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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

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

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

CdrCsvFilesSplitMode - CDR CSV files split mode: "day" or "month"
Default: day
Values: day, month
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

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 and TX audio streams. May end with '\', or may not end with '\'
Default: C:\debug_media
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
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
Default: 1
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

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

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

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

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

EnableSimManagement - Enables SIM management features when the software runs in softswitch mode for GSM termination
Default: 0
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

EnableTls - Enables TLS (SIPS) transport for SIP messages
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True

EnableWinpcapRtpSender - Enables "WinPCAP RTP sender" operation mode. In this mode the RTP packets are sent using fast WinPCAP raw sockets
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 custom IP address of media stream for SDP offer
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

GoogleSpeechAPIv2Key - Google speech API key, used in "inputspeech" CallXML element to recognize speech (speech-to-text)
Requires restart: False

GoogleSpeechAPIv2Language - Google speech API language, used in "inputspeech" CallXML element to recognize speech (speech-to-text)
Default: en-us
Requires restart: False

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
Default: 200
Requires restart: False

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

LocalHttpsCertificateSubjectName - HTTPS certificate common name, in "LocalMachine" or "LocalUser" certificate store
Values: sip.yourdomain.com, localhost
Requires restart: True

LocalNodeServiceHost - Local IP address to accept connections from "master" instance of SIP Tester. Is configured at "slave" instances when using SIP Tester in "scaled multiple instances" mode
Requires restart: False

LocalNodeServicePort - Local TCP port to accept connections from "master" instance of SIP Tester. Is configured at "slave" instances when using SIP Tester in "scaled multiple instances" mode
Default: 8085
Requires restart: False

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
Requires restart: True

LocalTlsPort - Specifies TLS TCP port number to send and listen encrypted SIP/TCP messages
Default: 5061
Requires restart: True

LogsDeleteIntervalInDays - Time period for deleting old logs and HTML reports
Default: 7
Requires restart: True

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

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

MaxDetectedCallDurationWithoutSipMessagesInMinutes - Timeout to forcefully abort deadlocked/zombie/hang captured SIP calls in "passive" mode
Default: 60
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, during setup
Default: 4
Requires restart: True

MediaTransportPoolMinPort - Base port number for RTP sockets
Default: 16000
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

OverwriteCsvCdrFilesAfterRestart - Enables overwriting of CSV CDR files after restart of the software
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True

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: 50000
Requires restart: True

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

ReuseExistingTcpSocketToTheSameDestination - Enables re-using of already opened socket (local TCP port) when sending INVITE/REGISTER to the same destination IP address and port
Default: 1
Values: 1 or 0
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

RtpTxPacketTime - Delay between transmitted RTP packets, RTP packet duration in milliseconds
Default: 20
Values: 10, 20, 30, 40, 50
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: False

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: 3600
Requires restart: False

StartCallGeneratorOnStartup - Automatically starts the call generator when the software starts
Default: 0
Values: 1 to enable, 0 to disable
Requires restart: True

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 protoicol version: TLSv1, SSLv2, SSLv3, SSLv23
Default: TLSv1
Values: TLSv1, SSLv2, SSLv3, SSLv23 (=TLSv1.2)
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

UseNicTimestampsForCdr - Use NIC timestamps instead of system time for CDR fields like DateCreated, DateAnswered, etc
Default: 1
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

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

WinpcapRtpSenderDscpField - IP DSCP (ToS) field for transmitted RTP packets in "WinPCAP RTP sender" mode
Default: 0
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

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