTest / Number of subscribers for transcoded stream, per encoding groups
E.g., three encoding groups with three subscribers in each
streaming_custom_stream_stress_test_encoding_subscriber_groups=3,3,3
streaming_custom_stream_stress_test_max_proxy_subscribers¶
Default:
1
Type:
Integer
Need restart: No
Description:
StreamingCustomStreamStressTest / Number of subscribers for non-transcoded stream (codecs, resolution and bitrate are the same for publisher and subscriber)
streaming_custom_stream_stress_test_rate¶
Default:
1000
Type:
Long
Need restart: No
Description:
StreamingCustomStreamStressTest / Period in milliseconds. Each period a new subscriber will be added
streaming_custom_stream_stress_test_stream_name¶
Default:
STRESS_TEST_STREAM
Type:
String
Need restart: No
Description:
StreamingCustomStreamStressTest / Name of stream published on WCS server, which will be used for the test
streaming_custom_stream_stress_test_subscriber_ttl_sec¶
Default:
30
Type:
Long
Need restart: No
Description:
StreamingCustomStreamStressTest / Lifetime of subscriber in seconds
streaming_distributor_audio_subgroup_queue_max_waiting_time¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Maximum time that subgroup thread will wait for frame arrival before executing next iteration
streaming_distributor_audio_subgroup_queue_size¶
Default:
300
Type:
Integer
Need restart: No
Description:
Size of queue for distribution subgroup)
streaming_distributor_audio_subgroup_size¶
Default:
500
Type:
Integer
Need restart: No
Description:
Video sessions per group
streaming_distributor_dump_interval¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
Interval in minutes for getting distributor thread dumps
streaming_distributor_queue_max_waiting_time¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Maximum time that distributor thread will wait for frame arrival before executing next iteration
streaming_distributor_queue_size¶
Default:
300
Type:
Integer
Need restart: Yes
Description:
Size of queue. Processor will block distributor queue upon it reaching this size (i.e., no more space for new frames)
streaming_distributor_queue_size_dump_threshold¶
Default:
0.95
Type:
Double
Need restart: No
Description:
Distributor queue size threshold for getting dump
streaming_distributor_queue_size_log_threshold¶
Default:
10
Type:
Integer
Need restart: Yes
Description:
Threshold for logging distributor queue size
streaming_distributor_subgroup_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable video distribution subgroups)
streaming_distributor_subgroup_queue_max_waiting_time¶
Default:
5000
Type:
Integer
Need restart: No
Description:
Maximum time that subgroup thread will wait for frame arrival before executing next iteration
streaming_distributor_subgroup_queue_size¶
Default:
300
Type:
Integer
Need restart: No
Description:
Size of queue for distribution subgroup)
streaming_distributor_subgroup_size¶
Default:
2
Type:
Integer
Need restart: No
Description:
Video sessions per group
streaming_distributor_video_proxy_pool_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Use thread pool for video distribution, proxy only
streaming_load_test_duration_minutes¶
Default:
5
Type:
Long
Need restart: No
Description:
StreamingLoadTest / Test duration in minutes
streaming_load_test_encoding_subscriber_groups¶
Default:
1
Type:
String
Need restart: No
Description:
StreamingLoadTest / Number of subscribers for transcoded stream, per encoding groups
E.g., two encoding groups: one with two subscribers and another with five
streaming_load_test_encoding_subscriber_groups =2,5
streaming_load_test_proxy_subscribers¶
Default:
1
Type:
Integer
Need restart: No
Description:
StreamingLoadTest / Number of subscribers for non-transcoded stream (codecs, resolution and bitrate are the same for publisher and subscriber)
streaming_processor_queue_max_waiting_time¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Maximum time that processor thread will wait for frame arrival before executing next iteration
streaming_processor_queue_size¶
Default:
300
Type:
Integer
Need restart: Yes
Description:
Size of queue. Feeding thread (e.g., RTP thread in case of WebRTC) will block processor queue upon it reaching this size (i.e., no more space for new frames)
streaming_sessions_keep_alive_app_keys¶
Default:
Type:
String
Need restart: No
Description:
Comma-separated list of appKeys of server-side applications. If set, WCS server will periodically send StreamKeepAliveEvent for all streams within the listed applications.
For example, if set defaultApp,myApp', the event will be sent for all streams connected to those two applications.
See also streaming_sessions_keep_alive_interval= option'
streaming_sessions_keep_alive_interval¶
Default:
10000
Type:
Long
Need restart: No
Description:
StreamKeepAliveEvent sending interval. See also streaming_sessions_keep_alive_app_keys= option
streaming_stress_test_duration_minutes¶
Default:
5
Type:
Long
Need restart: No
Description:
StreamingStressTest / Test duration in minutes
streaming_stress_test_encoding_subscriber_groups¶
Default:
1
Type:
String
Need restart: No
Description:
StreamingStressTest / Number of subscribers for transcoded stream, per encoding groups
E.g., five encoding groups with five or ten subscribers in each
streaming_stress_test_encoding_subscriber_groups=5,5,5,10,10
streaming_stress_test_max_proxy_subscribers¶
Default:
100
Type:
Integer
Need restart: Yes
Description:
Websocket connections to test
streaming_stress_test_rate¶
Default:
1000
Type:
Long
Need restart: No
Description:
StreamingStressTest / Period in milliseconds. Each period a new subscriber will be added
streaming_stress_test_subscriber_ttl_sec¶
Default:
30
Type:
Long
Need restart: No
Description:
StreamingStressTest / Lifetime of subscriber in seconds
streaming_tests¶
Default:
Type:
String
Need restart: No
Description:
Comma-separated list of tests which will be launched after WCS server startup if start_test=true.
Available tests:
- MP4AgentTest
- StreamingCustomStreamStressTest
- StreamingLoadTest
- StreamingStressTest
streaming_video_decoder_fast_start¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, all incoming streams are decoded.
If false, incoming stream is decoded only on demand, when codecs, resolution or bitrate are different for the stream publisher and subscriber
streaming_video_decoder_wait_for_distributors¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Stop decoding temporarily if one of the distributors fails to keep up with decoding
streaming_video_decoder_wait_for_distributors_max_queue_size¶
Default:
5
Type:
Integer
Need restart: Yes
Description:
Stop decoding when one of distributors queue reaches specified size (See streaming_video_decoder_wait_for_distributors)
streaming_video_decoder_wait_for_distributors_timeout¶
Default:
33
Type:
Integer
Need restart: Yes
Description:
Specifies how long decoding should wait before another distributors queue check (See streaming_video_decoder_wait_for_distributors)
streaming_video_decoder_warmup¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Warmup video decoder with P frame after I frame regardless of decoding point availability
streaming_video_decoder_warmup_frames¶
Default:
5
Type:
Integer
Need restart: No
Description:
How many P frames should be used for warmup
strict_get_callee_policy¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Not in use
stun_freshness_period¶
Default:
1500
Type:
Integer
Need restart: No
Description:
STUN freshness period in milliseconds
stun_freshness_timeout¶
Default:
15000
Type:
Integer
Need restart: No
Description:
STUN freshness timeout in milliseconds
stun_server¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
stun1.l.google.com:19302
Type:
String
Need restart: No
Description:
STUN server, which is used for WebRTC ICE, if enable_candidate_harvester=true
stun_socket_buffer_size¶
Default:
100
Type:
Integer
Need restart: No
Description:
Size of STUN socket buffer
stun_socket_queue_timeout¶
Default:
1500
Type:
Integer
Need restart: No
Description:
STUN socket queue timeout in milliseconds
stun_stack_default_thread_pool_size¶
Default:
0
Type:
Integer
Need restart: No
Description:
STUN default thread pool size
stun_wait_candidate_timeout¶
Default:
1000
Type:
Integer
Need restart: No
Description:
STUN waiting candidate timeout for nominate in milliseconds
suppress_audio¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, globally suppress audio on server. This feature is not available for Trial license
suppress_dynamic_logs¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, suppress dynamic logs update
suppress_dynamic_logs_to_server_log¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, suppress dynamic server logs update
sync_time_force_newest¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Force newest synctime after sync change, prevent new packet from getting synctime from past
tcp_relay_packetization2¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable TCP relay packetization for WSPlayer. Should be false when WSPLayer 1.0 is used
tcp_relay_packetization_time¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
20
Type:
Integer
Need restart: No
Description:
Experimental option, allows to send audio packets with custom ptime to WSPlayer 1.0. This property was not tested with new versions and should be removed
tcp_relay_rtcp_interval¶
Default:
2000
Type:
Integer
Need restart: No
Description:
RTCP packets generation interval for TCP relay in milliseconds. RTCP is used to carry stream synchronization
thread_pool_default_core_threads¶
Default:
4
Type:
Integer
Need restart: Yes
Description:
Default core threads count in thread pool (equal to CPUs count)
thread_pool_default_max_threads¶
Default:
8
Type:
Integer
Need restart: Yes
Description:
Maximum core threads count in thread pool
thread_pool_default_queue_size¶
Default:
100
Type:
Integer
Need restart: Yes
Description:
Default thread pool queue size
thread_pool_default_thread_timeout_sec¶
Default:
300
Type:
Integer
Need restart: Yes
Description:
Default thread timeout, in seconds
thread_pools_config_filename¶
Default:
thread_pools_config.json
Type:
String
Need restart: Yes
Description:
Thread pools config filename
throughput_test_receivers_qty¶
Default:
1
Type:
Integer
Need restart: No
Description:
Throughput test receivers quantity
throughput_test_sender_dst¶
Default:
localhost
Type:
String
Need restart: No
Description:
Throughput test sender destination host
throughput_test_senders_qty¶
Default:
1
Type:
Integer
Need restart: No
Description:
Throughput test senders quantity
timing_shift¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
null
Type:
String
Need restart: No
Description:
Timer ambiguity in milliseconds, which is used in a stream stagnation (in case the stream is too fast in relation to timestamps) and compensates inaccuracy of system timers.
Is used only if in_jitter_buffer_enabled=true
trace_socket_fd¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, trace usage of socket file descriptors for HLS, HTTP, RTSP, WebSockets and HTTP LB client
transcoder_agent_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
Transcoder agent will be terminated after {rtsp_activity_timer_cool_off_period * rtsp_activity_timer_timeout} since last subscriber activity for the corresponding RTSP stream
transcoder_agent_activity_timer_timeout¶
Default:
60000
Type:
Integer
Need restart: No
Description:
If there is no subscribers for an Transcoder agent stream within this timeout in milliseconds, corresponding RTSP session will be terminated
transcoder_agent_key_frame_interval¶
Default:
60
Type:
Integer
Need restart: No
Description:
Transcoder agent key frame interval
transcoder_agent_rtcp_send_interval¶
Default:
2000
Type:
Long
Need restart: No
Description:
Interval in ms for send rtcp from transcoder agent
transcoder_align_encoders¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Align video encoders of the same video input by key frames
transcoding_disabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Force transcoding disabling
turn.server_channel_receive_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: Yes
Description:
Receive buffer size for turn channels
turn.server_channel_send_buffer_size¶
Default:
1048576
Type:
Integer
Need restart: Yes
Description:
Send buffer size for turn channels
turn_ip¶
Default:
0.0.0.0
Type:
InetAddress
Need restart: Yes
Description:
TURN external IP address
turn_ip_local¶
Default:
0.0.0.0
Type:
InetAddress
Need restart: Yes
Description:
TURN internal IP address
turn_life_time¶
Default:
600
Type:
Integer
Need restart: Yes
Description:
TURN Allocation life time
turn_media_port_from¶
Default:
36001
Type:
Integer
Need restart: Yes
Description:
Beginning of media ports range for turn
turn_media_port_to¶
Default:
37000
Type:
Integer
Need restart: Yes
Description:
End of media ports range for turn
turn_media_ports_auditor_interval¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
Audit interval for busy and free ports, in milliseconds
turn_media_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: Yes
Description:
Number of audits to make sure freed port is not bound.
Freed port will be returned to the pool of free ports if this number of successfull audits is reached
turn_password¶
Default:
coM77EMrV7Cwhyan
Type:
String
Need restart: Yes
Description:
TURN password
turn_port¶
Default:
3478
Type:
Integer
Need restart: Yes
Description:
TURN server port
unsupported_messages¶
Default:
null
Type:
String
Need restart: No
Description:
If a message has body noted in this list, then such incoming message will be rejected. Can be useful for some service messages, when delivery to client is not required. The list consists of strings, divided by three colons :::
use_alaw_ulaw_speex_switch¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, switch to the local codec according to content received from SIP side.
If false, use Speex16
use_control_destination_from_incoming_rtcp¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, set RTCP destination by received RTCP packets
use_fdk_aac¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use the fdk-aac fro encoding and decoding
use_ip_local_in_call_id¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use value of ip_local= option when forming callID
use_mp4_h264_aac¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use H.264 + AAC in MP4 container
use_new_aac_encoder¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use the latest AAC encoder
use_new_injector¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable new injectors
use_new_rtcp¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use the latest RTCP module
use_rtcp_synch¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use RTCP synchronization for audio and video
use_rtmp_java_client¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use the latest implementation of RTMP agent for republishing
use_speex_java_impl¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, use Java implementation for Speex codec
use_strict_jitter_buffer¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enables strict jitter buffer
use_tcp_for_long_sip_messages¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, and size of SIP message is more than 1350 bytes, then such message will be sent via TCP.
By default, SIP messages are sent over UDP
use_trying_notification¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, then broadcast SIP response TRYING to client as a call status TRYING
user_agent¶
Default:
Flashphoner/1.0
Type:
String
Need restart: Yes
Description:
User-Agent header value
video_bitstream_normalizer_consecutive_ts_errors_threshold¶
Default:
90
Type:
Integer
Need restart: No
Description:
How many consecutive timestamp errors normalizer can absorb before falling back to original stream timestamp.
video_decoder_max_threads¶
Default:
2
Type:
Integer
Need restart: No
Description:
How many threads will be used for decoding
video_decoder_second_thread_threshold¶
Default:
777000
Type:
Integer
Need restart: No
Description:
Resolution threshold. Once it is reached, decoder should start using second thread.
Example: 800x600 = 480000, 1280x720=921600. So, by default all 720p streams will be decoded using two CPU threads
video_distributor_multi_test¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable video distributor multi test
video_enabled¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
Not in use
video_encoder_h264_gop¶
Default:
60
Type:
Integer
Need restart: No
Description:
GOP size for H.264 encoder
video_encoder_h265_preset¶
Default:
ultrafast
Type:
String
Need restart: No
Description:
Preset value for H.265 encoder
video_encoder_max_threads¶
Default:
3
Type:
Integer
Need restart: No
Description:
How many threads will be used for encoding
video_encoder_second_thread_threshold¶
Default:
777000
Type:
Integer
Need restart: No
Description:
Resolution threshold. Once it is reached, encoder should start using second thread.
Example: 800x600 = 480000, 1280x720=921600. So, by default all 720p streams will be encoded using two CPU threads
video_encoder_vp8_gop¶
Default:
900
Type:
Integer
Need restart: No
Description:
GOP size for VP8 encoder
video_encoding_quality¶
Default:
30
Type:
Integer
Need restart: No
Description:
See information on FFmpeg CRF
video_filter_enable_fps¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Enable video filter
video_filter_enable_rotate¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Enable video rotate filter
video_filter_fps¶
Default:
30
Type:
Long
Need restart: Yes
Description:
Video filter output fps
video_filter_fps_gap_coefficient¶
Default:
2.0
Type:
Double
Need restart: Yes
Description:
Video filter gap coefficient (max gap C x FPS)
video_filter_fps_gop_synchronization¶
Default:
0
Type:
Integer
Need restart: No
Description:
Filters gop value used to provide synchronization point for encoders, use with TRANSCODER_ALIGN_ENCODERS'
video_force_sync_timeout¶
Default:
100
Type:
Integer
Need restart: No
Description:
Waiting for RTCP sync packet on this interval in ms, for video
video_mixer_output_codec¶
Default:
h264
Type:
String
Need restart: No
Description:
Video mixer output codec (multiple codecs not allowed)
video_processor_multi_test¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable video processor multi test
video_reliable¶
Default:
partial
Type:
on
partial
off
Need restart: No
Description:
RTMFP, reliability for video
video_stream_mode_udp¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Not in use
video_streamer_generate_seq¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Should be set to true for transfer of video calls. Otherwise, there may be no video after transfer
video_transcoder_preserve_aspect_ratio¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Try to preserve original aspect ratio of incoming video during transcoding
video_transcoder_round_ratio¶
Default:
0
Type:
Integer
Need restart: No
Description:
Rounding up or down when preserving aspect ratio
vod_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
VOD agent will be terminated after {vod_activity_timer_cool_off_period * vod_activity_timer_timeout} since last subscriber activity for the corresponding RTSP stream
vod_live_loop¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, loop streaming MP4 file as VoD. EXPERIMENTAL
vod_mp4_container_isoparser_heap_datasource¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, use heap datasource
vod_mp4_container_new¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Use new implementation of mp4 container for vod
vod_mp4_container_new_buffer_ms¶
Default:
0
Type:
Integer
Need restart: No
Description:
New implementation of mp4 container will buffer specified time in milliseconds
vod_mp4_test_file¶
Default:
null
Type:
String
Need restart: No
Description:
Path to MP4 file. If start_test=true and streaming_tests=MP4AgentTest, VoD stream playing the file will be published when WCS server is started
vod_mp4_test_loop¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, loop streaming MP4 file. Not in use, replaced by vod_live_loop=
vod_mp4_test_stream_name¶
Default:
null
Type:
String
Need restart: No
Description:
This name will be used as name of VoD stream published for playing MP4 file for test MP4AgentTest.
See also vod_mp4_test_file= setting
vod_rtcp_send_interval¶
Default:
2000
Type:
Long
Need restart: No
Description:
RTCP Send interval for VOD
vod_sink_ready_checks¶
Default:
50
Type:
Integer
Need restart: No
Description:
Waiting for first packet on audio streamer. If no packets within the specified number of checks, then audio injection is aborted
vod_sink_wait_synch_time¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If false, not wait sync time for playing incoming traffic after audio sink
vod_stream_timeout¶
Default:
30000
Type:
Integer
Need restart: No
Description:
VoD stream with no subscribers will be terminated after this timeout in milliseconds
vow_wait_for_sync¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, session will wait for audio AND video before sending stream to client
vp8_buffer_nack_list_threshold¶
Default:
200
Type:
Integer
Need restart: No
Description:
JitterBuffer will be reset upon reaching this number of NACK packets
vp8_max_rtp_packet_size¶
Default:
1400
Type:
Integer
Need restart: Yes
Description:
Maximum size of VP8 carrying packet
vp8_new_buffer¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Not in use
wcs_activity_timer_cool_off_period¶
Default:
1
Type:
Integer
Need restart: No
Description:
WCS agent will be terminated after {wcs_agent_activity_timer_cool_off_period * wcs_agent_activity_timer_timeout} since last activity for the corresponding WCS agent session
wcs_activity_timer_timeout¶
Default:
60000
Type:
Integer
Need restart: No
Description:
If there is no activity within this timeout in milliseconds, corresponding WCS agent session will be terminated
wcs_agent_force_video_orientation¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Force negotiation of 3gpp video orientation extension for wcs agent's
wcs_agent_port_from¶
Default:
34001
Type:
Integer
Need restart: No
Description:
Beginning of range of ports for WCS agent
wcs_agent_port_to¶
Default:
35000
Type:
Integer
Need restart: No
Description:
End of range of ports for WCS agent
wcs_agent_ports_auditor_interval¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Audit interval for WCS agent ports, in milliseconds
wcs_agent_ports_auditor_max_attempts¶
Default:
3
Type:
Integer
Need restart: No
Description:
Number of audits to make sure freed port is not bound.
Freed WCS agent port will be returned to the pool of free ports if this number of successfull audits is reached
wcs_agent_session_alive_check_interval¶
Default:
30000
Type:
Integer
Need restart: No
Description:
Interval in milliseconds to check if WCS agent session is alive
wcs_agent_session_audit¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable WCS agent session audit
wcs_agent_session_connect_timeout¶
Default:
10000
Type:
Integer
Need restart: No
Description:
Connect timeout in milliseconds
wcs_agent_session_timeout¶
Default:
30000
Type:
Integer
Need restart: No
Description:
WCS agent session timeout in milliseconds
wcs_agent_session_use_keep_alive_timeout¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, WCS agent session will use keep alive timeout
wcs_agent_ssl¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable SSL for pulling/re-publishing streams
wcs_agent_uri_path¶
Default:
/websocket
Type:
String
Need restart: Yes
Description:
WCSAgent ws request uri path
wcs_sfu_bridge_enabled¶
Default:
false
Type:
Boolean
Need restart: No
Description:
With the WCS SFU bridge wcs will wrap streams into tracks
wcsoam_batch_timeout¶
Default:
500
Type:
Integer
Need restart: Yes
Description:
WCS OAM receive timeout
wcsoam_buffer_size¶
Default:
20000
Type:
Integer
Need restart: Yes
Description:
WCS OAM buffer size in kB
wcsoam_chunk_size¶
Default:
64
Type:
Integer
Need restart: Yes
Description:
WCS OAM send chunk size in kB
wcsoam_hostname¶
Default:
null
Type:
String
Need restart: Yes
Description:
WCS OAM server hostname
wcsoam_ip¶
Default:
null
Type:
String
Need restart: Yes
Description:
WCS OAM server IP address
wcsoam_keepalive_period¶
Default:
3000
Type:
Integer
Need restart: Yes
Description:
WCS OAM keep alive period
wcsoam_keepalive_timeout¶
Default:
8000
Type:
Integer
Need restart: Yes
Description:
WCS OAM keep alive timeout
wcsoam_ping_enabled¶
Default:
true
Type:
Boolean
Need restart: No
Description:
WCS OAM server ping enable
wcsoam_ping_interval¶
Default:
10000
Type:
Integer
Need restart: Yes
Description:
WCS OAM server ping interval in ms
wcsoam_port¶
Default:
7777
Type:
Integer
Need restart: Yes
Description:
WCS OAM server port
wcsoam_reconnect_interval¶
Default:
5000
Type:
Integer
Need restart: Yes
Description:
WCS OAM reconnect interval in ms
wcsoam_sha_salt¶
Default:
123
Type:
String
Need restart: Yes
Description:
WCS OAM server SHA salt
web_start_with_demo_user¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Enable demo user
web_token_life_time¶
Default:
3600000
Type:
Long
Need restart: No
Description:
Web token life time, default value 1 hour
webm_java_writer_enable¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Replace ffmpeg webm writer with java implementation
webrtc_aes_crypto_provider¶
Default:
BC
Type:
BC
JCE
Need restart: No
Description:
Crypto provider for WebRTC
webrtc_agent_use_webrtc¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, switch WebRTC push and pull to AVP profile
webrtc_cc2¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, the latest congestion control CC2 is used
webrtc_cc2_bitrate_overuse_event¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, enable NBE evant raising
webrtc_cc2_bitrate_overuse_event_interval¶
Default:
5000
Type:
Long
Need restart: No
Description:
NBE event will be raised periodically with this interval in milliseconds
webrtc_cc2_bitrate_overuse_event_threshold¶
Default:
0.05
Type:
Double
Need restart: No
Description:
NBE event will be raised when loss on stream being played reaches this value (5% by default)
webrtc_cc2_cc¶
Default:
false
Type:
Boolean
Need restart: No
Description:
If true, react upon WebRTC playback endpoint (e.g. Chrome) requests, e.g. request the publisher to decrease bitrate
webrtc_cc2_cc_interval¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
500
Type:
Long
Need restart: No
Description:
Congestion control interval, not in use
webrtc_cc2_cc_k_noise¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
0.1
Type:
Double
Need restart: No
Description:
Congestion control noise value, not in use
webrtc_cc2_cc_retransmit_rate_threshold¶
Default:
0.15
Type:
Double
Need restart: No
Description:
Fraction of send bitrate that retransmit bitrate can raise to. By default, retransmit bitrate can use 15% of send bitrate
webrtc_cc2_cc_track_joined_retransmit_bitrate¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, enable tracking of retransmit bitrate across all media groups
webrtc_cc2_enable_burst_grouping¶
Default:
false
Type:
Boolean
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public. CC2 estimation will account for packet burst
webrtc_cc2_local_congestion_event_interval¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
2000
Type:
Long
Need restart: No
Description:
Not in use, legacy code
webrtc_cc2_local_k_threshold¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
0.1
Type:
Double
Need restart: No
Description:
Not in use, legacy code
webrtc_cc2_min_remb_bitrate_bps¶
Default:
100000
Type:
Long
Need restart: No
Description:
Minimum value for received REMB (Receiver Estimated Max Bitrate) boundary in bps. Ignore the boundary if the received value is less than the minimum defined
webrtc_cc2_receiver_state_window¶
Default:
1000
Type:
Long
Need restart: No
Description:
Window size for receiver state, in milliseconds. Default: 1000 - keep and account reports received in last second
webrtc_cc2_twcc¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, enable TWCC reports. EXPERIMENTAL
webrtc_cc_bitrate_window¶
Default:
1000
Type:
Integer
Need restart: No
Description:
Time window in milliseconds. Bitrate estimator works on this time frame
webrtc_cc_initial_avg_noise¶
Default:
0.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_e_0_0¶
Default:
100.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_e_0_1¶
Default:
0.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_e_1_0¶
Default:
0.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_e_1_1¶
Default:
0.1
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_offset¶
Default:
0.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_process_noise_0¶
Default:
1.0E-13
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_process_noise_1¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
0.001
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_slope¶
Default:
0.015625
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_threshold¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
15.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_initial_var_noise¶
Default:
50.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_k_down¶
Default:
1.8E-4
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_k_up¶
Default:
0.01
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_max_bitrate¶
Default:
10000000
Type:
Long
Need restart: No
Description:
Maximum global bitrate for publishing WebRTC streams
webrtc_cc_min_bitrate¶
Default:
30000
Type:
Long
Need restart: No
Description:
Minimum global bitrate for publishing WebRTC streams
webrtc_cc_overusing_threshold¶
Default:
10.0
Type:
Double
Need restart: No
Description:
Internal bitrate estimation configuration, must not be exposed to public
webrtc_cc_use_sync_ts¶
Default:
true
Type:
Boolean
Need restart: No
Description:
If true, timestamp is used as synchronization source
webrtc_pre_init¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
If true, use WebRTC pre-initialization
webrtc_pre_init_timeout¶
Default:
20000
Type:
Integer
Need restart: Yes
Description:
Maximum allowed time for WebRTC preinit
webrtc_sdes_extensions¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Enable sdes rtp header extensions
webrtc_sdp_bandwidth_bps¶
Default:
0
Type:
Long
Need restart: No
Description:
b=AS/b=TIAS in publish sdp
webrtc_sdp_h264_exclude_profiles¶
Default:
Type:
String
Need restart: No
Description:
List of H264 profiles which should be excluded in response on SDP negotiation.
42 - Baseline, 4d - Main, 64 - High
webrtc_sdp_max_bitrate_bps¶
Default:
0
Type:
Long
Need restart: No
Description:
x-google-max-bitrate in publish sdp
webrtc_sdp_min_bitrate_bps¶
Default:
0
Type:
Long
Need restart: No
Description:
x-google-min-bitrate in publish sdp
websocket_uri_path¶
Default:
Type:
String
Need restart: Yes
Description:
WebSocket request uri path
work_around¶
Warning
Deprecated parameter. Will be deleted in future releases
Default:
false
Type:
Boolean
Need restart: No
Description:
Not in use
ws.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for WebSocket server
ws.dashboard.port¶
Default:
8086
Type:
Integer
Need restart: Yes
Description:
WebSocket dashboard connection port
ws.ip_forward_header¶
Default:
X-Real-IP
Type:
String
Need restart: No
Description:
Header for IP forwarding
ws.map_custom_headers¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
If true, parse and inject custom HTTP headers to REST requests
ws.port¶
Default:
8080
Type:
Integer
Need restart: Yes
Description:
WebSocket connection port
ws_client_id_unique_part¶
Default:
true
Type:
Boolean
Need restart: No
Description:
Add unique part to ws client id
ws_connections_test_run_for¶
Default:
1800
Type:
Integer
Need restart: Yes
Description:
Websocket connections test duration in seconds
ws_connections_test_uri¶
Default:
ws://192.168.88.100:8080
Type:
String
Need restart: Yes
Description:
Websocket connections test URI
ws_read_socket_timeout¶
Default:
true
Type:
Boolean
Need restart: Yes
Description:
Enable WebSocket read timeout
ws_read_socket_timeout_sec¶
Default:
120
Type:
Integer
Need restart: Yes
Description:
WebSocket read timeout value (if enabled)
wss.address¶
Default:
0.0.0.0
Type:
InetAddress[]
Need restart: Yes
Description:
Listening address for WebSocket SSL server
wss.cert.password¶
Default:
password
Type:
String
Need restart: Yes
Description:
Key password to the SSL certificate in keystore
wss.dashboard.port¶
Default:
8446
Type:
Integer
Need restart: Yes
Description:
Secure websocket dashboard connection port
wss.keystore.file¶
Default:
/usr/local/FlashphonerWebCallServer/conf/wss.jks
Type:
String
Need restart: Yes
Description:
Keystore file containing SSL certificate for secure WebSocket connection
wss.keystore.password¶
Default:
password
Type:
String
Need restart: Yes
Description:
SSL certificate keystore password
wss.port¶
Default:
8443
Type:
Integer
Need restart: Yes
Description:
WebSocket SSL connection port
wss.ssl.cache_size¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
SSL session objects cache size
wss.ssl.session_timeout¶
Default:
0
Type:
Integer
Need restart: Yes
Description:
Cached SSL session objects timeout, in seconds
zapp_conf_dir¶
Default:
/usr/local/FlashphonerWebCallServer/conf/zclient/conf/
Type:
String
Need restart: Yes
Description:
Zapp configuration directory
zgc_log_parser_enable¶
Default:
false
Type:
Boolean
Need restart: Yes
Description:
Launch ZGC parser for stat output
zgc_log_parser_path¶
Default:
logs/gc-core-[0-9]{4}-[0-9]{2}-[0-9]{2}_[0-9]{2}-[0-9]{2}.log
Type:
String
Need restart: Yes
Description:
ZGC logs path to parse
zgc_log_time_format¶
Default:
yyyy-MM-dd'T'HH:mm:ss.SSSZ
Type:
String
Need restart: Yes
Description:
ZGC logs time format used to parse gc pause time