Settings
This page describes configuration settings for StarTrinity SIP Tester and StarTrinity Softswitch.
The settings are stored in XML file in program folder near .exe file.
AffinityMaskForAudioVerificationThreads - Binary mask specifying which CPU cores are used by the offline (PESQ) audio verification threads. Example: "1000" - "use core #3"
Requires restart: True
AffinityMaskForMediaThreads - Binary mask specifying which CPU cores are used by the RTP media threads. Example: "1110" - "use core #1, #2, #3, don't use core #0"
Requires restart: True
AffinityMaskForProcess - Binary mask specifying which CPU cores are used by the software. Example: "1110" - "use core #1, #2, #3, don't use core #0"
Requires restart: True
AffinityMaskForSipThread - Binary mask specifying which CPU cores are used by the SIP thread. Example: "0010" - "use core #1"
Requires restart: True
AllowSRTP - Optional SRTP media transport mode. Enables declaration of 'crypto' attributes in SDP. See also other mode: 'RequireSRTP'
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
AudioVerificationThreadsCount - Number of threads used for offline audio verification (PESQ MOS mode)
Default: 2
Requires restart: True
AutomaticReRegisterOnFailure - Enables automatic re-sending REGISTER if UAC registration fails, in random time 0..10 seconds
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
AutomaticReRegisterOnFailureMaxAttemptsCount - Limits number of automatic UAC re-REGISTER's, default empty setting value means no limit. Note: please decrease setting "TimerDMs" to decrease number of REGISTER requests with same SIP Call-ID
Default:
Values:
Requires restart: False
AutomaticSoftwareRestartPeriodH - Optional setting to enable automatic software restarts every N hours. Is used to avoid memory leaks.
Default:
Requires restart: False
AutoUpdate - Enables automatic updating of software from web server
Values: 1 to enable, 0 to disable
Requires restart: True
BackupSettingsAndScripts - Enables automatic backup
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
BackupSettingsAndScriptsStorageTimeInDays - Max time to store files in backup directory, in days. Old files are deleted automatically
Default: 30
Requires restart: False
CdrCsvFilesSplitMode - CDR CSV files split mode: "day" or "month"
Default: day
Values: day, month
Requires restart: False
CdrDeleteIntervalInDays - Time period for deleting old CDR CSV files, max. age of stored CDR files
Default:
Requires restart: False
CdrDisplaySource - Selects which calls are displayed in CDR: CSV files or memory. 'memory' source is faster than 'CSV', but it gets erased when you restart the software
Default: memory
Values: memory, CSV
Requires restart: False
CdrOdbcConnectionString - Specifies connection to database where to save CDR records
Values: Driver={SQL Server};Server=localhost\SQLEXPRESS;Database=testCDR;Uid=sa;Pwd=pass;
Requires restart: False
CdrPath - Path to generated CSV CDR files
Default: [program_folder]\CDR
Requires restart: True
CdrReadTimeoutS - Max time for CDR requests. Is used in StarTrinity softswitch to avoid overloads
Default: 3600
Requires restart: False
CdrStatisticsReadTimeoutS - Max time for asynchronous CDR statistics requests. Is used in StarTrinity softswitch to avoid overloads
Default: 14400
Requires restart: False
CdrTableName - CDR table name in database
Requires restart: True
CheckFromHeaderInInviteAuthorization - Checks 'From' SIP header when authorizing incoming INVITE request, rejects the request if user ID in 'From' header (caller ID) does not match authenticated user ID
Default: 1
Requires restart: False
CheckToHeaderInRegisterAuthorization - Checks 'To' SIP header when authorizing incoming REGISTER request, rejects the request if user ID in 'To' header does not match authenticated user ID
Default: 1
Requires restart: False
ClearFileCacheOnCreateSingleCall - Automatically clears cached callxml, CSV, audio files when 'create single call' button is manually clicked
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
ConnectAllCallsToSoundCardMicrophone - Enables automatic audio connections from sound card microphone to all current SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
ConnectAllCallsToSoundCardSpeaker - Enables automatic audio connections from all current SIP calls to sound card speakers
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
CsvDelimiter - Delimiter between fields in CSV file, used to read or write CSV files. Default value is taken from system settings
Values: ; or ,
Requires restart: True
DeclareAllCodecsInSdpAnswer - Enables declaration of all locally available codecs in SDP answer
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DeclareG711AInSDP - Enables declaration of audio codec G.711A (payload type = 8) in SDP
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
DeclareG711UInSDP - Enables declaration of audio codec G.711U (payload type = 0) in SDP
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
DeclareG723InSDP - Enables declaration of audio codec G.723 (payload type = 4) in SDP
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
DeclareG729InSDP - Enables declaration of audio codec G.729 (payload type = 18) in SDP
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
DebugMedia - Enables writing of RX audio streams of each call
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DebugMediaAudioVerification - Enables saving audio signal into WAV files for "verifyaudio" CallXML elements in PESQ mode
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DebugMediaFileNamePattern - Path to recorded RX and TX audio streams. May end with '\', or may not end with '\'
Default: [year]_[month]_[day]\[hour]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId]
Requires restart: False
DebugMediaPath - Path to recorded RX, TX, RX+TX (mixed) WAV audio files. Ending '\' character is ignored
Default: C:\debug_media
Requires restart: False
DebugMediaSpeechRecognition - Enables saving audio signal into WAV files for "inputspeech" CallXML elements
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DebugMediaTX - Enables writing of TX audio streams of each call
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DeclareRFC2833InSDP - Enables declaration of RFC2833 payload type in SDP offer and answer
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
DefaultLocalIpAddress - Selects network interface which is used if GetBestInterface() WinAPI function returns 127.0.0.1, to select which local network adapter is used for media stream
Requires restart: True
DefaultSignalDetectorThresholdDb - Default signal detector threshold level in decibels. The threshold is used for "AudioSignalDelay" CDR field
Default: -24
Requires restart: False
DesiredAudioCodec - Audio codec preference when selecting one of multiple codecs in SDP offer. Warning: this setting caused issues with some SIP servers which did not process SDP answer correctly
Values: 0: PCMU (G.711 mu-law), 8: PCMA (G.711 a-law), 4: G.723, 18: G.729
Requires restart: False
DestroyCallOnSessionTimerExpiry - Enables termination of dead SIP calls if session timer is expired. Warning: setting this setting to 1 together with high CPS and high value of session-expires timer header can cause memory leak
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DetectAllSipCalls - Enables detection and measurement of SIP calls which are not generated or received by SIP Tester, turns on "passive" operation mode
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DetectedRtpStreamMaxIdleTimeMs - Timeout to remove detected RTP streams from memory when no more RTP packets are detected
Default: 10000
Requires restart: False
DetectRingbackTone - Enables detection of ringback tone in RTP early media signal for FAS (false answer) detection
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
DisableAnonymousReports - Disables reporting of crash reports and usage statistics to developer's server. The data is encrypted
Default: 0
Values: 1 to disable, 0 to enable
Requires restart: True
DisablePacketAnalysisOnIpAddresses - Semicolon-separated list of local NIC's IP addresses where to disable packet analyser. Is used to turn off RTP packet processing on selected interfaces. Disabling RTP packet analysis for 95% of calls, and processing RTP for only 5% of calls helps to reduce CPU load. Local NIC which is used for RTP in the SIP call is set in CallXML script
Requires restart: True
DisableWDT - Disables watchdog timer. When you set this setting to 0, it enables watchdog timer, which restarts the softswitch in case of deadlock
Default: 1
Values: 1 to disable, 0 to enable
Requires restart: True
DisableRtpPacketAnalysisOnIpAddresses - Semicolon-separated list of local NIC's IP addresses where to disable RTP packet analyser. Is used to turn off RTP packet processing on selected interfaces. Disabling RTP packet analysis for 95% of calls, and processing RTP for only 5% of calls helps to reduce CPU load. Local NIC which is used for RTP in the SIP call is set in CallXML script
Requires restart: True
DisableUacUnregisters - Disables sending un-REGISTERs in UAC registrations
Default: 0
Values: 1 to disable, 0 to enable
Requires restart: False
DnsMode - DNS record type used for UAC registrations (extensions), INVITE and REGISTER requests
Default: A
Values: A, SRV
Requires restart: True
DontPrintPortInRequestUri - Disables printing of port number in Request URI of INVITE SIP message
Default: 0
Values: 1 to disable, 0 to enable
Requires restart: False
DtmfDurationRfc2833Ms - Duration of transmitted RFC2833 (inband, RTP) DTMF signals
Default: 100
Values: Some SIP servers need at least 200ms DTMF signals
Requires restart: False
DtmfSendingIntervalMs - Interval between consequent RFC2833 (inband, RTP) DTMF signals
Default: 200
Requires restart: False
EnableAutomaticHtmlReports - Setting to 0 disables automatic saving HTML reports into Logs folder. By defaut SIP Tester saves report every 10 minutes
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableCallMeasurements - Setting to 0 disables measurements of call quality indicators for CDR, and turns off CDR, to reduce CPU load
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: True
EnableCloseProgramConfirmation - Enables confirmation message box when SIP Tester desktop GUI is closed
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableLightweightMediaProcessing - Enables "lightweight media processing" operation mode. In this mode only RTP playback from audio file is possible
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableMultiTenancy - Enables multi-tenancy for your softswitch (usage of single server by multiple independent VoIP business owners, shared hosting)
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnablePacketAnalyser - Enables winpcap-based RTP and SIP packet analyser/sniffer which is used to measure VoIP quality
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: True
EnablePacketAnalysisOnlyOnFollowingIpAddresses - Semicolon-separated list of local NIC's IP addresses where to run packet analyser. If empty, all available NICs are used for packet capturing
Requires restart: True
EnablePartialFrameOffsetInRealtimeAudioVerifier - Enables search partial-frame matches in realtime audio verifier. Partial-frame delay (not multiple of 10ms) of IVR audio response can be caused by a PSTN network between SIP Tester and IVR server
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
EnablePerformanceDebugger - Enables "performance debugger" operation mode, it is used by developers to check performance of specific modules, like RTP transmitter alone
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnablePLC - Enables packet loss concealment (PLC) for G.711 codec
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
EnableR2S - Send R2S packets when WG67 PTT is off. Applies to TX RTP only
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
EnableRtcpXr - Enables sending of RTCP extended reports (RTCP XR)
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableRtpStatisticsCalculationInPacketAnalyser - Enables calculation of RTP jitter, packet loss, G.107 MOS in packet analyser module.
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableSignalDetector - Enables audio signal measurement for "-24dB Audio Signal Delay" CDR field
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableSignalDetectorMaxLevelMeasurementsForCdrFields - Enables measurements for CDR fields: EarlyMediaPeakSignalLevelDb, ActiveMediaPeakSignalLevelDb
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
EnableSignalLevelProcessingInPacketAnalyser - Is enabled to measure "Caller mean audio signal level" CDR field in "packet analyser" module. The measurements can be disabled to save CPU resources in VoIP recorder mode
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
EnableSipTcpTransport - Turns on SIP/TCP transport, opens local TCP SIP server socket
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: True
EnableSoundCardModule - Turns on sound card module
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableSsrcRtpFilter - Enables SSRC filter in RTP receiver module ("normal media processing" operation mode). The SSRC filter drops RTP packets with altered SSRC field
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
EnableTls - Enables TLS (SIPS) transport for SIP messages
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableVoipMarketplaceFeatures - Enables VoIP marketplace features in the softswitch: bilateral agreements, client-defined routing in self-care web portal
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableVoipRetailFeatures - Enables VoIP retail features in the softswitch
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableWinpcapRtpProxy - Enables "WinPCAP RTP proxy" operation mode for StarTrinity softswitch
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
EnableWinpcapRtpSender - Enables "WinPCAP RTP sender" operation mode for all SIP calls. In this mode the RTP packets are sent using fast WinPCAP raw sockets. The WinPCAP RTP sender mode is also turned on by CallXML attribute "rtpDscp"
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
ForceContiguousMediaFlow - Enables sending of silent audio frames to call term when no audio source is present
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
ForcedAudioCodec - Forced audio codec preference
Values: 0: PCMU (G.711 mu-law), 8: PCMA (G.711 a-law), 4: G.723, 18: G.729
Requires restart: False
ForcedLocalAddress - Specifies IP address of network adapter to be used for RTP/SRTP media streams and for T.38 fax data streams
Requires restart: True
ForcedMappedAddress - Optional IP address to declare in SIP Contact header of INVITE and REGISTER, in SDP. It is used for making SIP calls through NAT. Overrides "StunServerAddress" setting
Requires restart: True
GoipRcsPort - Local TCP port number for the softswitch when running in SIM management mode with GoIP GSM gateways
Default: 7150
Requires restart: True
GuiUpdateIntervalMs - GUI update interval in milliseconds. Set 0 to disable automatic GUI updates on timer, to save CPU resources
Default: 2500
Requires restart: False
HttpProxyPassword - HTTP proxy password, is set in GUI and encrypted in settings file, used for updating software from developer's web server
Requires restart: False
HttpProxyUrl - HTTP proxy, used for updating software from developer's web server
Values: http://192.168.1.1:8080
Requires restart: False
HttpProxyUserName - HTTP proxy user name, used for updating software from developer's web server
Requires restart: False
HttpServerName - Local HTTP and HTTPS (web interface) server name
Requires restart: False
IdleMediaTimeToEndSession - Timeout for termination INVITE session if no media packets are received from peer, in seconds. Raises CallXML event 'idleMedia'
Default: 200
Requires restart: False
IgnoreUnknownRequiredSipExtensions - Ignores SIP extensions that are required in incoming INVITE, in SIP header "Required".
Set to 1 if the SIP Tester does not accept incoming call due to some custom required SIP extension sent from your side (and rejects with "420").
Is used for test purposees only
Default: 0
Values: 1 - to ignore
Requires restart: True
JitterBufferInit - The initial "prefetch delay" to be applied to the jitter buffer.
Prefetch buffering is keeping RTP frames in the dynamic jitter buffer after gets empty, until its size reaches this "prefetch delay", in milliseconds
Default: 10
Requires restart: False
JitterBufferMax - Maximum number of frames that can be kept in the dynamic jitter buffer, i.e. the maximum delay that may be introduced by the jitter buffer, in milliseconds
Default: 80
Requires restart: False
JitterBufferMinPre - Minimum delay that must be applied to incoming packets, in milliseconds
Default: 10
Requires restart: False
LocalHttpPort - HTTP port used for web interface
Default: 19019
Values: 0 - disable HTTP web interface and Visual CallXML editor
Requires restart: True
LocalHttpsPort - HTTPS port used for web interface
Default: 19020
Values: 0 - disable HTTPS web interface
Requires restart: True
LocalHttpsCertificateIssuerName - Embedded web server HTTPS certificate issuer name
Values:
Requires restart: True
LocalHttpsCertificateSubjectName - Embedded web server HTTPS certificate common name, in "LocalMachine" or "LocalUser" certificate store
Values: sip.yourdomain.com, localhost
Requires restart: True
LocalSIPAddress - Specifies local network adapter IP address to send and listen SIP messages
Default:
Values: empty - use all local IP addresses
Requires restart: True
LocalSIPAddresses - Specifies local network adapter IP addresses to send and listen SIP messages, semicolon-separated list
Default:
Values: empty - use all local IP addresses
Requires restart: True
LocalSIPPort - Specifies UDP and TCP port number to send and listen SIP messages
Default: 5060
Values: 0 - use random non-busy port
Requires restart: True
LocalSIPPortRange - Specifies local SIP ports range, if using multiple local SIP port numbers
Default: 1
Values: Example: set LocalSIPPort=5060, LocalSIPPortRange=10 to use SIP ports 5060..5069
Requires restart: True
LocalTlsPort - Specifies TLS TCP port number to send and listen encrypted SIP/TCP messages
Default: 5061
Requires restart: True
LogsDeleteIntervalInDays - Max. age to automatically delete old logs and HTML reports
Default: 7
Requires restart: False
LogLevel - Filter level for log file writer
Default: 0
Values: 0: Error, 1: Warning, 2: Info, 3: Debug, 4: Trace
Requires restart: False
LogPath - Path to logs. If not set, logs are stored in directory near exe file
Default: [program_folder]\Logs
Requires restart: True
MailSenderFrom - SMTP parameter for sending email
Values: myusername@gmail.com
Requires restart: False
MailSenderPassword - SMTP server password for sending email. Is set in GUI and encrypted when stored in settings file
Requires restart: False
MailSenderPort - SMTP server port for sending email
Default: 25
Requires restart: False
MailSenderServer - SMTP server domain name for sending email
Values: smtp.gmail.com
Requires restart: False
MailSenderUserName - SMTP parameter for sending email
Values: myusername@gmail.com
Requires restart: False
MailSenderUseSsl - SMTP server transport level encription
Values: 0 or 1 or [empty - try to send email both with and without SSL]
Requires restart: False
MaxCallLifeTimeInHours - Timeout to forcefully abort deadlocked/zombie/hang SIP calls
Default: 48
Requires restart: False
MaxCaptureBufferDelayMs - Sound card buffer size to compensate delays between sound capture (microphone) thread and RTP thread. Please increase the buffer size to avoid glitches in sound
Default: 100
Requires restart: False
MaxCdrReaderThreadsCount - Maximal number of concurrent CDR requests. Is used in StarTrinity softswitch to avoid overloads
Default: 4
Requires restart: False
MaxDetectedCallDurationWithoutSipMessagesInMinutes - Timeout to forcefully abort deadlocked/zombie/hang captured SIP calls in "passive" mode
Default: 60
Requires restart: False
MaxMemoryCdrCallsCount - Max number of calls to store in CDR memory
Default: 2000
Requires restart: False
MaxPlaybackBufferDelayMs - Sound card buffer size to compensate delays between sound playback (speakers) thread and RTP thread. Please increase the buffer size to avoid glitches in sound
Default: 100
Requires restart: False
MaxRegistrationsPerSecond - Limit of new outgoing (UAC) REGISTER requests per second. Is set to avoid overloading of registration server or PBX
Default:
Requires restart: False
MediaClockPeriodMs - Period of media procesor clock procedure. If RTP packet time is 20ms, the setting can be set to 20 in order to minimize jitter
Default: 10
Requires restart: True
MediaClockUseSpinWait - Enables spinning technique in RTP media clock to achieve lowest transmitted RTP jitter (below 1ms). In this way the software takes 100% of CPU core for every media thread, so you need to set "MediaThreadsCount" to "1" or "2". If not set to 1, Sleep() WinAPI function is used in the media clock, it generates jitter of 5-15ms
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
MediaReceiverThreadsCount - Number of concurrent threads which receive RTP/UDP data and put it into jitter buffer
Default: 8
Requires restart: True
MediaThreadsCount - Number of concurrent threads which process (encode and decode, mix) and send RTP audio data
Default: 16
Requires restart: True
MediaTransportPoolInitialSocketsCountPerInterface - Number of RTP sockets (local RTP ports) to open initially on local network interface, during setup
Default: 4
Requires restart: True
MediaTransportPoolMinIdleTimeMs - Minimal time to keep RTP socket in idle (unused) state before reusing by next call
Default: 10000
Requires restart: False
MediaTransportPoolMinPort - Base port number for RTP sockets
Default: 16000
Requires restart: True
MediaTransportPoolMaxSocketsCountPerInterface - Maximal number of RTP sockets (local RTP ports) on a single local network interface
Default: 8000
Requires restart: True
MinimizeMiscThreadPoolPriority - Experimental setting, set to 1 to reduce number of worker and IOCP threads and reduce their priority, to avoid overloading of RTP and SIP threads. If set to 1, IOCP threads are forced to run at CPU core#1 (affinity mask 0b1110)
Default: 0
Values: 0, 1
Requires restart: True
NatList - List of mapped IP addresses, used for passive monitoring and VoIP recording, to find RTP streams by information from SDP. Semicolon-separated list of mappings from public IP address (declared in SDP) to internal IP address (actually captured RTP streams)
Values: LocalIP1:PublicIP1;LocalIP2:PublicIP2;...
Requires restart: False
NumberListIdsToLoadAtStartup - List of number list IDs to load into RAM at startup. By default the number lsits are loaded into RAM at first execution of number lookup procedure
Values: list1.txt,list2.txt,...
Requires restart: False
OverwriteCsvCdrFilesAfterRestart - Enables overwriting of CSV CDR files after restart of the software
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
PassUacRegistrationAuthInfoToIncomingCalls - Enables using of UAC registration credentials to received incoming calls for BYE requests
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
PerformanceIndicatorsHistoryMaxDays - Number of days to keep historical chart data for reports
Default: 10
Values:
Requires restart: True
PesqTruncateRemainderData - Removes remainder in recorded (degraded) audio data or in reference audio data, before passing the audio data to PESQ measurement procedure
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
Realm - Realm, is used for SIP authentication
Default: *
Requires restart: False
RecordedWavFileNamePattern - Specifies path to recorded wav file with mixed RX+TX RTP audio, see 'recordcall' CallXML element
Default: [year]_[month]_[day]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId]
Requires restart: False
RecordedWavFilesAudioCodec - Audio codec for recorded WAV files
Default: 8
Values: 0: PCMU (G.711 mu-law), 8: PCMA (G.711 a-law), 4: G.723, 18: G.729
Requires restart: False
RecordedWavFilesDurationLimitS - Max. length of recorded files, in seconds. Limits duration of all recorded WAV files, is used to save disk space
Requires restart: False
RecorderQueueMaxCount - Max number of audio frames being stored in RAM queue before writing them to recorded WAV file.
The RAM queue is used to move disk write operations away from media processing thread, to avoid high jitter
Default: 4096
Requires restart: True
RecordingsAutoDeleteMaxAgeInDays - Max. age to automatically delete old recorded WAV files
Default: 0
Requires restart: False
RecordSilenceWithoutRtp - Enables writing of silence into recorded WAV files when no RTP packets are received yet
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
RemoteRtcpMonitorAddress - Specifies an optional third-party destination (monitor) host name where to send RTCP and RTCP-XR packets
Requires restart: False
RemoteRtcpMonitorPort - Specifies an optional third-party destination (monitor) port where to send RTCP and RTCP-XR packets
Requires restart: False
Require100rel - Enables PRACK behaviour and "Require: 100rel" header for UAC and UAS SIP calls
Default: 0
Values: 1 or 0
Requires restart: False
RequireAuthorization - Enables authentication of incoming INVITE and REGISTER requests.
List of valid users and passwords is defined in 'Extensions/UAS registrations'.
INVITEs from trunks/SIP registrars (UAC registrations) are authenticated by source IP address, if setting "RequireAuthorizationForRequestsFromRegistrar" is "0"
Default: 0
Values: 1 or 0
Requires restart: False
RequireAuthorizationForRequestsFromRegistrar - Enables authentication of INVITE messages received from trunks/SIP registrars (UAC registrations)
Default: 0
Values: 1 or 0
Requires restart: False
RequireSRTP - Mandatory SRTP media transport mode. Requires "RTP/SAVP" transport declared in SDP. See also other mode: 'AllowSRTP'
Default: 0
Values: 1 or 0
Requires restart: False
ResetCallXmlSessionIdWhenStartedCallGenerator - Resets CallXML variable 'Id' when call generator is started
Default: 0
Values: 1 or 0
Requires restart: False
RestartSoftwareWhenConsumedRamIsAboveMB - If set, the software automatically restarts itself when consumed RAM memory reaches this threshiold (in megabytes). Is used as a workaround to fix memory leaks in the software
Default:
Values: example: 4000, in megabytes
Requires restart: False
RFC2833RXPayloadType - Expected RFC2833 DTMF event payload type
Default: 101
Requires restart: False
RtcpSendIntervalMs - Interval to send RTCP packets
Default: 1000
Requires restart: False
RtcpXrSendIntervalMs - Interval to send RTCP-XR packets
Default: 1000
Requires restart: False
RtpClockFrequencyHz - RTP clock frequency in Hz, is used to simulate RTP clock skew - transmit RTP packets with innacurate time intervals
Default: 8000
Requires restart: True
RtpRxBufferSize - RTP receiver windows socket buffer size (SO_RCVBUF)
Default: 0
Values: 0 to use default Windows setting
Requires restart: True
RtpTxBufferSize - RTP sender windows socket buffer size (SO_SNDBUF)
Default: 1048576
Values: 0 to use default Windows setting
Requires restart: True
RtpTxPacketTime - Delay between transmitted RTP packets, RTP packet duration in milliseconds
Default: 20
Values: 10, 20, 30, 40, 50
Requires restart: False
RtpTxTTL - IP TTL field for transmitted RTP packets. Works only in "WinPCAP RTP sender" mode
Default: 128
Requires restart: False
SaveRtpPacketsToMixedWav - For passive mode VoIP recording only: enables saving of captured RTP packets to disk as WAV file with mixed streams between A and B. Packets are saved into separate .wav files for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SaveRtpPacketsToMixedWav_PathPattern - Specifies path to written .wav file, optionally it may contain list of filters with multiple patterns for each filter expression.
Example: FILTER{CallerID startswith 1 or CalledID startswith 1}:c:\recordings_1\[sipCallId].wav;FILTER{CallerID startswith 2}:C:\recordings_2\[sipCallId].wav;c:\recordings\[sipCallId].wav
List of available CDR fields for the filter expression is displayed in GUI (CDR - fields..)
Default: recordings\[year]_[month]_[day]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId].wav
Values: May contain relative or absolute path
Requires restart: False
SaveSipAndRtpPacketsToDisk - Enables saving of captured SIP and RTP packets to disk for further debugging. Packets are splitted into separate .pcap files for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SaveSipAndRtpPacketsToDisk_PathPattern - Specifies path to written .pcap file, optionally it may contain list of filters with multiple patterns for each filter expression.
Example: FILTER{CallerID startswith 1 or CalledID startswith 1}:c:\pcap_1\[sipCallId]_SIP+RTP.pcap;FILTER{CallerID startswith 2}:C:\pcap_2\[sipCallId]_SIP+RTP.pcap;c:\pcap\[sipCallId]_SIP+RTP.pcap
List of available CDR fields for the filter expression is displayed in GUI (CDR - fields..)
Default: pcap\[year]_[month]_[day]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId]_sip_rtp.pcap
Values: May contain relative or absolute path
Requires restart: False
SaveSipPacketsToDisk - Enables saving of captured SIP packets to disk as PCAP file for further debugging. Packets are splitted into separate .pcap files for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SaveSipPacketsToDisk_PathPattern - Specifies path to written .pcap file, optionally it may contain list of filters with multiple patterns for each filter expression.
Example: FILTER{CallerID startswith 1 or CalledID startswith 1}:c:\pcap_1\[sipCallId]_SIP.pcap;FILTER{CallerID startswith 2}:C:\pcap_2\[sipCallId]_SIP.pcap;c:\pcap\[sipCallId]_SIP.pcap
List of available CDR fields for the filter expression is displayed in GUI (CDR - fields..)
Default: pcap\[year]_[month]_[day]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId]_sip.pcap
Values: May contain relative or absolute path
Requires restart: False
SaveSipPacketsToDiskAsTxt - Enables saving of captured SIP packets to disk as TXT file for further debugging. Packets are splitted into separate .txt files for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SaveSipPacketsToDiskAsTxt_PathPattern - Specifies path to written .txt file, optionally it may contain list of filters with multiple patterns for each filter expression.
Example: FILTER{CallerID startswith 1 or CalledID startswith 1}:c:\siptrace_1\[sipCallId]_SIP.txt;FILTER{CallerID startswith 2}:C:\siptrace_2\[sipCallId]_SIP+RTP.txt;c:\siptrace\[sipCallId]_SIP.txt
List of available CDR fields for the filter expression is displayed in GUI (CDR - fields..)
Default: sip_trace\[year]_[month]_[day]\[hour]_[minute]_[second]_[millisecond]_[callerId]_[calledId]_[sipCallId]_sip.txt
Values: May contain relative or absolute path
Requires restart: False
SaveRtpPacketsToMemory - Enables saving of captured RTP packets to memory for further processing via CDR history for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SaveSipPacketsToMemory - Enables saving of captured SIP packets to memory for further processing via CDR history for individual SIP calls
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SessionRefresher - RFC 4028 session refresher
Default: uac
Values: uac or uas
Requires restart: True
SessionExpiresInSeconds - Session-Expires header field value. The software sends keep-alive re-invites to refresh the SIP call according to RFC 4028.
Interval between the re-invite packets is SessionExpiresInSeconds * 0.5.
From RFC: The absolute minimum for the Session-Expires header field is 90 seconds
Default: 72000
Requires restart: False
SimulatedAckPacketLossProbability - Probability of not sending ACK in INVITE - 200 OK - ACK sequence, from 0 to 1. Is used to simulate abnormal UAC behaviour
Default: 0
Requires restart: True
SipUdpSendBufferSize - SIP/UDP windows socket sender buffer size. Is passed into setsockopt() winsockwets API as SO_SNDBUF
Default: 0 to use default Windows settings
Requires restart: True
SipUdpRecvBufferSize - SIP/UDP windows socket receiver buffer size. Is passed into setsockopt() winsockwets API as SO_RCVBUF
Default: 0 to use default Windows settings
Requires restart: True
SipThreadMaxQueueCountToRejectNewCall - Configuration of SIP thread anti-throttling mechanism, the SIP thread does not accept new calls if queue count gets over this threshold
Default: 30000
Requires restart: True
SipThreadMaxQueueDelayMsToRejectNewCall - Configuration of SIP thread anti-throttling mechanism, the SIP thread does not accept new calls if queue delay gets over this threshold (in milliseconds)
Default: 2000
Requires restart: True
StartCallGeneratorOnStartup - Automatically starts the call generator when the software starts
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True
StarTrinityCrmInstanceId - Identifies the installed software instance in StarTrinity CRM system
Default:
Requires restart: False
StunServerAddress - Address of STUN server which is used to get external IP address for making SIP calls through NAT
Values: 64.69.76.21
Requires restart: True
StunServerPort - Port of STUN server
Default: 3478
Requires restart: True
SymmetricRtp - Enables sending RTP packets back to peer through NAT. If enabled, the software will use source IP address from RTP packet instead of the one which is declared in SDP
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SymmetricSip - Enables sending SIP packets back to peer through NAT. If enabled, the software will use source IP address from SIP packet instead of the one which is declared in Contact header, if the source IP address belongs to range of IANA private IP addresses: 10.0.0.0-10.255.255.255, 172.16.0.0-172.31.255.255, 192.168.0.0-192.168.255.255
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SystemErrorsNotificationEmailIsEnabled - Enables email notification about system errors (from system log)
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False
SystemErrorsNotificationEmail - Specifies where to send email about system errors. It can be one email or semicolon-separated list of emails
Requires restart: False
T38AutoReceivePath - Path for automatic receiving of T.38 faxes. If specified, the software automatically answers to fax RE-INVITE and saves fax as TIFF file into this folder
Requires restart: False
T38AutoSendFileName - TIFF image file name for automatic sending of T.38 faxes. If specified, the software automatically answers to fax RE-INVITE and sends fax from this TIFF file
Requires restart: False
T38LocalMaxPort - Max port number for T.38 media sessions
Default: 14000
Requires restart: True
T38LocalMinPort - Base port number for T.38 media sessions
Default: 12000
Requires restart: True
ThreadPoolSize - Number of threads to initialize and run text-to-speech engine
Default: 4
Requires restart: True
Timer1Ms - RFC3261 T1 timer value in milliseconds (RTT Estimate)
Default: 500
Requires restart: True
Timer2Ms - RFC3261 T2 timer value in milliseconds (The maximum retransmit interval for non-INVITE requests and INVITE responses)
Default: 4000
Requires restart: True
Timer4Ms - RFC3261 T4 timer value in milliseconds (Maximum duration a message will remain in the network)
Default: 5000
Requires restart: True
TimerDMs - RFC3261 timer D value in milliseconds (Wait time for response retransmits)
Default: 32000
Requires restart: True
TlsCertificateOfAuthorityListFile - Certificate of Authority (CA) list file used for TLS (SIPS) transport
Default: StarTrinity.SIPTester.example_cert.pem
Requires restart: True
TlsCertificateFile - Public endpoint certificate file, which is be used as client-side (UAC) certificate for outgoing TLS connection and server-side (UAS) certificate for incoming TLS connections
Default: StarTrinity.SIPTester.example_cert.pem
Requires restart: True
TlsMethod - SIPS TLS transport protocol version: TLS 1.0, TLS 1.1, TLS 1.2
Default: TLSv12
Values: TLSv10, TLSv11, TLSv12
Requires restart: True
TlsPrivateKeyFile - Optional private key file of the endpoint certificate to be used
Default: StarTrinity.SIPTester.example_key.pem
Requires restart: True
TlsPrivateKeyFilePassword - Password to open private key file
Default: startrinity
Requires restart: True
TlsTimeoutS - TLS negotiation timeout to be applied for both outgoing and incoming connection. If set to zero, the SSL negotiation doesn't have a timeout
Default: 3
Requires restart: True
CdrTimestampsMode - Specifies method to get timestamps for CDR fields like DateCreated, DateAnswered, etc. "NIC" - use network adapter timestamps, "PreciseSystemClock" - use DateTime.Now (system time) and QueryPerformanceCounter(), "SystemClock" - use DateTime.Now (system time)
Default: NIC
Values: NIC;SystemClock;PreciseSystemClock
Requires restart: False
UseRawSystemClockTimestampForCdr - Use low-precision function DateTime.Now for CDR fields like DateCreated, DateAnswered, etc
Default: 0
Requires restart: True
UseSipsInRequestLineUriForTls - 1 to use "sips:" in request line when using TLS transport, 0 to use "sip:"
Default: 1
Requires restart: False
UserAgentAndServerHeader - Custom 'User-Agent' or 'Server' header for SIP messages which are sent by the software
Requires restart: False
UseRFC2833ToSendDTMF - Method of sending DTMF
Default: 1
Values: 1 to use RFC2833 (RTP in-band), 0 to use SIP INFO
Requires restart: False
VideoSdpPayloadType - Payload type for video RTP streams
Default: 99
Requires restart: False
WebAdminPassword - WebUI password for "admin". Is set in GUI, encrypted when stored in settings file
Requires restart: False
WebApiTrustedIpAddresses - Trusted (whitelisted) HTTP client IP addresses for web API and web UI to be allowed without HTTP authentication. Multiple IP addresses are separated by semicolon
Values: 127.0.0.1;192.168.0.5
Requires restart: False
WebServerAccessControlAllowOriginHeader - Specifies "Access-Control-Allow-Origin" HTTP header which is returned by the embedded web server
Requires restart: False
WinpcapRtpSenderDscpField - DSCP field bits for transmitted RTP packets in "WinPCAP RTP sender" mode, the bits are transmitted in IP ToS header. ECN bits are set to zero
Default: 0
Values: 0, 46
Requires restart: False
WriteCdrToCsv - Enables saving of CDR reports into daily CSV files
Default: 1
Values: 1 to enable, 0 to disable
Requires restart: False
WriteCdrToDb - Enables saving of CDR reports into database. Database connection is configured with "CdrOdbcConnectionString"
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: False