Main server settings
Outdated or invalid setting are highlighted in grey. They were used in previous versions. These settings will be probably deleted in further WCS updates.
Option | Default | Type | Need restart | Description |
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. |
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 | 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. |
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. |
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 |
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 | 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 |
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: |
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. |
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 |
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). |
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. |
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 |
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 | 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. |
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. |
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. |
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 |
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). |
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. |
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. |
min_drop_rate | null | String | false | Queue size will be decreased if loss reduces to this value. |
min_queue_size | null | String | false | Queue size will not be decreased lower that this minimum value. |
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 | 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. |
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. |
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: |
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). |
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. |
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. |
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. |
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. |
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,...}). |
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. |
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. |
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. |
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). |
rtp_send_buffer_size | 65536 | Integer | false | Buffer size for outgoing RTP and SRTP (WebRTC). |
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. |
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 |
sip_ports_auditor_max_attempts | 3 | Integer | false | Number of audits to make sure freed port is not bound. |
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. |
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. |
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. | |
stream_record_policy_template | String | false | If set, name of recorded file will be built using the specified template. | |
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 |
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 |
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. |
use_alaw_ulaw_speex_switch | true | Boolean | false | If true, switch to the local codec according to content received from SIP side. |
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. |
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 |
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. |
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. |
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 |
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. |
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 |