StarTrinity.com

VoIP software

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"

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"

AffinityMaskForProcess - Binary mask specifying which CPU cores are used by the software. Example: "1110" - "use core #1, #2, #3, don't use core #0"

AffinityMaskForSipThread - Binary mask specifying which CPU cores are used by the SIP thread. Example: "0010" - "use core #1"

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

AudioVerificationThreadsCount - Number of threads used for offline audio verification (PESQ MOS mode)
Default: 2

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

AutoUpdate - Enables automatic updating of software from web server
Values: 1 to enable, 0 to disable

BackupSettingsAndScripts - Enables automatic backup
Default: 1
Values: 1 to enable, 0 to disable

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

CdrOdbcConnectionString - Specifies connection to database where to save CDR records
Values: Driver={SQL Server};Server=localhost\SQLEXPRESS;Database=testCDR;Uid=sa;Pwd=pass;

CdrPath - Path to generated CSV CDR files
Default: [program_folder]\CDR

CdrTableName - CDR table name in database

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

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

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

ConnectAllCallsToSoundCardMicrophone - Enables automatic audio connections from sound card microphone to all current SIP calls
Default: 0
Values: 1 to enable, 0 to disable

ConnectAllCallsToSoundCardSpeaker - Enables automatic audio connections from all current SIP calls to sound card speakers
Default: 0
Values: 1 to enable, 0 to disable

CsvDelimiter - Delimiter between fields in CSV file, used to read or write CSV files. Default value is taken from system settings
Values: ; or ,

DeclareAllCodecsInSdpAnswer - Enables declaration of all locally available codecs in SDP answer
Default: 0
Values: 1 to enable, 0 to disable

DeclareG711AInSDP - Enables declaration of audio codec G.711A (payload type = 8) in SDP
Default: 1
Values: 1 to enable, 0 to disable

DeclareG711UInSDP - Enables declaration of audio codec G.711U (payload type = 0) in SDP
Default: 1
Values: 1 to enable, 0 to disable

DeclareG723InSDP - Enables declaration of audio codec G.723 (payload type = 4) in SDP
Default: 1
Values: 1 to enable, 0 to disable

DeclareG729InSDP - Enables declaration of audio codec G.729 (payload type = 18) in SDP
Default: 1
Values: 1 to enable, 0 to disable

DebugMedia - Enables writing of RX audio streams of each call
Default: 0
Values: 1 to enable, 0 to disable

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]

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

DebugMediaTX - Enables writing of TX audio streams of each call
Default: 0
Values: 1 to enable, 0 to disable

DeclareRFC2833InSDP - Enables declaration of RFC2833 payload type in SDP offer and answer
Default: 1
Values: 1 to enable, 0 to disable

DefaultLocalIpAddress - Selects network interface which is used if GetBestInterface() WinAPI function returns 127.0.0.1

DefaultSignalDetectorThresholdDb - Default signal detector threshold level in decibels. The threshold is used for "AudioSignalDelay" CDR field
Default: -24

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

DestroyCallOnSessionTimerExpiry - Enables termination of dead SIP calls if session timer is expired
Default: 1
Values: 1 to enable, 0 to disable

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

DetectedRtpStreamMaxIdleTimeMs - Timeout to remove detected RTP streams from memory when no more RTP packets are detected
Default: 10000

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

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

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

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

DisableUacUnregisters - Disables sending un-REGISTERs in UAC registrations
Default: 0
Values: 1 to disable, 0 to enable

DontPrintPortInRequestUri - Disables printing of port number in Request URI of INVITE SIP message
Default: 0
Values: 1 to disable, 0 to enable

DtmfDurationRfc2833Ms - Duration of transmitted RFC2833 (inband, RTP) DTMF signals
Default: 100
Values: Some SIP servers need at least 200ms DTMF signals

DtmfSendingIntervalMs - Interval between consequent RFC2833 (inband, RTP) DTMF signals
Default: 200

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

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

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

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

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

EnablePLC - Enables packet loss concealment (PLC) for G.711 codec
Default: 0
Values: 1 to enable, 0 to disable

EnableR2S - Send R2S packets when WG67 PTT is off. Applies to TX RTP only
Default: 0
Values: 1 to enable, 0 to disable

EnableRtcpXr - Enables sending of RTCP extended reports (RTCP XR)
Default: 1
Values: 1 to enable, 0 to disable

EnableRtpStatisticsCalculationInPacketAnalyser - Enables calculation of RTP jitter, packet loss, G.107 MOS in packet analyser module.
Default: 1
Values: 1 to enable, 0 to disable

EnableSignalDetector - Enables audio signal measurement for "-24dB Audio Signal Delay" CDR field
Default: 1
Values: 1 to enable, 0 to disable

EnableSignalDetectorMaxLevelMeasurementsForCdrFields - Enables measurements for CDR fields: EarlyMediaPeakSignalLevelDb, ActiveMediaPeakSignalLevelDb
Default: 0
Values: 1 to enable, 0 to disable

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

EnableSoundCardModule - Turns on sound card module
Default: 0
Values: 1 to enable, 0 to disable

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

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

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

ForcedAudioCodec - Forced audio codec preference
Values: 0: PCMU (G.711 mu-law), 8: PCMA (G.711 a-law), 4: G.723, 18: G.729

ForcedLocalAddress - Specifies custom IP address of media stream for SDP offer

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

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

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

GuiUpdateIntervalMs - GUI update interval in milliseconds. Set 0 to disable automatic GUI updates on timer, to save CPU resources
Default: 2500

HttpProxyPassword - HTTP proxy password, is set in GUI and encrypted in settings file, used for updating software from developer's web server

HttpProxyUrl - HTTP proxy, used for updating software from developer's web server
Values: http://192.168.1.1:8080

HttpProxyUserName - HTTP proxy user name, used for updating software from developer's web server

HttpServerName - Local HTTP and HTTPS (web interface) server name

IdleMediaTimeToEndSession - Timeout for termination INVITE session if no media packets are received from peer, in seconds
Default: 200

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

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

JitterBufferMinPre - Minimum delay that must be applied to incoming packets, in milliseconds
Default: 10

LocalHttpPort - HTTP port used for web interface
Default: 19019
Values: 0 - disable HTTP web interface and Visual CallXML editor

LocalHttpsPort - HTTPS port used for web interface
Default: 19020
Values: 0 - disable HTTPS web interface

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

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

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

LocalSIPPort - Specifies UDP and TCP port number to send and listen SIP messages
Default: 5060
Values: 0 - use random non-busy port

LocalSIPPortRange - Specifies local SIP ports range, if using multiple local SIP port numbers
Default: 1

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

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

LogLevel - Filter level for log file writer
Default: 0
Values: 0: Error, 1: Warning, 2: Info, 3: Debug, 4: Trace

LogPath - Path to logs. If not set, logs are stored in directory near exe file
Default: [program_folder]\Logs

MailSenderFrom - SMTP parameter for sending email
Values: myusername@gmail.com

MailSenderPassword - SMTP server password for sending email. Is set in GUI and encrypted when stored in settings file

MailSenderPort - SMTP server port for sending email
Default: 25

MailSenderServer - SMTP server domain name for sending email
Values: smtp.gmail.com

MailSenderUserName - SMTP parameter for sending email
Values: myusername@gmail.com

MailSenderUseSsl - SMTP server transport level encription
Values: 0 or 1 or [empty - try to send email both with and without SSL]

MaxCallLifeTimeInHours - Timeout to forcefully abort deadlocked/zombie/hang SIP calls
Default: 48

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

MaxMemoryCdrCallsCount - Max number of calls to store in CDR memory
Default: 2000

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

MaxDetectedCallDurationWithoutSipMessagesInMinutes - Timeout to forcefully abort deadlocked/zombie/hang captured SIP calls in "passive" mode
Default: 60

MaxRegistrationsPerSecond - Limit of new outgoing (UAC) REGISTER requests per second. Is set to avoid overloading of registration server or PBX
Default:

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

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

MediaReceiverThreadsCount - Number of concurrent threads which receive RTP/UDP data and put it into jitter buffer
Default: 8

MediaThreadsCount - Number of concurrent threads which process (encode and decode, mix) and send RTP audio data
Default: 16

MediaTransportPoolInitialSocketsCountPerInterface - Number of RTP sockets (local RTP ports) to open initially, during setup
Default: 4

MediaTransportPoolMinPort - Base port number for RTP sockets
Default: 16000

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;...

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

Realm - Realm, is used for SIP authentication
Default: *

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]

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

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

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

RemoteRtcpMonitorAddress - Specifies an optional third-party destination (monitor) host name where to send RTCP and RTCP-XR packets

RemoteRtcpMonitorPort - Specifies an optional third-party destination (monitor) port where to send RTCP and RTCP-XR packets

Require100rel - Enables PRACK behaviour and "Require: 100rel" header for UAC and UAS SIP calls
Default: 0
Values: 1 or 0

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

RequireAuthorizationForRequestsFromRegistrar - Enables authentication of INVITE messages received from trunks/SIP registrars (UAC registrations)
Default: 0
Values: 1 or 0

RequireSRTP - Mandatory SRTP media transport mode. Requires "RTP/SAVP" transport declared in SDP. See also other mode: 'AllowSRTP'
Default: 0
Values: 1 or 0

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

RFC2833RXPayloadType - Expected RFC2833 DTMF event payload type
Default: 101

RtcpSendIntervalMs - Interval to send RTCP packets
Default: 1000

RtcpXrSendIntervalMs - Interval to send RTCP-XR packets
Default: 1000

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

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

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

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

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

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

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

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

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

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

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

SessionRefresher - RFC 4028 session refresher
Default: uac
Values: uac or uas

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

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

StunServerAddress - Address of STUN server which is used to get external IP address for making SIP calls through NAT
Values: 64.69.76.21

StunServerPort - Port of STUN server
Default: 3478

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

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

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

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

T38LocalMaxPort - Max port number for T.38 media sessions
Default: 14000

T38LocalMinPort - Base port number for T.38 media sessions
Default: 12000

ThreadPoolSize - Number of threads to initialize and run text-to-speech engine
Default: 4

Timer1Ms - RFC3261 T1 timer value in milliseconds (RTT Estimate)
Default: 500

Timer2Ms - RFC3261 T2 timer value in milliseconds (The maximum retransmit interval for non-INVITE requests and INVITE responses)
Default: 4000

Timer4Ms - RFC3261 T4 timer value in milliseconds (Maximum duration a message will remain in the network)
Default: 5000

TimerDMs - RFC3261 timer D value in milliseconds (Wait time for response retransmits)
Default: 32000

TlsCertificateOfAuthorityListFile - Certificate of Authority (CA) list file used for TLS (SIPS) transport
Default: StarTrinity.SIPTester.example_cert.pem

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

TlsMethod - SIPS TLS transport protoicol version: TLSv1, SSLv2, SSLv3, SSLv23
Default: TLSv1
Values: TLSv1, SSLv2, SSLv3, SSLv23 (=TLSv1.2)

TlsPrivateKeyFile - Optional private key file of the endpoint certificate to be used
Default: StarTrinity.SIPTester.example_key.pem

TlsPrivateKeyFilePassword - Password to open private key file
Default: startrinity

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

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

UseSipsInRequestLineUriForTls - 1 to use "sips:" in request line when using TLS transport, 0 to use "sip:"
Default: 1

UserAgentAndServerHeader - Custom 'User-Agent' or 'Server' header for SIP messages which are sent by the software

UseRFC2833ToSendDTMF - Method of sending DTMF
Default: 1
Values: 1 to use RFC2833 (RTP in-band), 0 to use SIP INFO

WebAdminPassword - WebUI password for "admin". Is set in GUI, encrypted when stored in settings file

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

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

WriteCdrToCsv - Enables saving of CDR reports into daily CSV files
Default: 1
Values: 1 to enable, 0 to disable

WriteCdrToDb - Enables saving of CDR reports into database. Database connection is configured with "CdrOdbcConnectionString"
Default: 0
Values: 1 to enable, 0 to disable

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