Опция

Значение по умолчанию

Тип

Требуется перезапуск

Описание

aac_bitrate

128000

Integer

false

AAC encoding bitrate

aac_encoder_sync_drop_threshold

1000

Long

true

JitterBuffer will be reset upon reaching this number of dropped sync packets

aac_test_start_codec

20

Integer

true

AAC test codecs count

aac_test_transcode_iterations

1000

Integer

true

AAC test interval

add_register_auth_headers

false

Boolean

false

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

false

Boolean

false

If true, enable local agent session debug,[]allow_domains,null,String,null,false,null,allow_domains,If set

allow_outside_codecs

true

Boolean

false

If false, dont add outside (browser) codecs to SDP

allow_reinvite_in_hold_state

true

Boolean

false

If true, process re-INVITE requests within the session even if the call is in hold state,[]answer_with_one_codec_in_sdp,false,Boolean,false,false,null,answer_with_one_codec_in_sdp,If true

audio_frames_per_packet

6

Integer

false

RTMFP. Audio will be flushed after this number of audio frames in the packet is reached

audio_incoming_buffer_size

20

Integer

false

Waiting for RTCP sync packet on this interval in packets, for audio,[]audio_incoming_min_buffer_size,2,Integer,2,false,null,audio_incoming_min_buffer_size,Waiting for RTCP sync packet at least on this interval in packets

audio_mixer_max_delay

300

Integer

false

Audio mixer max delay in milliseconds

audio_mixer_output_codec

opus

String

false

Audio mixer output codec (multiple codecs not allowed)

audio_mixer_output_sample_rate

48000

Integer

false

Audio mixer output samle rate in Hz

audio_reliable

partial

on
partial
off

false

RTMFP, reliability for audio,[]audio_stream_mode_udp,true,Boolean,true,true,null,audio_stream_mode_udp,Not in use,deprecatedauto_login_url,null,String,null,false,null,auto_login_url,Not in use,deprecatedav_paced_sender,false,Boolean,false,false,null,av_paced_sender,If true

av_paced_sender_max_buffer_size

5000

Integer

false

Max size of audio or video buffer. Once size is reached buffers are cleared

avcc_buffer_wait_frames_count

5

Integer

false

Wait until the buffer is filled with frames

avcc_send_buffer_size

500000

Integer

false

Avcc send buffer size in bytes

aws_s3_credentials

null

String

true

AWS s3 credentials: region;accessKey;secretKey

balance_header

balance

String

false

This SIP header will be sent to client as a balance

burst_avoidance_count

100

String

false

Burst avoidance count

busy_state

null

String

false

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,com.flashphoner.server.client.DefaultCallRecordListener,String,com.flashphoner.server.client.DefaultCallRecordListener,false,null,call_record_listener,Full name of Java class that implements interface ICallRecordListener
public interface ICallRecordListener {
void onRecordReport(RecordReport recordReport);
},[]cdn_advertise_pulled,false,Boolean,false,true,null,cdn_advertise_pulled,If true

cdn_advertise_streams_by_kframe

false

Boolean

false

Advertise stream to CDN by key frame

cdn_allowed_ips

95.191.131.65

95.191.131.64

88.198.99.220

cdn_force_version

2.0

String

true

Force to set CDN version

cdn_group_origin_to_transcoder_relation

false

Boolean

true

Use CDN group indications to relate origin to transcoder rather than transcoder to edge

cdn_groups

ArrayList

true

CDN groups for this node

cdn_inbound_auditor_interval

1000

Integer

true

Time interval to check inbound connections, in milliseconds,[]cdn_inbound_connection_unanswered_pings,3,Integer,3,true,null,cdn_inbound_connection_unanswered_pings,Inbound connection unanswered pings number.
Connection considered to be lost when this number is reached,[]cdn_inbound_ws_read_socket_timeout,true,Boolean,true,true,null,cdn_inbound_ws_read_socket_timeout,Enable WebSocket read timeout for inbound cdn connactions,[]cdn_inbound_ws_read_socket_timeout_sec,60,Integer,60,true,null,cdn_inbound_ws_read_socket_timeout_sec,WebSocket read timeout value (if enabled) for inbound cdn connections,[]cdn_ip,95.191.130.39,String,null,true,null,cdn_ip,CDN node IP address (or domain name when cdn_nodes_resolve_ip=true),[]cdn_load_interval,500,Integer,500,true,null,cdn_load_interval,load interval,[]cdn_load_node,false,Boolean,false,true,null,cdn_load_node,Turn on cdn load behaviour,[]cdn_load_pool_size,500,Integer,500,true,null,cdn_load_pool_size,load pool,[]cdn_load_pool_size_change_interval,-1,Integer,-1,true,null,cdn_load_pool_size_change_interval,Change pool size every interval,[]cdn_load_pool_size_lower_threshold,-1,Integer,-1,true,null,cdn_load_pool_size_lower_threshold,Lower threshold for pool size change,[]cdn_load_pool_size_upper_threshold,-1,Integer,-1,true,null,cdn_load_pool_size_upper_threshold,Upper threshold for pool size change,[]cdn_load_proto_pull,websocket,String,websocket,true,null,cdn_load_proto_pull,CDN load protocol stream,[]cdn_load_reserved_stream,,String,,true,null,cdn_load_reserved_stream,CDN load reserved stream,[]cdn_load_step,10,Integer,10,true,null,cdn_load_step,load step,[]cdn_load_unique_streams,false,Boolean,false,true,null,cdn_load_unique_streams,Pull only unique streams,[]cdn_load_use_profile_name,false,Boolean,false,true,null,cdn_load_use_profile_name,Put profile name in stream name. Use if entry point is edge,[]cdn_load_use_profiles,false,Boolean,false,true,null,cdn_load_use_profiles,Pull with profiles,[]cdn_node_load_average_threshold,1.0,Double,1.0,true,null,cdn_node_load_average_threshold,If threshold reached node will advertise its state as GROUP_CONNECTIONS

cdn_nodes_acl_refresh_interval

60000

Integer

true

Time interval to refresh CDN node acl list, in milliseconds,[]cdn_nodes_auditor_interval,1000,Integer,1000,true,null,cdn_nodes_auditor_interval,Time interval to check available CDN nodes

cdn_nodes_group_refresh_interval

60000

Integer

true

Time interval to refresh CDN node group, in milliseconds,[]cdn_nodes_resolve_ip,false,Boolean,false,true,null,cdn_nodes_resolve_ip,If true

cdn_nodes_role_refresh_interval

60000

Integer

true

Time interval to refresh CDN node role, in milliseconds,[]cdn_nodes_route_refresh_interval,60000,Integer,60000,true,null,cdn_nodes_route_refresh_interval,Time interval to refresh CDN routes

cdn_nodes_state_refresh_interval

60000

Integer

true

Time interval to refresh CDN node state, in milliseconds,[]cdn_nodes_timeout,-1,Integer,-1,true,null,cdn_nodes_timeout,CDN nodes timeout in seconds. -1 means nodeTimeout disabled,[]cdn_nodes_version_refresh_interval,90000,Integer,90000,true,null,cdn_nodes_version_refresh_interval,Time interval to refresh CDN node version

cdn_origin_allowed_to_transcode

false

Boolean

true

In case no transcoders left node will request transcoding profile from origin

cdn_origin_to_origin_route_propagation

false

Boolean

true

If true, origin sends routes to other origins,[]cdn_outbound_auditor_interval,2000,Integer,2000,true,null,cdn_outbound_auditor_interval,Time interval to check outbound connections

cdn_outbound_connection_timeout

6000

Integer

true

Outbound connection timeout, in milliseconds,[]cdn_outbound_ws_read_socket_timeout,true,Boolean,true,true,null,cdn_outbound_ws_read_socket_timeout,Enable WebSocket read timeout for outbound cdn connactions,[]cdn_outbound_ws_read_socket_timeout_sec,60,Integer,60,true,null,cdn_outbound_ws_read_socket_timeout_sec,WebSocket read timeout value (if enabled) for outbound cdn connections,[]cdn_point_of_entry,,String,,true,null,cdn_point_of_entry,CDN point of entry node IP address (or domain name when cdn_nodes_resolve_ip=true),[]cdn_port,8084,Integer,8084,true,null,cdn_port,CDN server port,[]cdn_role,ORIGIN,ORIGIN
EDGE
TRANSCODER
CONTROLLER,EDGE,true,null,cdn_role,CDN role:
origin - the source of media streams for other CDN nodes
edge (default) pulls media streams from origin CDN node(s),[]cdn_skip_pulled_streams,true,Boolean,true,true,null,cdn_skip_pulled_streams,If true

cdn_ssl

false

Boolean

true

If true, enables SSL,[]cdn_standalone,false,Boolean,false,true,null,cdn_standalone,If true

cdn_strict_transcoding_boundaries

false

Boolean

true

Prevent transcoding to the same or higher resolution of original stream by placing resolution boundary

cdn_strict_transcoding_throws_exception

false

Boolean

true

Whether to fail play or substitute requested profile with original stream if profile hit the strict transcoding boundary

cdn_test_enabled

false

Boolean

true

Turn on cdn tests

cdn_test_interval

500

Integer

true

test interval

cdn_test_max_subscribers_for_stream

10

Integer

true

Max subscribers for each CDN stream. Edge-only setting

cdn_test_pool_size

500

Integer

true

test pool

cdn_test_step

10

Integer

true

test step

cdn_transcoder_degraded_streams_threshold

-1

Integer

true

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

10000

Integer

true

CDN transcoder cache expire for new stream requests

cdn_transcoder_threshold_state

GROUP_CONNECTIONS_ALLOWED

UNKNOWN
PASSIVE
GROUP_CONNECTIONS_ALLOWED
CONNECTIONS_ALLOWED
NEW_STREAMS_ALLOWED

true

If threshold reached node will change state to provided value

cdn_transcoder_video_decoders_load_threshold

-1

Integer

true

If decoders load threshold reached node will advertise it's state as GROUP_CONNECTIONS. Threshold is the sum of decoderWidth*decoderHeight*decoderFPS

cdn_transcoder_video_encoders_load_threshold

-1

Integer

true

If encoders load threshold reached node will advertise it's state as GROUP_CONNECTIONS. Threshold is the sum of encoderWidth*encoderHeight*encoderFPS

cdn_transcoder_video_encoders_threshold

10000

Integer

true

If threshold reached node will advertise it's state as GROUP_CONNECTIONS

chat_listener

null

String

false

Full name of Java class that implements interface IChatListener
public interface IChatListener {
void onMessage(InstantMessage message);
}

check_receiver_origin

false

Boolean

false

If true, check origin of RTCP packet and discard if unknown,[]cli_auth_keys,/usr/local/FlashphonerWebCallServer/.ssh/authorized_keys,String,/usr/local/FlashphonerWebCallServer/.ssh/authorized_keys,true,null,cli_auth_keys,CLI Auth keys file path,[]cli_enabled,true,Boolean,true,true,null,cli_enabled,If true

cli_port

2001

Integer

true

CLI server port

cli_v2_port

2002

Integer

true

CLI V2 server port

client_acl_property_name

aclAuth

String

true

Access list identifier property key that server should look for in custom config when client connects

client_dump_level

0

Integer

false

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,null,String,null,true,null,client_handler,Not in use,deprecatedclient_log_exclude,,String,,false,null,client_log_exclude,Do not log events listed,[]client_log_force_debug,false,Boolean,false,false,null,client_log_force_debug,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,60,Integer,60,false,null,client_log_force_debug_timeout,Timeout after which client logs will be turned off,[]client_log_level,debug,String,INFO,false,null,client_log_level,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,true,Boolean,true,false,null,client_mode,If true

client_subscribe_streams_max

10

Integer

false

Max subscribe streams allowed for client

client_timeout

3600000

Integer

false

Client timeout value in milliseconds

codec_terminator_timeout

5000

Integer

false

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,opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv,String,null,false,null,codecs,List of supported codecs ordered by priority,[]codecs_exclude_cdn,null,String,null,false,null,codecs_exclude_cdn,Comma-separated list of codecs which will not be used for CDN,[]codecs_exclude_sip,mpeg4-generic,flv,mpv,String,null,false,null,codecs_exclude_sip,Comma-separated list of codecs which will not be used for SIP phone cases,[]codecs_exclude_sip_rtmp,opus,g729,g722,mpeg4-generic,vp8,mpv,String,null,false,null,codecs_exclude_sip_rtmp,Comma-separated list of codecs which will not be used for SIP as RTMP case,[]codecs_exclude_streaming,flv,telephone-event,String,null,false,null,codecs_exclude_streaming,Comma-separated list of codecs which will not be used for streaming,[]complex_test_config,,String,,false,null,complex_test_config,Complex transcoder test configuration,[]complex_test_decode,false,Boolean,false,false,null,complex_test_decode,Enable decoding during complex transcoding test,[]complex_test_fps,15,Integer,15,true,null,complex_test_fps,Complex transcoder test FPS,[]complex_test_replay,3,Integer,3,true,null,complex_test_replay,Complex transcoder test repeats count,[]complex_test_thread,3,Integer,3,true,null,complex_test_thread,Complex transcoder test threads count,[]core_standalone_web_dir,null,String,null,false,null,core_standalone_web_dir,Web directory for standalone mode,[]cost_header,cost,String,cost,false,null,cost_header,This SIP header will be sent to client as a call cost,[]cps_client,null,String,null,false,null,cps_client,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,1000,Long,1000,false,null,cps_interval,Time window for measuring CPS

cps_node

2147483647

Integer

false

Global CPS limitation for node

cpu_load_avg_size

20

Integer

true

CPU load average size

cpu_load_refresh

50

Integer

true

CPU load refresh rate

cpu_load_reject

false

Boolean

false

If true, reject streams when CPU load exceeds treshold,[]cpu_load_threshold,80,Integer,80,true,null,cpu_load_threshold,CPU load treshold,[]cpu_load_window,2000,Integer,2000,true,null,cpu_load_window,Timeslice to estimate CPU load,[]custom_ice_agent,true,Boolean,true,false,null,custom_ice_agent,If true

custom_stats_script

String

false

Script can be used to provide custom stat params with action=stat request

custom_watermark_filename

null

String

false

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

data_packet_decoder_fire_null_messages

true

Boolean

false

If true, pass special data packet up the RTP process chain when original received data failed to decode,[]datagram_channel_factory,NioDatagramChannelFactory,String,NioDatagramChannelFactory,true,null,datagram_channel_factory,NioDatagramChannelFactory

decoded_frame_interceptor

null

String

false

Full name of Java class that implements interface IDecodedFrameInterceptor. This class should be wrapped to .jar file placed in /usr/local/FlashphonerWebCallServer/lib directory

decoder_binary_log_enable

false

Boolean

false

Binary log decoder

decoder_binary_log_size

5

Integer

true

Binary log decoder size

decoder_buffer_pool

false

Boolean

true

Enable buffer pool usage during video decoding

decoder_buffer_pool_stats

false

Boolean

false

Enable buffer pool stats, might slow down video transcoding,[]decoder_mode,JNI,QUEUE
JNI,JNI,false,null,decoder_mode,Decoder mode,[]decoder_priority,FF,OPENH264,String,FF,OPENH264,false,null,decoder_priority,Decoder priority,[]decoder_stat_log,false,Boolean,false,false,null,decoder_stat_log,Enable decoder statistics logging,[]default_sdp_state,sendrecv,String,sendrecv,false,null,default_sdp_state,If SDP from SIP side comes without sendrecv

degraded_streams_threshold

20

Integer

true

Degraded streams threshold

degraded_streams_window

2000

Integer

true

Timeslice to estimate stream degradation

delta_threshold

100

Integer

false

RTMFP. If delta between UDP media packets is greater than the threshold, it will be reported,[]detect_flash_2_flash_calls,true,Boolean,true,false,null,detect_flash_2_flash_calls,If true

disable_drop_aac_frame

true

Boolean

false

If true, disables dropping AAC frames,[]disable_manager_rmi,true,Boolean,true,true,null,disable_manager_rmi,If true

disable_rest_auth

true

Boolean

false

If true, disables authorization in rest api,[]disable_rest_requests,false,Boolean,false,true,null,disable_rest_requests,If true

disable_rtc_ata

null

String

false

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,deprecateddisable_rtc_avoid_transcoding_alg,false,Boolean,false,false,null,disable_rtc_avoid_transcoding_alg,If true

disable_streaming_proxy

false

Boolean

false

If true, disable proxy and enable transcoding for all streams. For debug only,[]disable_streaming_proxy_aac,false,Boolean,false,false,null,disable_streaming_proxy_aac,If false

domain

null

String

false

SIP domain. If this parameter is set, it will redefine values that were transmitted during connection,[]dtls0_ua_match_substring,false,Boolean,false,false,null,dtls0_ua_match_substring,If true

dtls_close_socket_after_tries

10

Integer

false

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,false,Boolean,false,false,null,dtls_force_version_0,Force DTLS version 1.0,[]dtls_message_timeout,15,Integer,15,false,null,dtls_message_timeout,DTLS handshake timeout in seconds

dtls_socket_timeout_ms

1000

Integer

false

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,true,Boolean,true,false,null,dtls_use_socket_timeout,If true

dtmf

null

String

false

This type will be used if DTMF type (INFO, INFO_RELAY, RFC2833) was not specified when DTMF was sent,[]dump_avcc_relay,false,Boolean,false,false,null,dump_avcc_relay,If true

enable_candidate_harvester

false

Boolean

false

If true, gather ICE candidates using external STUN server,deprecatedenable_empty_shift_writer,false,Boolean,false,false,null,enable_empty_shift_writer,Enable empty shift writer for conference,deprecatedenable_extended_logging,true,Boolean,true,false,null,enable_extended_logging,When extended logging is enabled

enable_flight_recorder

false

Boolean

false

Enable flight recorder

enable_flight_recorder_test

false

Boolean

false

Enable flight recorder test

enable_local_videochat

false

Boolean

false

Not in use

enable_new_client_logger

true

Boolean

false

If true, enable new client logger,[]enable_rtc_video_generator,false,Boolean,false,false,null,enable_rtc_video_generator,Designed to avoid video negotiation issue in SIP cases. If true

enable_sip_stack_thread_audit

true

Boolean

false

If true, enable audit of SIP stack,[]enable_sync_time_normalizer,false,Boolean,false,true,null,enable_sync_time_normalizer,If true

encode_record_name

null

String

true

Encode record name setting

encoder_buffer_length_sec

1

Integer

false

Encoding buffer for audio, in seconds,[]encoder_default_video_resolution,640x480,String,640x480,false,null,encoder_default_video_resolution,encoder_default_video_resolution,[]encoder_mode,JNI,QUEUE
JNI,JNI,false,null,encoder_mode,Encoder mode,[]encoder_priority,FF,OPENH264,String,FF,OPENH264,false,null,encoder_priority,Encoder priority,[]encoder_stat_log,false,Boolean,false,false,null,encoder_stat_log,Enable encoder statistics logging,[]event_scanner_cached_pool,false,Boolean,false,false,null,event_scanner_cached_pool,If true

event_scanner_pool_size

10

Integer

false

Event scanner pool size

exclude_record_name_characters

null

String

true

Exclude characters from record name

fetch_caller_from_pai

false

Boolean

false

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,false,Boolean,false,false,null,fetch_caller_from_pai_set_from_if_empty,If true

file_recorder_thread_pool_max_size

4

Integer

true

Maximum core threads count in record thread pool

flash_codecs

speex16

ulaw

h264

alaw

flash_policy.port

843

Integer

true

Listening port for flash policy requests to crossdomain.xml file

flash_rtp_activity_enabled

false

Boolean

false

If true, enable RTP activity for Flash streams,[]flash_streaming_enable,true,Boolean,true,false,null,flash_streaming_enable,Not in use,deprecatedflight_recorder_capacity,500,Integer,500,false,null,flight_recorder_capacity,Flight recorders buffer capacity in records

flight_recorder_categories

NONE

NONE
WCS1438

true

Flight recorder categories

flush_audio_interval

80

Integer

true

RTMFP flush interval in milliseconds for flash-audio data from server

flush_video_interval

0

Integer

true

RTMFP flush interval in milliseconds for flash-video data from server

force_client_requested_video_resolution

true

Boolean

false

If true, use client-specified resolution passed in Stream object,[]force_expires,-1,Integer,-1,false,null,force_expires,If this parameter is set

force_local_audio_codec

null

String

false

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,true,Boolean,true,false,null,force_periodic_fir_request_for_sip_as_rtmp,If true

force_profile_level

null

String

false

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,null,String,null,false,null,force_rtmp_audio_codec,Forced codec for old as-RTMP cases using RTMPOutputWriter and for the latest HLS writer,[]force_sendrecv_for_outgoing_calls,false,Boolean,false,false,null,force_sendrecv_for_outgoing_calls,If true

generate_av_for_ua

null

String

true

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

0

Integer

true

Generator start delay in ms, 0 - no delay,[]get_callee_url,null,String,null,false,null,get_callee_url,Not in use,deprecatedglobal_bandwidth_check_enabled,false,Boolean,false,false,null,global_bandwidth_check_enabled,If true

h264_buffer_nack_list_threshold

30

Integer

false

JitterBuffer will be reset upon reaching this number of NACK packets

h264_check_and_skip_annexb

false

Boolean

false

Check and skip annexB magic bytes

h264_encoder_rc_buffer_size

2

Integer

false

Coefficient for rc buffer

h264_max_nalu_size

1346

Integer

true

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

false

Boolean

false

Not in use

h264_sps_buff_scale

1.6

Double

false

Buffer scale for H264 SPS

h264_sps_default_size

100

Integer

false

Default size of H264 sps buffer

h264_sps_rbsp_scale

1.5

Double

false

Buffer scale for H264 SPS RBSP

h264_strict_kframe_detect

false

Boolean

true

If true, set frame as keyframe only if contains SPS and PPS NAL units or IDR NAL,[]handler_async_disconnect,true,Boolean,true,false,null,handler_async_disconnect,If true

hangup_incoming_call_state

null

String

false

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,false,Boolean,false,false,null,hide_all,If true

hls.address

0.0.0.0

InetAddress[]

true

client that want's to get ABR version instead of ordinary version should append this suffix to original stream name

hls_access_control_headers

null

String

true

HLS response headers

hls_auth_enabled

false

Boolean

false

Enable check auth tokens for hls

hls_auth_token_cache

10

Integer

false

Timeout for cache auth tokens in seconds

hls_auto_start

false

Boolean

false

If true, enable HLS autostart,[]hls_dir,hls,String,hls,true,null,hls_dir,HLS base folder,[]hls_disable_cleanup,false,Boolean,false,false,null,hls_disable_cleanup,Do not remove inactive hls files from hdd,[]hls_discontinuity_enabled,false,Boolean,false,false,null,hls_discontinuity_enabled,If true

hls_enable_session_debug

false

Boolean

false

If true, enable debug logging for HLS session,[]hls_enabled,true,Boolean,true,false,null,hls_enabled,If true

hls_hold_segments_before_delete

false

Boolean

false

If true, hold segments on disk before delete,[]hls_hold_segments_size,5,Integer,5,false,null,hls_hold_segments_size,How many segments to hold

hls_list_size

10

Integer

false

Maximum number of segments in playlist

hls_manager_provider_timeout

300

Integer

false

HLS manager provider timeout

hls_manifest_file

index.m3u8

String

true

HLS master playlist file name. Default is 'index.m3u8'

hls_min_list_size

1

Integer

false

Minimum number of segments in playlist (should be less than 11)

hls_player_height

480

Integer

false

HLS player height

hls_player_width

640

Integer

false

HLS player width

hls_preloader_dir

hls/.preloader

String

false

HLS preloader dir

hls_preloader_enabled

true

Boolean

false

If true, enables HLS preloader,[]hls_preloader_time_min,2000,Long,2000,false,null,hls_preloader_time_min,Minimal size of preloaders HLS segment in milliseconds

hls_segment_name_suffix_randomizer_enabled

false

Boolean

false

HLS segment name suffix randomizer

hls_server_enabled

true

Boolean

true

If true, activate HLS server,[]hls_static_dir,client2/examples/demo/streaming/hls_static,String,client2/examples/demo/streaming/hls_static,false,null,hls_static_dir,HLS static dir,[]hls_static_enabled,false,Boolean,false,false,null,hls_static_enabled,If true

hls_store_segment_in_memory

false

Boolean

false

Store HLS segments in memory

hls_test_interval

182000

Integer

true

HLS test interval

hls_test_run_for

180

Integer

true

HLS test duration in seconds

hls_test_start_streams

10

Integer

true

HLS test streams count

hls_test_start_writers

10

Integer

true

HLS test writers count

hls_time

4

Integer

false

Size of one HLS segment in seconds

hls_time_min

2000

Long

false

Minimal size of one HLS segment in milliseconds

hls_version

8

Integer

false

HLS version

hls_wrap

20

Integer

false

Maximum number of ts-files. The option is necessary to prevent disc overflow

http.address

0.0.0.0

InetAddress[]

true

https.addresses

https.port

8444

Integer

true

WCS server HTTPS port

https_server_enabled

true

Boolean

true

If true, activate HTTPS server,[]ice_add_ipv6_candidate,false,Boolean,false,false,null,ice_add_ipv6_candidate,If true

ice_authorize_by_address

false

Boolean

false

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,true,Boolean,true,false,null,ice_consent_freshness,If true

ice_keep_alive_enabled

true

Boolean

false

If true, enables ICE keep-alive,[]ice_keep_alive_timeout,15,Integer,15,false,null,ice_keep_alive_timeout,ICE establishing timeout in seconds. By default

ice_tcp_channel_high_water_mark

52428800

Integer

true

High watermark for ICE tcp channels

ice_tcp_channel_low_water_mark

5242880

Integer

true

Low watermark for ICE tcp channels

ice_tcp_nio

true

Boolean

false

If true, use NIO for ICE tcp channels,[]ice_tcp_receive_buffer_size,1048576,Integer,1048576,true,null,ice_tcp_receive_buffer_size,Receive buffer size for ice tcp channels,[]ice_tcp_send_buffer_size,1048576,Integer,1048576,true,null,ice_tcp_send_buffer_size,Send buffer size for ice tcp channels,[]ice_tcp_transport,false,Boolean,false,false,null,ice_tcp_transport,If true

ice_tcp_transport_force

false

Boolean

false

If true, use tcp transport regardless of client config,[]ice_timeout,15,Integer,15,false,null,ice_timeout,ICE keep-alive timeout in seconds. By default

ice_transport_new

true

Boolean

false

If true, use new udp transport,deprecatedice_udp_nio,true,Boolean,true,false,null,ice_udp_nio,If true

ice_udp_transport_new

true

Boolean

false

If true, use new udp transport,deprecatedignore_incoming_call_if_sip_login_port_does_not_match_request_uri,false,Boolean,false,false,null,ignore_incoming_call_if_sip_login_port_does_not_match_request_uri,If true

in_jitter_buffer_enabled

false

Boolean

false

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,deprecatedinbound_video_rate_stat_send_interval,1,Integer,0,false,null,inbound_video_rate_stat_send_interval,Inbound video rate stat send interval in sec needed to calculate channel quality. 0 - disabled,[]increase_equals_timestamp,100,Integer,100,false,null,increase_equals_timestamp,Timestamps are equal within this interval in milliseconds,[]ip,95.191.130.39,String,0.0.0.0,true,null,ip,External IPv4 address. This IP address will differ from specified with ip_local option when WCS server is behind NAT,[]ip_local,95.191.130.39,String,0.0.0.0,true,null,ip_local,WCS server will create sockets and listen on this interface,[]ip_v6,,String,,true,null,ip_v6,External IPv6 address. This IP address will differ from specified with ip_local option when WCS server is behind NAT,[]jitter_threshold,50,Integer,50,false,null,jitter_threshold,RTMFP. If jitter between UDP media packets is greater than the threshold

jni_cache_class

true

Boolean

false

If true, cache JNI Class object,deprecatedkeep_alive.algorithm,HIGH_LEVEL,INTERNAL
NONE
HIGH_LEVEL,HIGH_LEVEL,true,null,keep_alive.algorithm,Keep-alive algorithm: INTERNAL

keep_alive.enabled

String

rtmfp

websocket

null

keep_alive.peer_interval

2000

Integer

true

Keep-alive peer interval (Not in use)

keep_alive.probes

10

Integer

true

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,5000,Integer,5000,true,null,keep_alive.server_interval,Interval in milliseconds between attempts to ping connected client (WebSocket

keep_alive_streaming_sessions_enabled

false

Boolean

true

If true, server sends keep-alive REST requests to check if stream playback is allowed to continue / resume,[]kill_event_scanner,false,Boolean,false,false,null,kill_event_scanner,Debug option

load_balancing_acao_header

String

true

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

false

Boolean

true

If true, activate loadbalancer,[]manager_http_ports_enabled,true,Boolean,true,true,null,manager_http_ports_enabled,If true

max_callid_length

32

Integer

false

Maximum length of SIP callID. If the length of generated callID exceeds this value, it will be cut to this length,[]max_drop_rate,null,String,null,false,null,max_drop_rate,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,deprecatedmax_queue_size,null,String,null,false,null,max_queue_size,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,deprecatedmedia_dir,media,String,media,true,null,media_dir,Media base folder,[]media_port_from,10001,Integer,31001,true,null,media_port_from,Beginning of media ports range for ICE

media_port_stress_test_iterations

1

Integer

false

Media port stress test iterations

media_port_stress_test_thread_sleep

5

Integer

false

Media port stress test thread sleeping interval

media_port_stress_test_threads

5

Integer

false

Media port stress test threads count

media_port_to

32000

Integer

true

End of media ports range for ICE, RTP, SRTP, RTCP,[]media_ports_auditor_interval,5000,Integer,5000,true,null,media_ports_auditor_interval,Audit interval for busy and free ports

media_ports_auditor_max_attempts

3

Integer

true

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

min_drop_rate

null

String

false

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

null

String

false

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

1

Integer

false

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

60000

Integer

false

If there is no streams added to mixer within this timeout in milliseconds, corresponding mixer will be terminated,[]mixer_app_name,defaultApp,String,defaultApp,false,null,mixer_app_name,AppName for mixer streams,[]mixer_audio_enabled,true,Boolean,true,false,null,mixer_audio_enabled,When false

mixer_audio_only_height

360

Integer

true

Height constraint for mixer audio only frame

mixer_audio_only_width

640

Integer

true

Width constraint for mixer audio only frame

mixer_audio_silence_threshold

-50.0

Double

false

Audio silence threshold in db

mixer_auto_create_delimiter

#

String

false

Mixer auto create stream/room delimiter

mixer_auto_start

true

Boolean

false

If true, enable mixer autostart,[]mixer_autoscale_desktop,true,Boolean,true,false,null,mixer_autoscale_desktop,Separate screen share font size from other frames,[]mixer_debug_mode,false,Boolean,false,false,null,mixer_debug_mode,Turns on debug mode

mixer_desktop_align

TOP

TOP
BOTTOM
LEFT
RIGHT
CENTER

false

Alignment of screen sharing stream

mixer_display_stream_name

false

Boolean

false

Output stream name to mixer's canvas

mixer_font_size

20

Integer

false

Font size for stream name and debug info

mixer_font_size_audio_only

40

Integer

false

Font size for stream name and debug info for audio only streams

mixer_idle_timeout

60000

Long

false

Mixer idle timeout in milliseconds

mixer_in_buffering_ms

200

Integer

false

How much stream should be buffered before it gets into mix

mixer_incoming_time_rate_lower_threshold

0.95

Double

false

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,1.05,Double,1.05,false,null,mixer_incoming_time_rate_upper_threshold,Relation between incoming stream time and actual machine mixing time

mixer_layout_class

com.flashphoner.media.mixer.video.presentation.GridLayout

String

true

Name of class for custom mixer layout

mixer_lossless_video_processor_enabled

false

Boolean

false

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,200,Integer,200,false,null,mixer_lossless_video_processor_max_mixer_buffer_size_ms,Max size that is allowed for mixer's incoming buffer

mixer_lossless_video_processor_wait_time_ms

20

Integer

false

How long to wait before checking mixer's incoming buffer again in case it was full

mixer_mcu_audio

false

Boolean

false

Enable mcu like audio mixing, each added stream will have dedicated audio mix available as a separate stream,[]mixer_mcu_video,false,Boolean,false,false,null,mixer_mcu_video,Works only with mcu audio

mixer_minimal_font_size

1

Integer

false

Minimal font size for stream name if autoscaling is on

mixer_out_buffer_enabled

false

Boolean

false

If true, enable buffer for out mixer streams,[]mixer_out_buffer_initial_size,2000,Long,2000,false,null,mixer_out_buffer_initial_size,Initial size of output mixer buffer in milliseconds,[]mixer_out_buffer_max_bufferings_allowed,-1,Integer,-1,false,null,mixer_out_buffer_max_bufferings_allowed,mixer_out_buffer_max_bufferings_allowed,[]mixer_out_buffer_polling_time,100,Long,100,false,null,mixer_out_buffer_polling_time,Output mixer buffer polling time in milliseconds,[]mixer_out_buffer_start_size,150,Long,150,false,null,mixer_out_buffer_start_size,Start size of output mixer buffer in milliseconds,[]mixer_prune_streams,false,Boolean,false,false,null,mixer_prune_streams,When true

mixer_realtime

true

Boolean

false

Turns on realtime version of mixer

mixer_show_separate_audio_frame

true

Boolean

false

Show audio frame for audio+video stream if added with hasVideo: false

mixer_text_autoscale

true

Boolean

false

Enable stream name autoscaling

mixer_text_cut_top

3

Integer

false

Clip top part of the text

mixer_text_padding_bottom

5

Integer

false

Padding for the bottom side of text in pixels

mixer_text_padding_left

5

Integer

false

Padding for the left side of text in pixels

mixer_text_padding_right

4

Integer

false

Padding for the right side of text in pixels

mixer_text_padding_top

5

Integer

false

Padding for the top side of text in pixels

mixer_thread_priority

5

Integer

false

Mixer thread priority, min 1 max 10,[]mixer_thread_timeout_ms,33,Integer,33,false,null,mixer_thread_timeout_ms,Mixer thread timeout,[]mixer_use_sdp_state,true,Boolean,true,false,null,mixer_use_sdp_state,Enable audio/video only stream detection via sdp state,[]mixer_video_background_filename,null,String,null,false,null,mixer_video_background_filename,Mixer video background. Example: background.png,[]mixer_video_bitrate_kbps,2000,Integer,2000,false,null,mixer_video_bitrate_kbps,Encoded video bitrate kbps,[]mixer_video_buffer_length,1000,Integer,1000,false,null,mixer_video_buffer_length,Video buffer length for decoded frames,[]mixer_video_desktop_layout_inline_padding,10,Integer,10,false,null,mixer_video_desktop_layout_inline_padding,Padding between video streams in bottom row (under screen sharing stream),[]mixer_video_desktop_layout_padding,30,Integer,30,false,null,mixer_video_desktop_layout_padding,Padding between top row (screen sharing stream) and bottom row (other streams),[]mixer_video_enabled,true,Boolean,true,false,null,mixer_video_enabled,When false

mixer_video_fps

30

Integer

false

Fps constraint for mixer stream

mixer_video_grid_layout_middle_padding

10

Integer

false

Padding between video streams in one row (when there is no screen sharing stream)

mixer_video_grid_layout_padding

30

Integer

false

Padding between rows of video streams (when there is no screen sharing stream)

mixer_video_height

720

Integer

false

Height constraint for mixer stream

mixer_video_layout_desktop_key_word

desktop

String

false

Keyword for screen sharing streams

mixer_video_profile_level

42c02a

String

false

Mixer video profile and level in hex. Example: 42c02a

mixer_video_quality

24

Integer

false

Encoded video quality (CRF)

mixer_video_stable_fps_threshold

15

Integer

false

Streams with fps lower then threshold won't trigger buffering of the stream if video buffer was exhausted

mixer_video_width

1280

Integer

false

Width constraint for mixer stream

mixer_voice_activity

true

Boolean

false

Enable/disable voice activity frame

mixer_voice_activity_frame_position_inner

false

Boolean

false

Draw voice activity frame inside the frame. If false - draw around the frame

mixer_voice_activity_frame_thickness

6

Integer

false

Thickness of voice activity frame

mp4_container_moov_first

true

Boolean

false

When recording mp4 write moov atom first so recording can be played/downloaded progressively

mp4_container_moov_first_reserve_space

false

Boolean

false

Turn on space reservation for moov atom to avoid additional filesystem copy

mp4_container_moov_reserved_space_size

2048

Integer

false

When writing moov first how much space should be reserved for moov atom in kilobytes

mpeg1.gop_size

60

Integer

false

GOP size or k-frame interval

mpeg1.qmax

24

Integer

false

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,4,Integer,4,false,null,mpeg1.qmin,Minimum value of quality parameter. The lower the value

mpeg1.trellis

0

Integer

false

Trellis quantization

msrp_port

2855

Integer

false

Port for receiving MSRP / TCP connections

multipart_message_service_uri

null

String

false

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

native_test_aac

true

Boolean

true

If true, enable AAC native test,[]native_test_decoder,true,Boolean,true,true,null,native_test_decoder,If true

native_test_encoder

true

Boolean

true

If true, enable encoder native test,[]native_test_opus,true,Boolean,true,true,null,native_test_opus,If true

native_test_resampler

true

Boolean

true

If true, enable native test resampler,[]native_test_run_for,180,Integer,180,true,null,native_test_run_for,Native test duration,[]native_test_start_threads,10,Integer,10,true,null,native_test_start_threads,Native test threads count,[]native_test_thread_interval,200,Integer,200,true,null,native_test_thread_interval,Native test interval,[]netty_deadlock_aware_server_workers,true,Boolean,true,false,null,netty_deadlock_aware_server_workers,If true

netty_deadlock_aware_worker_timeout

10000

Integer

false

Timeout to detect SSL connection with Netty deadlock

no_media_dump_interval

15000

Long

false

Period in milliseconds, within which media traffic should be captured by tcpdump when client sends bug report with no_media type,[]notify_message_call_timeout,null,String,null,false,null,notify_message_call_timeout,Timeout in milliseconds to wait for client confimation of receiving an incoming message.
When an incoming message is received

on_record_hook_script

on_record_hook.sh

String

false

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,null,String,null,false,null,options2flash_delegate,If true

opus.encoder.bitrate

-1

Integer

false

Target bitrate for Opus encoder, in bps,[]opus.encoder.complexity,-1,Integer,-1,false,null,opus.encoder.complexity,Target complexity for Opus encoder,[]opus_formats,null,String,null,false,null,opus_formats,Comma-separated list of Opus formats (name=value).
Example: maxaveragebitrate=20000.
These formats will be listed in SDP,[]order_threads_by_seq,true,Boolean,true,false,null,order_threads_by_seq,If true

out_jitter_buffer_enabled

null

String

false

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,deprecatedoutbound_port,null,String,null,false,null,outbound_port,SIP port. If this parameter is set

outbound_proxy

null

String

false

SIP outbound proxy. If this parameter is set, it will redefine values that were transmitted during connection,[]outbound_video_rate_stat_send_interval,1,Integer,0,false,null,outbound_video_rate_stat_send_interval,Outbound video rate stat send interval in sec needed to calculate channel quality. 0 - disabled,[]parse_system_stats,false,Boolean,false,false,null,parse_system_stats,If true

periodic_fir_request

false

Boolean

false

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

periodic_fir_request_interval

5000

Integer

false

Interval to send RTCP FIR in milliseconds

play_stream_force_video_orientation

true

Boolean

false

Force negotiation of 3gpp video orientation extension for play stream requests

port_from

30000

Integer

false

Beginning of range of ports for SIP signaling

port_to

31000

Integer

false

End of range of ports for SIP signaling

preserve_non_mixed_recorded_files

false

Boolean

false

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,false,Boolean,false,false,null,print_publication_tables,RTMFP. If true

print_rtcp_stats

false

Boolean

false

If true, print RTCP report on end of session,deprecatedpriority_outside_codecs,false,Boolean,false,false,null,priority_outside_codecs,If true

process_remote_sdp_candidates

true

Boolean

false

If true, process candidates from SDP,[]profiles,42e01f,String,42e01f,false,null,profiles,Comma-separated list of H.264 profiles. These profiles will be used in SDP for video calls,[]proxy_propagate_fir,false,Boolean,true,false,null,proxy_propagate_fir,Propagate FIR requests through proxy,[]proxy_use_h264_packetization_mode_1_only,true,Boolean,true,false,null,proxy_use_h264_packetization_mode_1_only,If true

ptime

20

Integer

false

Packetization time. Use carefully

ptime_corrector_enabled

true

Boolean

false

Enabling corrector by required packetization time

publication_report_format

null

String

false

RTMFP. Sets format for statistics.
Possible value: csv

pull_streams

null

String

true

Comma separated list of urls to pull from at server startup

queue_ping_period

2000

Integer

true

Queue ping interval in ms

queue_stat_log

true

Boolean

false

Enable queue statistics logging

queue_transcoder_core_router_uri

tcp://127.0.0.1:5555

String

false

Queue transcoder core router URI

queue_transcoder_receive_timeout

500

Integer

true

Queue transcoder receive timeout

queue_transcoder_shm_path

/dev/shm/

String

false

Path to shared memory objects for queue transcoder

queue_transcoder_shm_size

5

Integer

true

Shared memory object size for queue transcoder

queue_transcoder_transmit_timeout

500

Integer

true

Queue transcoder transmit timeout

queue_transcoder_worker_router_uri

ipc:///tmp/flashphoner.pipe

String

false

Queue transcoder core router URI

record

null

String

false

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,100,Integer,100,false,null,record_audio_buffer_max_size,Record audio buffer size,[]record_audio_codec_channels,2,Integer,2,false,null,record_audio_codec_channels,Codec channel count used for recording streams,[]record_audio_codec_sample_rate,44100,Integer,44100,false,null,record_audio_codec_sample_rate,Codec sample rate used for recording streams,[]record_audio_processor_pcm,false,Boolean,false,false,null,record_audio_processor_pcm,If true

record_close_scheduling_period

20

Integer

true

Buffer check period for closing a record in milliseconds

record_dir

records

String

true

Record base folder

record_fdk_aac_bitrate_mode

5

Integer

false

Record FDK bitrate mode. 0 - CBR, 1-5 - VBR,record_filename_template,null,String,null,false,null,record_filename_template,Filename template for an audio call recording. Besides the default fields

record_flash_published_streams

false

Boolean

false

If true, record streams published from native Flash clients and RTMP live encoders such as Wirecast, FFmpeg, FMLE, etc.,[]record_h264_to_ts,false,Boolean,false,false,null,record_h264_to_ts,If set

record_mixer_streams

false

Boolean

false

When true, mixer streams are recorded,[]record_response_content_disposition_header_value,null,String,null,false,null,record_response_content_disposition_header_value,/client/records/ path content-disposition header,[]record_rotation,null,String,null,false,null,record_rotation,If set

record_rotation_index_enabled

true

Boolean

false

If true, rotation for stream recording files is enabled,[]record_rtsp_streams,false,Boolean,false,false,null,record_rtsp_streams,If true

record_stop_timeout

15

Integer

false

Record stop timeout in seconds

record_streams

true

Boolean

false

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,[]recording_by_user,false,Boolean,false,true,null,recording_by_user,If true

reg_expires

3600

Integer

false

Value in seconds, which will be used in Expires header when SIP REGISTER request is sent,[]remove_ssrc_attr,null,Boolean,null,false,null,remove_ssrc_attr,If true

replace_cached_pool_with_default_pool

false

Boolean

true

If true, replaces cached thread pool with default,[]resample_video,true,Boolean,true,false,null,resample_video,If true

rest_access_control_allow_credentials

true

Boolean

false

Rest-api response access_control_allow_credentials header

rest_access_control_allow_headers

String

x-requested-with

content-type

null

rest_access_control_allow_methods

POST

String

false

Rest-api response access_control_allow_methods header

rest_access_control_allow_origin

*

String

false

Rest-api response access_control_allow_origin header

rest_access_control_headers

null

String

true

REST response headers

rest_hook_secret_key

null

String

false

Rest hook secret key

rest_hook_send_hash

false

Boolean

false

Rest hook send hash

rest_max_connections

200

Integer

true

Rest max connextions

rest_request_timeout

15

Integer

true

Rest request timeout in seconds

rfc2833_packets_count

null

String

false

Number of RTP packets for sending one DTMF

rmi.port

1098

Integer

true

Internal RMI port for communications with WCS Manager

rtc_ice_add_local_component

true

Boolean

false

If true, add local component for ICE candidates,[]rtc_ice_add_local_interface,false,Boolean,false,false,null,rtc_ice_add_local_interface,If true

rtc_ip

null

String

false

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

null

String

false

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

1000

Long

false

Minimal waiting time to send PLI after receiving K-frame

rtcp_sender_interval

0.1

Double

false

Guard RTCP interval based on the specified fraction of RTCP bitrate

rtmfp.address

0.0.0.0

InetAddress[]

true

UDP

rtmp.address

0.0.0.0

InetAddress[]

true

TCP

rtmp.server_buffer_enabled

false

Boolean

false

Enable/disable buffering rtmp data on java's heap if socket buffer is full

rtmp.server_channel_high_water_mark

52428800

Integer

true

High watermark for connected rtmp channels

rtmp.server_channel_low_water_mark

5242880

Integer

true

Low watermark for connected rtmp channels

rtmp.server_channel_send_buffer_size

1048576

Integer

true

Send buffer size for rtmp channels

rtmp.server_read_socket_timeout

0

Integer

true

TCP socket write timeout for RTMP server, in seconds,[]rtmp.server_socket_timeout,0,Integer,0,true,null,rtmp.server_socket_timeout,TCP socket write and read timeout for RTMP server for

rtmp.server_write_socket_timeout

0

Integer

true

TCP socket write timeout for RTMP server, in seconds,[]rtmp.use_server_socket_timeout,false,Boolean,false,true,null,rtmp.use_server_socket_timeout,DEPRECATED (use rtmp.server_socket_timeout

rtmp_activity_timer_cool_off_period

1

Integer

false

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

60000

Integer

false

If there is no subscribers for an RTMP stream within this timeout in milliseconds, corresponding RTMP session will be terminated,[]rtmp_appkey_source,default,String,default,false,null,rtmp_appkey_source,RTMP appkey source: default/app,[]rtmp_command_amf3,true,Boolean,true,true,null,rtmp_command_amf3,rtmp_command_amf3,[]rtmp_flash_ver_publisher,FMLE/3.0,String,FMLE/3.0,false,null,rtmp_flash_ver_publisher,RTMP publisher Flash version,[]rtmp_flash_ver_subscriber,LNX 9,0,124,2,String,LNX 9,0,124,2,false,null,rtmp_flash_ver_subscriber,RTMP subscriber Flash version,[]rtmp_in_buffer_enabled,false,Boolean,false,false,null,rtmp_in_buffer_enabled,If true

rtmp_in_buffer_initial_size

2000

Long

false

Initial size of incoming RTMP buffer in milliseconds

rtmp_in_buffer_max_bufferings_allowed

-1

Integer

false

rtmp_in_buffer_max_bufferings_allowed

rtmp_in_buffer_polling_time

100

Long

false

Incoming RTMP buffer polling time in milliseconds

rtmp_in_buffer_start_size

300

Long

false

Start size of incoming RTMP buffer in milliseconds

rtmp_metadata_to_sdp_state

true

Boolean

false

Translate publishers metadata into sdp state

rtmp_out_buffer_enabled

false

Boolean

false

If true, enable buffer for outgoing RTMP streams,[]rtmp_out_buffer_initial_size,2000,Long,2000,false,null,rtmp_out_buffer_initial_size,Initial size of outgoing RTMP buffer in milliseconds,[]rtmp_out_buffer_max_bufferings_allowed,-1,Integer,-1,false,null,rtmp_out_buffer_max_bufferings_allowed,rtmp_out_buffer_max_bufferings_allowed,[]rtmp_out_buffer_polling_time,50,Long,50,false,null,rtmp_out_buffer_polling_time,Outgoing RTMP buffer polling time in milliseconds,[]rtmp_out_buffer_start_size,300,Long,300,false,null,rtmp_out_buffer_start_size,Start size of outgoing RTMP buffer in milliseconds,[]rtmp_output_writer_bufsize,0,Integer,0,false,null,rtmp_output_writer_bufsize,Buffer time for FFRtmpOutputWriter old outbound buffer for as-RTMP cases,deprecatedrtmp_port_from,33001,Integer,33001,false,null,rtmp_port_from,First port in RTMP ports range

rtmp_port_to

34000

Integer

false

Last port in RTMP ports range, for RTMP republisher,[]rtmp_ports_auditor_interval,10000,Integer,10000,false,null,rtmp_ports_auditor_interval,Audit interval for RTMP ports

rtmp_ports_auditor_max_attempts

3

Integer

false

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

String

true

IPv4 address for outgoing RTMP publishing

rtmp_publisher_start_time_ts

1000

Long

false

RTMP publisher start time

rtmp_pull_agent_account_for_lost_audio

false

Boolean

false

If true, enable RTMP pull agent account for lost audio,[]rtmp_pull_rtp_activity_detection,true,Boolean,true,false,null,rtmp_pull_rtp_activity_detection,If true

rtmp_push_auto_start

false

Boolean

false

If true, enable RTMP push autostart for newly published streams,[]rtmp_push_auto_start_url,null,String,null,false,null,rtmp_push_auto_start_url,RTMP server address to auto start pushing to,[]rtmp_push_restore,false,Boolean,false,false,null,rtmp_push_restore,If true

rtmp_push_restore_attempts

3

Integer

false

RTMP push reconnect attempts

rtmp_push_restore_interval_ms

5000

Integer

false

RTMP push reconnect interval in ms

rtmp_receive_buffer_size_predictor_factory

2053

Integer

true

RTMP receive buffer size predictor factory in bytes

rtmp_send_video_first

false

Boolean

true

Send video first in RTMP

rtmp_server_channel_receive_buffer_size

0

Integer

true

RTMP receive buffer size in bytes

rtmp_transponder_force_kframe_interval

true

Boolean

false

If true, force k-frame interval for transponder in latest cases as-RTMP'. It is implemented sending RTCP PLI

rtmp_transponder_full_url

false

Boolean

false

If true, ignore streamName and use full rtmpUrl in transponders and as RTMP' cases.
If false

rtmp_transponder_kframe_interval

60

Integer

false

Forced k-frame interval in frames. See also rtmp_transponder_force_kframe_interval= setting.

rtmp_transponder_metadata

null

String

false

RTMP transponder metadata

rtmp_transponder_send_metadata

false

Boolean

false

If true, RTMP transponder will send metadata,[]rtmp_transponder_stream_name_prefix,,String,rtmp_,false,null,rtmp_transponder_stream_name_prefix,The specified prefix is added for all as-RTMP stream names. By default

rtmp_use_stream_params_as_connection

false

Boolean

true

Use stream params as connection

rtp_activity_audio

true

Boolean

false

If true, RTP activity check is enabled for audio.,[]rtp_activity_detecting,null,String,null,false,null,rtp_activity_detecting,Disables / enables and sets value of RTP activity timeout

rtp_activity_timeout

60

Long

false

RTP activity timer in seconds

rtp_activity_video

true

Boolean

false

If true, RTP activity check is enabled for video.
If false, this check is enabled for audio only,[]rtp_bundle,true,Boolean,true,false,null,rtp_bundle,Enable rtp bundle,[]rtp_elapsed_time_threshold,10000,Long,10000,false,null,rtp_elapsed_time_threshold,RTP elapsed time threshold

rtp_in_buffer_initial_size

2000

Integer

false

Initial size of incoming RTP buffer in milliseconds

rtp_in_buffer_polling_time

100

Long

false

Incoming RTP buffer polling time in milliseconds

rtp_in_reset_marker

false

Boolean

false

If true, use RTP in reset marker,[]rtp_paced_sender,false,Boolean,false,false,null,rtp_paced_sender,If true

rtp_paced_sender_capacity

200000000

Long

false

RTP paced sender capacity

rtp_paced_sender_increase_interval

50

Integer

false

Paced sender increase interval

rtp_paced_sender_initial_rate

200000

Integer

false

Paced sender initial rate

rtp_paced_sender_k_deviation

0.02

Double

false

Paced sender K deviation

rtp_paced_sender_k_down

0.02

Double

false

Paced sender K down

rtp_paced_sender_k_up

0.04

Double

false

Paced sender K up

rtp_paced_sender_period

1000

Long

false

RTP paced sender period

rtp_paced_sender_queue_size

2000

Integer

false

Outgoing queue maximum size

rtp_paced_sender_refill

200000000

Long

false

RTP paced sender refill

rtp_packet_cache_size

250

Integer

false

Cache size for sent packets. This is used only on video sessions to provide response to NACK requests

rtp_receive_buffer_predicator_size

1500

Integer

false

DatagramSocket constructing: receiveBufferSizePredictorFactory size

rtp_receive_buffer_size

65536

Integer

false

Buffer size for incoming RTP and SRTP (WebRTC).
DatagramSocket constructing: receiveBufferSize

rtp_send_buffer_size

65536

Integer

false

Buffer size for outgoing RTP and SRTP (WebRTC).
DatagramSocket constructing: sendBufferSize

rtp_session_init_always

false

Boolean

false

If true init rtp session for all media providers

rtsp.address

0.0.0.0

InetAddress[]

true

corresponding RTSP session will be terminated

rtsp_auth_cnonce

1234567890

String

true

RTSP server port

rtsp_client_address

0.0.0.0

InetAddress

true

RTSP client address

rtsp_client_strip_audio_codecs

null

String

false

Comma-separated list of audio codecs which will not be used for RTSP

rtsp_fail_on_error_track

true

Boolean

true

If true, RTSP pulling fails on error in any track,[]rtsp_in_buffer,false,Boolean,false,false,null,rtsp_in_buffer,If true

rtsp_interleaved_channels

null

String

false

Interleaved mode channels: audio channels;video channels. Default: dynamic channels

rtsp_interleaved_enable_rtcp

true

Boolean

false

If true, enable replying to RTCP packets on the RTSP interleaved channel,deprecatedrtsp_interleaved_mode,true,Boolean,true,false,null,rtsp_interleaved_mode,If true

rtsp_pcap_server_handler_redirect_url

null

String

true

Rtsp pcap server redirect URL

rtsp_pcap_server_redirect_method

OPTIONS

String

true

Rtsp pcap server redirect method: OPTIONS/DESCRIBE

rtsp_port_from

32001

Integer

false

First TCP port in the port range for RTSP pooling agent

rtsp_port_to

33000

Integer

false

Last TCP port in the port range for RTSP pooling agent

rtsp_ports_auditor_interval

10000

Integer

false

Audit interval for RTSP ports, in milliseconds,[]rtsp_ports_auditor_max_attempts,3,Integer,3,false,null,rtsp_ports_auditor_max_attempts,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,5,Integer,5,false,null,rtsp_refresh_requests_limit,Maximum number of non-answered GET_PARAMETER refresh requests. Stop sending refresh requests if the limit has been reached,[]rtsp_server_auth_enabled,false,Boolean,false,false,null,rtsp_server_auth_enabled,If true

rtsp_server_enabled

true

Boolean

true

If true, activate RTSP server,[]rtsp_server_forse_interleave,false,Boolean,false,false,null,rtsp_server_forse_interleave,If true

rtsp_server_packetization_mode

null

String

false

H.264 packetization mode for RTSP server. FU-A by default

rtsp_server_profile_level_id

null

String

false

H.264 profile-level-id for RTSP server

rtsp_user_agent

String

false

User agent indicated in RTSP packets

rvg_timer_activity

500

Integer

false

RVG timer interval in milliseconds

rvg_timer_delay

500

Integer

false

RVG timer initial delay in milliseconds

scheduling_service_core_threads

5

Integer

true

Core threads count for scheduling service

send_receive_buffer_size

1600

Integer

true

RTMFP buffer size in bytes

send_receive_on_incoming_re_invite

true

Boolean

false

If true, send receive' on incoming re-INVITE

session_idle_timeout

300000

Integer

true

RTMFP server-side timeout in milliseconds if no UDP messages received over RTMFP/UDP session

sessions_auditor_interval

60000

Integer

true

Audit interval for pending media sessions

sessions_auditor_session_timeout

60000

Integer

true

Audit timeout for pending media sessions

set_sync_time_from_ts

false

Boolean

false

Workaround for SIP audio only

sip.pre_init

true

Boolean

true

If true, use SIP pre-initiation,[]sip_add_contact_id,true,Boolean,true,false,null,sip_add_contact_id,If true

sip_as_rtmp_java_client

true

Boolean

false

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,live,String,live,false,null,sip_as_rtmp_stream_type,Sets RTMP AMF stream type for as-RTMP cases,[]sip_auditor_dialog_timeout,10000,Integer,10000,false,null,sip_auditor_dialog_timeout,SIP auditor dialog timeout,[]sip_auditor_transaction_timeout,50000,Integer,50000,false,null,sip_auditor_transaction_timeout,SIP auditor transaction timeout,[]sip_dns_failover,false,Boolean,false,false,null,sip_dns_failover,If true

sip_force_rtcp_feedback

false

Boolean

false

If true, force rtcp feedback to sip provider,[]sip_force_session_expires,1800,Integer,1800,false,null,sip_force_session_expires,Forced session expiration timeout in seconds. WCS server will send refresh request before the timeout is reached,[]sip_force_tcp,false,Boolean,false,false,null,sip_force_tcp,If true

sip_invite_params_to_headers

false

Boolean

false

If true, place SIP INVITE parameters to headers,[]sip_msg_listener,com.flashphoner.sdk.sip.ChangeCallIdListener,String,com.flashphoner.sdk.sip.ChangeCallIdListener,false,null,sip_msg_listener,Full name of Java class that implements interface ISipMessageListener
public interface ISipMessageListener {
void processMessage(SIPMessage sipMessage);
},[]sip_ports_auditor_interval,10000,Integer,10000,false,null,sip_ports_auditor_interval,Audit interval for SIP ports

sip_ports_auditor_max_attempts

3

Integer

false

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

false

Boolean

false

If true, record SIP as RTMP stream and SIP as stream,[]sip_remove_video_sdp_section_instead_of_adding_inactive_with_zero_port,false,Boolean,false,false,null,sip_remove_video_sdp_section_instead_of_adding_inactive_with_zero_port,If true

sip_sdp_unsupported_protocols

UDP/BFCP

UDP/UDT/IX

String

UDP/BFCP

sip_session_expires_header

true

Boolean

false

If true, use Expires header,[]sip_single_route_only,false,Boolean,false,false,null,sip_single_route_only,If true

sip_srv_lookup

false

Boolean

false

If true, enable DNS SRV lookup.
See also sip_dns_failover= option,[]sip_thread_pool_size,null,String,null,false,null,sip_thread_pool_size,Size of SIP thread pool,[]sip_timer,null,String,null,false,null,sip_timer,Value of timer T1 according to RFC 3261

sip_traffic_class

null

String

false

QoS class for SIP traffic

sip_use_netty

false

Boolean

false

If true, use Netty,[]sip_use_reentrant_listener,false,Boolean,false,false,null,sip_use_reentrant_listener,If true

sip_use_tls

false

Boolean

false

If true, TLS used for SIP connections,[]sip_user_agent_shutdown_timeout,5000,Integer,5000,false,null,sip_user_agent_shutdown_timeout,Timeout for remove sip user agent for unregister in sip provider. Default is 5000 ms,[]snapshot_auto_dir,/usr/local/FlashphonerWebCallServer/snapshots,String,/usr/local/FlashphonerWebCallServer/snapshots,false,null,snapshot_auto_dir,Snapshots dir,snapshot_auto_enabled,false,Boolean,false,false,null,snapshot_auto_enabled,If true

snapshot_auto_naming

mediaSessionId

String

false

Snapshot auto naming

snapshot_auto_rate

60

Integer

false

Snapshot rate. By default save every 60 frame

snapshot_auto_retention

20

Integer

false

Snapshot retention. By default keep last 20 frames

speex_g711_speex_transcoding

false

Boolean

false

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,deprecatedspeex_in_policy,null,String,null,false,null,speex_in_policy,Speex encoding settings used in transcoding featuring the codec.
Default:
8 - Quality
false - VBR encoding
8 - Quality of VBR
4 - Algorithmic complexity,[]start_test,false,Boolean,false,false,null,start_test,If true

stats

false

Boolean

true

If true, enable sampling for streams. The sampling is used for charts,[]stats_average_calculation_window,30,Integer,30,true,null,stats_average_calculation_window,Window size for general average stats calculation,[]stats_bitrate_window,1000,Integer,1000,false,null,stats_bitrate_window,Window size to collect bitrate statistics,[]stats_fps_window,1000,Integer,1000,false,null,stats_fps_window,Window size to collect FPS statistics,[]stats_incoming_stream_monitor_deviation_threshold,20,Integer,20,false,null,stats_incoming_stream_monitor_deviation_threshold,If deviation between audio and video is greater than the threshold in milliseconds

stats_sampling_frequency

1000

Long

true

Interval in milliseconds. Stream sampling will be taken with the specified frequency

stream_idle_bitrate_monitoring

false

Boolean

false

Enable monitoring of published streams based on bitrate

stream_idle_bitrate_monitoring_threshold_bps

10000

Long

false

Lowest bitrate possible for the active stream

stream_idle_bitrate_monitoring_window_sec

120

Integer

false

Mean stream bitrate calculation window in seconds

stream_record_policy

String

false

Available values: streamName, template.
By default, WCS server generates filename based on mediaSessionId and login.
If set to streamName'

stream_record_policy_template

String

false

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,1,String,1,false,null,streaming_custom_stream_stress_test_encoding_subscriber_groups,StreamingCustomStreamStressTest / Number of subscribers for transcoded stream

streaming_custom_stream_stress_test_max_proxy_subscribers

1

Integer

false

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,1000,Long,1000,false,null,streaming_custom_stream_stress_test_rate,StreamingCustomStreamStressTest / Period in milliseconds. Each period a new subscriber will be added,[]streaming_custom_stream_stress_test_stream_name,STRESS_TEST_STREAM,String,STRESS_TEST_STREAM,false,null,streaming_custom_stream_stress_test_stream_name,StreamingCustomStreamStressTest / Name of stream published on WCS server

streaming_custom_stream_stress_test_subscriber_ttl_sec

30

Long

false

StreamingCustomStreamStressTest / Lifetime of subscriber in seconds

streaming_distributor_dump_interval

10

Integer

true

Interval in minutes for getting distributor thread dumps

streaming_distributor_queue_max_waiting_time

5000

Integer

true

Maximum time that distributor thread will wait for frame arrival before executing next iteration

streaming_distributor_queue_size

300

Integer

true

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,0.95,Double,0.95,false,null,streaming_distributor_queue_size_dump_threshold,Distributor queue size threshold for getting dump,[]streaming_distributor_queue_size_log_threshold,10,Integer,10,true,null,streaming_distributor_queue_size_log_threshold,Threshold for logging distributor queue size,[]streaming_distributor_video_proxy_pool_enabled,false,Boolean,false,false,null,streaming_distributor_video_proxy_pool_enabled,Use thread pool for video distribution

streaming_load_test_duration_minutes

5

Long

false

StreamingLoadTest / Test duration in minutes

streaming_load_test_encoding_subscriber_groups

1

String

false

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,1,Integer,1,false,null,streaming_load_test_proxy_subscribers,StreamingLoadTest / Number of subscribers for non-transcoded stream (codecs

streaming_processor_queue_max_waiting_time

5000

Integer

true

Maximum time that processor thread will wait for frame arrival before executing next iteration

streaming_processor_queue_size

300

Integer

true

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,,String,,false,null,streaming_sessions_keep_alive_app_keys,Comma-separated list of appKeys of server-side applications. If set

streaming_sessions_keep_alive_interval

10000

Long

false

StreamKeepAliveEvent sending interval. See also streaming_sessions_keep_alive_app_keys= option

streaming_stress_test_duration_minutes

5

Long

false

StreamingStressTest / Test duration in minutes

streaming_stress_test_encoding_subscriber_groups

1

String

false

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,100,Integer,100,true,null,ws_connections_test_connections,Websocket connections to test,[]streaming_stress_test_rate,1000,Long,1000,false,null,streaming_stress_test_rate,StreamingStressTest / Period in milliseconds. Each period a new subscriber will be added,[]streaming_stress_test_subscriber_ttl_sec,30,Long,30,false,null,streaming_stress_test_subscriber_ttl_sec,StreamingStressTest / Lifetime of subscriber in seconds,[]streaming_tests,,String,,false,null,streaming_tests,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,false,Boolean,false,false,null,streaming_video_decoder_fast_start,If true

streaming_video_decoder_wait_for_distributors

true

Boolean

false

Stop decoding temporarily if one of the distributors fails to keep up with decoding

streaming_video_decoder_wait_for_distributors_max_queue_size

5

Integer

true

Stop decoding when one of distributors queue reaches specified size (See streaming_video_decoder_wait_for_distributors)

streaming_video_decoder_wait_for_distributors_timeout

33

Integer

true

Specifies how long decoding should wait before another distributors queue check (See streaming_video_decoder_wait_for_distributors)

streaming_video_decoder_warmup

true

Boolean

false

Warmup video decoder with P frame after I frame regardless of decoding point availability

streaming_video_decoder_warmup_frames

5

Integer

false

How many P frames should be used for warmup

strict_get_callee_policy

false

Boolean

false

Not in use

stun_freshness_period

1500

Integer

false

STUN freshness period in milliseconds

stun_freshness_timeout

15000

Integer

false

STUN freshness timeout in milliseconds

stun_server

stun1.l.google.com:19302

String

false

STUN server, which is used for WebRTC ICE, if enable_candidate_harvester=true,deprecatedstun_socket_buffer_size,100,Integer,100,false,null,stun_socket_buffer_size,Size of STUN socket buffer,[]stun_socket_queue_size,100,Integer,100,false,null,stun_socket_queue_size,Size of STUN socket queue,[]stun_socket_queue_timeout,1500,Integer,1500,false,null,stun_socket_queue_timeout,STUN socket queue timeout in milliseconds,[]stun_stack_default_thread_pool_size,0,Integer,0,false,null,stun_stack_default_thread_pool_size,STUN default thread pool size,[]stun_wait_candidate_timeout,1000,Integer,1000,false,null,stun_wait_candidate_timeout,STUN waiting candidate timeout for nominate in milliseconds,[]suppress_audio,false,Boolean,false,false,null,suppress_audio,If true

suppress_dynamic_logs

false

Boolean

false

If true, suppress dynamic logs update,[]suppress_dynamic_logs_to_server_log,false,Boolean,false,false,null,suppress_dynamic_logs_to_server_log,If true

tcp_relay_packetization2

true

Boolean

false

If true, enable TCP relay packetization for WSPlayer. Should be false when WSPLayer 1.0 is used,[]tcp_relay_packetization_time,20,Integer,20,false,null,tcp_relay_packetization_time,Experimental option

tcp_relay_rtcp_interval

2000

Integer

false

RTCP packets generation interval for TCP relay in milliseconds. RTCP is used to carry stream synchronization

thread_pool_default_core_threads

4

Integer

true

Default core threads count in thread pool (equal to CPUs count)

thread_pool_default_max_threads

8

Integer

true

Maximum core threads count in thread pool

thread_pool_default_queue_size

100

Integer

true

Default thread pool queue size

thread_pool_default_thread_timeout_sec

300

Integer

true

Default thread timeout, in seconds,[]throughput_test_receivers_qty,1,Integer,1,false,null,throughput_test_receivers_qty,Throughput test receivers quantity,[]throughput_test_sender_dst,localhost,String,localhost,false,null,throughput_test_sender_dst,Throughput test sender destination host,[]throughput_test_senders_qty,1,Integer,1,false,null,throughput_test_senders_qty,Throughput test senders quantity,[]timing_shift,null,String,null,false,null,timing_shift,Timer ambiguity in milliseconds

trace_socket_fd

false

Boolean

true

If true, trace usage of socket file descriptors for HLS, HTTP, RTSP, WebSockets and HTTP LB client,[]transcoder_agent_activity_timer_cool_off_period,1,Integer,1,false,null,transcoder_agent_activity_timer_cool_off_period,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,60000,Integer,60000,false,null,transcoder_agent_activity_timer_timeout,If there is no subscribers for an Transcoder agent stream within this timeout in milliseconds

transcoder_agent_rtcp_send_interval

2000

Long

false

Interval in ms for send rtcp from transcoder agent

transcoder_align_encoders

false

Boolean

false

Align video encoders of the same video input by key frames

transcoding_disabled

false

Boolean

false

Force transcoding disabling

turn.server_channel_receive_buffer_size

1048576

Integer

true

Receive buffer size for turn channels

turn.server_channel_send_buffer_size

1048576

Integer

true

Send buffer size for turn channels

turn_ip

null

String

true

TURN IP address

turn_life_time

600

Integer

true

TURN Allocation life time

turn_media_port_from

36001

Integer

true

Beginning of media ports range for turn

turn_media_port_to

37000

Integer

true

End of media ports range for turn

turn_media_ports_auditor_interval

5000

Integer

true

Audit interval for busy and free ports, in milliseconds,[]turn_media_ports_auditor_max_attempts,3,Integer,3,true,null,turn_media_ports_auditor_max_attempts,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,coM77EMrV7Cwhyan,String,coM77EMrV7Cwhyan,true,null,turn_password,TURN password,[]turn_port,3478,Integer,3478,true,null,turn_port,TURN server port,[]unsupported_messages,null,String,null,false,null,unsupported_messages,If a message has body noted in this list

use_alaw_ulaw_speex_switch

true

Boolean

false

If true, switch to the local codec according to content received from SIP side.
If false, use Speex16,deprecateduse_control_destination_from_incoming_rtcp,true,Boolean,true,false,null,use_control_destination_from_incoming_rtcp,If true

use_fdk_aac

true

Boolean

false

If true, use the fdk-aac fro encoding and decoding,use_ip_local_in_call_id,true,Boolean,true,false,null,use_ip_local_in_call_id,If true

use_java_hls_writer

true

Boolean

false

If true, use Java HLS implementation,[]use_mp4_h264_aac,true,Boolean,true,false,null,use_mp4_h264_aac,If true

use_new_aac_encoder

true

Boolean

false

If true, use the latest AAC encoder,deprecateduse_new_rtcp,true,Boolean,true,false,null,use_new_rtcp,If true

use_rtcp_synch

true

Boolean

false

If true, use RTCP synchronization for audio and video,deprecateduse_rtmp_java_client,true,Boolean,true,false,null,use_rtmp_java_client,If true

use_speex_java_impl

true

Boolean

true

If true, use Java implementation for Speex codec,[]use_tcp_for_long_sip_messages,false,Boolean,false,false,null,use_tcp_for_long_sip_messages,If true

use_trying_notification

false

Boolean

false

If true, then broadcast SIP response TRYING to client as a call status TRYING,[]user_agent,Flashphoner/1.0,String,Flashphoner/1.0,true,null,user_agent,User-Agent header value,[]video_decoder_max_threads,2,Integer,2,false,null,video_decoder_max_threads,How many threads will be used for decoding,[]video_decoder_second_thread_threshold,777000,Integer,777000,false,null,video_decoder_second_thread_threshold,Resolution threshold. Once it is reached

video_distributor_multi_test

false

Boolean

false

Enable video distributor multi test

video_enabled

true

Boolean

false

Not in use

video_encoder_h264_gop

60

Integer

false

GOP size for H.264 encoder

video_encoder_max_threads

2

Integer

false

How many threads will be used for encoding

video_encoder_second_thread_threshold

777000

Integer

false

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,900,Integer,900,false,null,video_encoder_vp8_gop,GOP size for VP8 encoder,[]video_encoding_quality,30,Integer,30,false,null,video_encoding_quality,See information on FFmpeg CRF,[]video_filter_enable_fps,false,Boolean,false,true,null,video_filter_enable_fps,Enable video filter,[]video_filter_enable_rotate,false,Boolean,false,true,null,video_filter_enable_rotate,Enable video rotate filter,[]video_filter_fps,30,Long,30,true,null,video_filter_fps,Video filter output fps,[]video_filter_fps_gap_coefficient,2.0,Double,2.0,true,null,video_filter_fps_gap_coefficient,Video filter gap coefficient (max gap C x FPS),[]video_filter_fps_gop_synchronization,0,Integer,0,false,null,video_filter_fps_gop_synchronization,Filter's gop value used to provide synchronization point for encoders

video_incoming_buffer_size

20

Integer

false

Waiting for RTCP sync packet on this interval in packets, for video,[]video_processor_multi_test,false,Boolean,false,false,null,video_processor_multi_test,Enable video processor multi test,[]video_reliable,partial,on
partial
off,partial,false,null,video_reliable,RTMFP

video_stream_mode_udp

false

Boolean

true

Not in use

video_streamer_generate_seq

false

Boolean

false

Should be set to true for transfer of video calls. Otherwise, there may be no video after transfer,[]video_transcoder_preserve_aspect_ratio,true,Boolean,true,true,null,video_transcoder_preserve_aspect_ratio,Try to preserve original aspect ratio of incoming video during transcoding,[]vod_activity_timer_cool_off_period,1,Integer,1,false,null,vod_activity_timer_cool_off_period,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,true,Boolean,false,false,null,vod_live_loop,If true

vod_mp4_container_isoparser_heap_datasource

true

Boolean

false

If true, use heap datasource,[]vod_mp4_container_new,false,Boolean,false,false,null,vod_mp4_container_new,Use new implementation of mp4 container for vod,[]vod_mp4_container_new_buffer_ms,0,Integer,0,false,null,vod_mp4_container_new_buffer_ms,New implementation of mp4 container will buffer specified time in milliseconds,[]vod_mp4_test_file,null,String,null,false,null,vod_mp4_test_file,Path to MP4 file. If start_test=true and streaming_tests=MP4AgentTest

vod_mp4_test_loop

true

Boolean

false

If true, loop streaming MP4 file. Not in use, replaced by vod_live_loop=,deprecatedvod_mp4_test_stream_name,null,String,null,false,null,vod_mp4_test_stream_name,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,2000,Long,2000,false,null,vod_rtcp_send_interval,RTCP Send interval for VOD,[]vod_sink_ready_checks,50,Integer,50,false,null,vod_sink_ready_checks,Waiting for first packet on audio streamer. If no packets within the specified number of checks

vod_sink_wait_synch_time

true

Boolean

false

If false, not wait sync time for playing incoming traffic after audio sink,[]vod_stream_timeout,14400000,Integer,30000,false,null,vod_stream_timeout,VoD stream with no subscribers will be terminated after this timeout in milliseconds,[]vow_wait_for_sync,false,Boolean,false,false,null,vow_wait_for_sync,If true

vp8_buffer_nack_list_threshold

200

Integer

false

JitterBuffer will be reset upon reaching this number of NACK packets

vp8_max_rtp_packet_size

1400

Integer

true

Maximum size of VP8 carrying packet

vp8_new_buffer

false

Boolean

false

Not in use

wcs_activity_timer_cool_off_period

1

Integer

false

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

60000

Integer

false

If there is no activity within this timeout in milliseconds, corresponding WCS agent session will be terminated,[]wcs_agent_force_video_orientation,true,Boolean,true,false,null,wcs_agent_force_video_orientation,Force negotiation of 3gpp video orientation extension for wcs agents

wcs_agent_port_from

34001

Integer

false

Beginning of range of ports for WCS agent

wcs_agent_port_to

35000

Integer

false

End of range of ports for WCS agent

wcs_agent_ports_auditor_interval

10000

Integer

false

Audit interval for WCS agent ports, in milliseconds,[]wcs_agent_ports_auditor_max_attempts,3,Integer,3,false,null,wcs_agent_ports_auditor_max_attempts,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,30000,Integer,30000,false,null,wcs_agent_session_alive_check_interval,Interval in milliseconds to check if WCS agent session is alive,[]wcs_agent_session_audit,true,Boolean,true,false,null,wcs_agent_session_audit,If true

wcs_agent_session_connect_timeout

10000

Integer

false

Connect timeout in milliseconds

wcs_agent_session_timeout

30000

Integer

false

WCS agent session timeout in milliseconds

wcs_agent_session_use_keep_alive_timeout

true

Boolean

true

If true, WCS agent session will use keep alive timeout,[]wcs_agent_ssl,false,Boolean,false,false,null,wcs_agent_ssl,If true

wcsoam_batch_timeout

500

Integer

true

WCS OAM receive timeout

wcsoam_buffer_size

20000

Integer

true

WCS OAM buffer size in kB

wcsoam_chunk_size

64

Integer

true

WCS OAM send chunk size in kB

wcsoam_hostname

null

String

true

WCS OAM server hostname

wcsoam_ip

null

String

true

WCS OAM server IP address

wcsoam_keepalive_period

3000

Integer

true

WCS OAM keep alive period

wcsoam_keepalive_timeout

8000

Integer

true

WCS OAM keep alive timeout

wcsoam_ping_enabled

true

Boolean

false

WCS OAM server ping enable

wcsoam_ping_interval

10000

Integer

true

WCS OAM server ping interval in ms

wcsoam_port

7777

Integer

true

WCS OAM server port

wcsoam_reconnect_interval

5000

Integer

true

WCS OAM reconnect interval in ms

wcsoam_sha_salt

123

String

true

WCS OAM server SHA salt

web_start_with_demo_user

false

Boolean

false

Enable demo user

web_token_life_time

3600000

Long

false

Web token life time, default value 1 hour,[]webrtc_aes_crypto_provider,BC,BC
JCE,BC,false,null,webrtc_aes_crypto_provider,Crypto provider for WebRTC,[]webrtc_agent_use_webrtc,true,Boolean,true,false,null,webrtc_agent_use_webrtc,If true

webrtc_cc2

true

Boolean

false

If true, the latest congestion control CC2 is used,[]webrtc_cc2_bitrate_overuse_event,false,Boolean,false,false,null,webrtc_cc2_bitrate_overuse_event,If true

webrtc_cc2_bitrate_overuse_event_interval

5000

Long

false

NBE event will be raised periodically with this interval in milliseconds

webrtc_cc2_bitrate_overuse_event_threshold

0.05

Double

false

NBE event will be raised when loss on stream being played reaches this value (5% by default)

webrtc_cc2_cc

false

Boolean

false

If true, react upon WebRTC playback endpoint (e.g. Chrome) requests, e.g. request the publisher to decrease bitrate,[]webrtc_cc2_cc_interval,500,Long,500,false,null,webrtc_cc2_cc_interval,Congestion control interval

webrtc_cc2_cc_k_noise

0.1

Double

false

Congestion control noise value, not in use,deprecatedwebrtc_cc2_cc_retransmit_rate_threshold,0.15,Double,0.15,false,null,webrtc_cc2_cc_retransmit_rate_threshold,Fraction of send bitrate that retransmit bitrate can raise to. By default

webrtc_cc2_cc_track_joined_retransmit_bitrate

true

Boolean

false

If true, enable tracking of retransmit bitrate across all media groups,[]webrtc_cc2_enable_burst_grouping,false,Boolean,false,false,null,webrtc_cc2_enable_burst_grouping,Internal bitrate estimation configuration

webrtc_cc2_local_congestion_event_interval

2000

Long

false

Not in use, legacy code,deprecatedwebrtc_cc2_local_k_threshold,0.1,Double,0.1,false,null,webrtc_cc2_local_k_threshold,Not in use

webrtc_cc2_min_remb_bitrate_bps

100000

Long

false

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

1000

Long

false

Window size for receiver state, in milliseconds. Default: 1000 - keep and account reports received in last second,[]webrtc_cc2_twcc,false,Boolean,false,false,null,webrtc_cc2_twcc,If true

webrtc_cc_bitrate_window

1000

Integer

false

Time window in milliseconds. Bitrate estimator works on this time frame

webrtc_cc_initial_avg_noise

0.0

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_initial_e_0_0,100.0,Double,100.0,false,null,webrtc_cc_initial_e_0_0,Internal bitrate estimation configuration

webrtc_cc_initial_e_0_1

0.0

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_initial_e_1_0,0.0,Double,0.0,false,null,webrtc_cc_initial_e_1_0,Internal bitrate estimation configuration

webrtc_cc_initial_e_1_1

0.1

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_initial_offset,0.0,Double,0.0,false,null,webrtc_cc_initial_offset,Internal bitrate estimation configuration

webrtc_cc_initial_process_noise_0

1.0E-13

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_initial_process_noise_1,0.001,Double,0.001,false,null,webrtc_cc_initial_process_noise_1,Internal bitrate estimation configuration

webrtc_cc_initial_slope

0.015625

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_initial_threshold,15.0,Double,15.0,false,null,webrtc_cc_initial_threshold,Internal bitrate estimation configuration

webrtc_cc_initial_var_noise

50.0

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_k_down,1.8E-4,Double,1.8E-4,false,null,webrtc_cc_k_down,Internal bitrate estimation configuration

webrtc_cc_k_up

0.01

Double

false

Internal bitrate estimation configuration, must not be exposed to public,[]webrtc_cc_max_bitrate,10000000,Long,10000000,false,null,webrtc_cc_max_bitrate,Maximum global bitrate for publishing WebRTC streams,[]webrtc_cc_min_bitrate,30000,Long,30000,false,null,webrtc_cc_min_bitrate,Minimum global bitrate for publishing WebRTC streams,[]webrtc_cc_overusing_threshold,10.0,Double,10.0,false,null,webrtc_cc_overusing_threshold,Internal bitrate estimation configuration

webrtc_cc_use_sync_ts

true

Boolean

false

If true, timestamp is used as synchronization source,[]webrtc_sdes_extensions,false,Boolean,false,false,null,webrtc_sdes_extensions,Enable sdes rtp header extensions,[]webrtc_sdp_bandwidth_bps,0,Long,0,false,null,webrtc_sdp_bandwidth_bps,b=AS/b=TIAS in publish sdp,[]webrtc_sdp_h264_exclude_profiles,,String,,false,null,webrtc_sdp_h264_exclude_profiles,List of H264 profiles which should be excluded in response on SDP negotiation.
42 - Baseline

webrtc_sdp_max_bitrate_bps

0

Long

false

x-google-max-bitrate in publish sdp

webrtc_sdp_min_bitrate_bps

0

Long

false

x-google-min-bitrate in publish sdp

work_around

false

Boolean

false

Not in use

ws.address

0.0.0.0

InetAddress[]

true

parse and inject custom HTTP headers to REST requests

ws.port

8080

Integer

true

WebSocket connection port

ws_client_id_unique_part

true

Boolean

false

Add unique part to ws client id

ws_connections_test_run_for

1800

Integer

true

Websocket connections test duration in seconds

ws_connections_test_uri

ws://192.168.88.100:8080

String

true

Websocket connections test URI

ws_read_socket_timeout

true

Boolean

true

Enable WebSocket read timeout

ws_read_socket_timeout_sec

120

Integer

true

WebSocket read timeout value (if enabled)

wss.address

0.0.0.0

InetAddress[]

true

in seconds