StarTrinity.com

VoIP software

SIP Tester Change Log

List of regular technical changes to the SIP Tester software
DateDescription
2017-03-24 Softswitch: API method "/API/MainViewModel/Originators/CreateOriginator" to integrate softswitch with VoIP originator's signup webpage
2017-03-13 CallXML: "testId" parameter for "call"
2017-03-06 CallXML: saving multiple SDP attributes with same name into separate CallXML variables (it is useful to access "fmtp" SDP attributes)
2017-03-02 Softswitch: flag "enabled" for routes and tariff rates
2017-02-21 CallXML: getcdrcallscount element
2017-02-19 Reusing previously generated "Authorization" header in re-REGISTER on timer
2017-02-10 CallXML: "floor()", "ceil()", "round()" expressions in element "assign", attribute "mathvalue"
2017-02-02 Implemented saving SIP trace as TXT files, splitted for every SIP call
2017-01-10 CallXML variable "callNumberInBurst" for advanced testing of conference servers
2017-01-04 Improved tables layout in web UI
2016-12-25 Added parameter "sipCallId" into web API method "GetCallsJSON"
2016-12-18 New web API method: /API/MainViewModel/Cdr/GetCallsJSON
2016-12-07 Setting "EnableRtcpXr"
2016-12-06 Supported parsing SDP with SRTP key larger than 128 bits
2016-12-02 Supported SSL v.2, SSL v.3, TLS v.1.2 for SIPS transport
2016-11-30 New API method: GetCurrentCallsCount
2016-11-27 Supported SRV DNS records (for customers in US and Canada)
2016-11-17 Implemented limit of REGISTER requests per second to avoid overloads of tested PBX
2016-11-06 CallXML: implemented modulo operator (%) in "assign:mathvalue" syntax analyser
2016-10-31 CallXML: implemented "considerThisCall" parameter for "ifcallexists" element
2016-10-10 Implemented "ForcedMappedAddress" setting for WinPCAP RTP sender operation mode
2016-09-22 Improved VoIP recording algorithm in case of VAD and RTP clock skew
2016-09-15 Boolean expressions engine: added support of strings in single quotes
2016-09-13 (SIP Tester, softswitch) WebUI to manage audio files
2016-09-08 WebUI: better view of tables in HTML + code cleaning
2016-08-30 Softswitch: username and password authentication mode for originators
2016-08-28 Softswitch: dynamic IP address mode for terminator
2016-08-25 new setting "EnableRtpStatisticsCalculationInPacketAnalyser" to save CPU resources for VoIP recording
2016-08-18 added "direction" parameter into "SetCallXmlVariables" API
2016-08-16 improved linking of SIP call to RTP stream in environmanets with NAT
2016-08-08 SoftSwitch: Minor improvements in CDR and status web UI
2016-08-01 Added CDR field "CalledIP" (requested by some telco)
2016-07-27 SoftSwitch: improved routing
2016-07-10 CDR web UI: added min and max datetime filter
2016-07-05 Extended web API method "CreateCall_Post", added new method "DestroyCall"
2016-07-05 WebUI for passive mode: showing UAC and UAS SIP messages in different colors
2016-07-05 Loading SIP trace from previously saved PCAP file, if displayed call is loaded from CDR data file after restarting the software
2016-07-03 Added setting "CdrDisplaySource" to automatically load calls from CSV CDR files
2016-06-23 Added "TenantId" CDR field for VoIP recording mode
2016-06-21 Added setting "NatList" to explicitly handle NAT when linking captured RTP streams to SIP call
2016-06-12 refactored web authentication procedures
2016-06-12 refactored CDR collection procedures
2016-06-11 SoftSwitch: implemented call transfer mode "suppressAnswerBeforeRbt", parameters "disconnectOnSilenceLevel" and "disconnectOnSilenceTimeout"
2016-06-08 Added CDR field "Recorded_Mix_FileName"
2016-06-06 SoftSwitch: implemented multi-tenancy (usage of single server by multiple independent VoIP business owners, shared hosting)
2016-06-01 Added setting "UseSipsInRequestLineUriForTls" to enable "sip:" request-line URI with TLS transport
2016-05-31 Changed order of displayed calls in CDR: newest calls go first now
2016-05-31 Changed order of log messages: newest first now
2016-05-28 New feature for SIP Tester as VoIP recorder: setting "SaveRtpPacketsToMixedWav", requested by someone from Turkey
2016-05-22 SoftSwitch: implemented AAA and billing, tested
2016-05-08 UAC registrations: new field "Use for calls only" = "Don't send REGISTER"
2016-05-06 CallXML: added element "disableuacregistration" for someone in US
2016-05-06 Refactored writing to system log files
2016-05-05 Refactored procedure of getting timestamps for CDR without WinPCAP: using QueryPerformanceCounter() now
2016-05-03 Improved RTP stream lookup procedure for passive-mode monitoring (for a major telco in Brasil)
2016-04-28 CallXML: "maxansweredtime" parameter for "transfer", "maxansweredtime" event
2016-04-27 Added "LocalIpAddress" field into "UAC registrations"
2016-04-27 CallXML: added "contactHeaderFormat" into "register" element
2016-04-27 CLI: "infinite" TotalCalls parameter value, to allow continuous call generation from command line
2016-04-21 WebUI: call details web page with CDR fields and SIP trace
2016-04-18 CallXML: added "var" attribute into "call" element (requested by customer)
2016-04-16 CallXML: added "from" and "to" attributes into "switch" - "case" elements
2016-04-16 Improved software update procedure
2016-04-12 Extended settings SaveSip(AndRtp)PacketsToDisk_PathPattern settings: now it can contain different patterns for different calls (e.g. incoming and outgoing calls)
2016-04-11 Added a command-line parameter "NumberOfCallsPerBurst" (was requested by customer)
2016-04-07 Sending 200 OK response for PUBLISH SIP request
2016-03-26 CallXML: extended "setcallgeneratorparams" for random interval between calls (for someone in Beirut)
2016-03-23 Added CDR field "RTP packets count"
2016-03-23 Boolean expressions syntax: "contains" operator
2016-03-23 RFC2833 DTMF generation in WinPCAP RTP sender mode
2016-03-21 Manual aborting (hangup) of current calls in GUI
2016-03-20 Improved performance: got 8000 G711 channels with RTP WAV audio playback
2016-03-15 CallXML: renamed parameter "maxtime" into "maxringtime" for "call" element
2016-03-15 Settings for sound card buffer size
2016-03-14 Major improvement in PESQ MOS algorithm: jitter buffer frame loss compensation
2016-03-12 Minor improvements in PESQ MOS algorithm: delay calculation
2016-03-08 CallXML: "expires" parameter for "register" element
2016-03-07 Supported SIP URIs like "sip:xxxx@domain;user=phone" for outgoing calls
2016-03-06 UAC REGISTER performance improvements: simulated 100K extensions without CPU overload
2016-03-02 CallXML: "httpUrl" attribute for "include" element
2016-03-01 Added a setting "WebApiTrustedIpAddresses" to make it easier for applications to use the Web API
2016-02-28 Implemented change of codec in "reinvite" CallXML element
2016-02-27 Setting "CdrPath"
2016-02-27 "Manual tests" window for individual SIP calls: connect to sound device, send DTMF
2016-02-26 "Expires" field for adding batch of UAC registrations
2016-02-24 Sound card module: audio connection to current SIP calls for manual real-time testing and demonstration
2016-02-22 CallXML: "setinterval" element needed for timers
2016-02-22 CallXML: extended "setcallgeneratorparams"
2016-02-22 Added "Require100rel" setting to test PRACK (RFC3262) SIP call flows
2016-02-21 "POST" method for "sendhttprequest" - needed for integration with 3rd party APIs
2016-02-21 API method "/API/MainViewModel/CreateCall" to integrate StarTrinity softswitch with a web application (for someone in France)
2016-02-21 Extended web API framework to support invoking methods. Added an API "/API/MainViewModel/CurrentCallExists" for integration between 2 instances of SIP Tester
2016-02-13 CallXML: "sendhttprequest" element to integrate with third party APIs
2016-02-09 Packet loss concealment (PLC) for G.711 codec, needed to avoid skew in recorded WAV files when getting lost frames in jitter buffer (necessary for PESQ)
2016-02-04 Added setting "DontPrintPortInRequestUri" (requires restart of software)
2016-02-02 "Host for calls" and "port for calls" fields in "UAC registrations", used to set different Request URI for REGISTER and INVITE
2016-02-01 CallXML: "disableRtp" parameter for "call" and "accept" (for someone from Russia)
2016-01-29 Custom "Contact" SIP header format for UAC REGISTER (for someone in Israel)
2016-01-18 SIP Tester GUI: "repeatCount" parameter to read destinations from CSV
2016-01-13 SRTP media transport support
2016-01-12 CallXML: "sipCallId" attribute for "call" element to set custom Call-ID header in INVITE (requested by someone from US)
2015-12-23 Supported SIPS (SIP over TLS transport). Included sample certificate files into installer
2015-12-12 Setting "ReuseExistingTcpSocketToTheSameDestination" (for someone in US, Silicon Valley)
2015-12-06 Selecting previous test ID(s) to load from CDR database
2015-12-06 Enabled using of ODBC drivers in "requestdb" CallXML element
2015-12-05 Loading CDR data from database (via ODBC driver)
2015-12-05 Added new fields into CDR database and CSV file: "100 delay" and "Test ID"
2015-11-27 Generation of "multipart/mixed" SIP packet body
2015-11-20 CDR field and statistics for "SDP negotiation - RTP delay" (for someone in Canada)
2015-11-17 CallXML: "readdb" element to use list of numbers from MySQL, MSSQL, PostgreSQL and other databases
2015-11-13 Setting "WinpcapRtpSenderDscpField"
2015-11-11 WebUI: added "Abort all calls" button
2015-11-10 CallXML: "dynamicblacklist" element to implement fast RAM-based dynamic blacklists
2015-11-10 Achieved RTP jitter less than 1ms with new setting "MediaClockUseSpinWait"
2015-11-08 Supported SIP headers with same name in SIP messages
2015-11-07 CallXML: added "var" parameter for "maxtime" element
2015-10-28 "Max CPS" parameter to control stepwise testing
2015-10-28 Setting "RecordedWavFilesAudioCodec" for some client who uses SIP Tester as SIPREC recorder
2015-10-16 CDR fields "RTP_Called_MinDelta" and "RTP_Caller_MinDelta"
2015-10-13 CallXML: "$timeMs();" syntax element to calculate delays in script (for someone in Canada)
2015-10-02 Settings to set CPU affinity masks for media, SIP threads, and for entire process
2015-09-30 UAC registrations: "expiration interval" setting
2015-09-30 Setting "StartCallGeneratorOnStartup"
2015-09-30 CallXML variable "answerDelay"
2015-09-29 Setting "MediaClockPeriodMs" to minimize transmitted RTP jitter when ptime=20
2015-09-25 Implemented pre-caching of audio files
2015-09-22 Run SIP Tester as windows service - start automatically when windows starts
2015-09-19 CallXML: waitForRingbackToneAbsence, requested by client
2015-09-14 Processing Ctrl+C in command-line mode
2015-09-08 Added CDR field "CallerIP"
2015-09-08 Filter for "reports/statistics", allowing to view stats for subset of calls
2015-09-08 "Max CDR calls count" settings are stored in config file now
2015-08-25 Boolean expression interpreter for CDR filter and CallXML "test" function
2015-08-19 Loading previously saved CDRs from CSV files
2015-08-18 CallXML: "randfile" syntax to play random file from directory (for someone in US, FL)
2015-08-11 UAC registrations: import/export from/to CSV files
2015-08-09 Added option to filter displayed configuration settings by name and description to optimize UX of configuration
2015-08-07 Saving/restoring call generator schedule parameters to/from settings file
2015-08-07 Setting "LocalHTTpPort" = "0" disables VisualCallXML editor - having strange problems on Windows XP
2015-08-05 CallXML: $randdigits(number_of_digits); syntax for random CLI generation
2015-07-30 Added option of saving SIP trace as TXT file and saving to clipboard
2015-07-27 CallXML: reporting line number of CallXML element and its contents in case of error
2015-07-27 CallXML: implemented "less or equal" and "greater or equal" syntax for "if" and "block"
2015-07-26 CallXML: implemented "enabled" attribute to temporarily turn off execution of CallXML elements
2015-07-22 Setting "DisableRtpPacketAnalysisOnIpAddresses" to turn off RTP packet processing on selected interfaces
2015-07-17 Optimized SIP thread: now reaching 1000 calls per second with i7-3770 in "WinPCAP RTP sender" operation mode
2015-07-16 Embedded Visual CallXML script editor into SIP Tester desktop GUI
2015-07-13 WebUI: Fixed compatibility of visual CallXML editor with Internet Explorer and Firefox
2015-07-04 WebUI: Visual CallXML editor
2015-07-02 WebUI: improved call generation page
2015-06-24 CallXML: "localRtpAddress" and "sendSdp" parameters for "reinvite"
2015-06-18 CallXML: "srand" element to reproduce random tests
2015-06-18 CallXML: added "callerId" and "calledId" criteria for "ifcallexists"
2015-06-18 Automatic inserting of custom CDR columns into database schema for every processed call
2015-06-18 CallXML: "waitforringbacktone" element to test SIP trunks and GSM gateways
2015-06-17 Refactoring "signal detector" module and its interface
2015-06-11 "CDR delimiter" setting in GUI - a better representation
2015-06-11 Web interface: "CDR columns configuration" page
2015-06-10 New media processing mode: "WinPCAP RTP sender"
2015-06-07 Setting to control duration of transmitted RFC2833 DTMF signals: DtmfDurationRfc2833Ms
2015-06-07 CallXML: "transferred" event for REFER processing
2015-06-07 Web interface: "current calls" page
2015-06-06 Add batch of UAC registrations: new "password" field
2015-06-06 Fixed processing of Refer-To and Referred-By headers in REFER module
2015-06-03 Support of partial-RTP-frame delays in real-time audio verification algorithm (for someone in Denmark)
2015-06-03 CallXML: "settimeout" element (for someone in Finland)
2015-06-02 CallXML: added a "reinvite" event for custom handling of RE-INVITEs
2015-06-01 CallXML: added "localRtpAddress" parameter to "call" and "accept" to handle multiple network interfaces
2015-05-23 CDR columns "EarlyMediaPeakSignalLevelDb", "ActiveMediaPeakSignalLevelDb"
2015-05-19 Automatic re-register on failure
2015-05-15 CDR database schema verification: automatic creation of table and/or missing columns
2015-05-08 Implemented ringback tone detector. Added CDR field "Latest RBT Delay" which can be used to detect FAS
2015-04-29 CallXML: "on" event handlers now can be located inside current element
2015-04-28 Web interface: CallXML log
2015-04-22 CLI parameter "IncomingCallsExitTimeoutSec"
2015-04-20 Getting CSV delimiter from system regional settings
2015-04-18 Extended "sendsipmessage": added "maxtime", "var", "delayvar"
2015-04-17 Implemented DC offset removing in signal detector. It is useful for G711 codec
2015-04-11 CallXML: "graceful" mode for "exitcli" element
2015-04-10 CallXML: added "level" to "log" element
2015-04-10 Optimized GUI in SIP Tester: display of current SIP calls
2015-04-07 New parameter to make outgoing SIP calls: LocalSipPort
2015-04-04 Added a parameter "debugRecordingThreshold" to "verifyaudio" to debug low confidence scores
2015-03-30 CallXML: "waitforsilence" element for IVR test scripts
2015-03-26 Optimized real-time audio verification algorithm
2015-03-22 UAC registrations: "add batch" - now you can easily add thousand of extensions
2015-03-21 Added setting "LocalSIPPortRange" - now the software can use a range of local SIP ports
2015-03-20 Packet analyzer: implemented TCP packets processing
2015-03-20 Packet analyzer: supported VLAN headers
2015-03-15 Real-time mode for audio verification. In this mode time of verification is calculated automatically
2015-03-13 Improved audio verification algorithm. Now it is faster and more accurate
2015-03-10 Command line parameter "TotalIncomingCalls"
2015-03-10 CallXML: "exitcli" element
2015-03-01 Command line parameter "RetryToOpenLocalSipPort" to avoid hanging of CLI when another program already uses local SIP port
2015-03-01 Added settings to configure SMTP server port number and SSL mode
2015-02-23 CallXML: recordcall parameter "mode" - "rx", "tx", "mix"
2015-02-20 CallXML: Implemented dynamic referencing of variables: "$prefix$indexVar;;" and "$prefix[$indexVar;];"
2015-02-19 CallXML: "exec" element, similar to "eval" in JavaScript
2015-02-18 Sending 200 OK response for OPTIONS SIP request, it is used to ping SIP servers
2015-02-18 Setting "OverwriteCsvCdrFilesAfterRestart"
2015-02-17 CallXML: "recordmessage", "searchfile" elements
2015-02-15 CallXML: "$env.x;" syntax to use Windows environment variables in scripts
2015-02-15 CallXML: "eofMode", "rowIndex" parameters for "readcsv"
2015-02-12 Editing CSV read start index in GUI
2015-02-12 CallXML: "enqueue", "dequeue" elements
2015-02-11 CallXML: "getaudiofilelength" element
2015-02-10 CallXML: syntax for default parameter values of functions
2015-02-10 Developed a new fast algorithm for audio verification. Tested with 1200 concurrent G.711 calls on 4x3.4GHz i7 CPU
2015-02-08 CallXML: "getfilenamefrompath" element
2015-02-08 Decoding G.729 and G.711 RTP streams and exporting to WAV files
2015-02-08 New CDR field: "audio signal level" for both entire call and early media
2015-02-07 CallXML: "includefunctions" element to build complex IVR tests
2015-02-07 Changed "Total calls per second" calculation: now it uses time between first and last call instead of total measurement time
2015-02-05 CallXML: added "cache" parameter to "include"
2015-02-05 PCAP file import
2015-01-27 Created a button to create burst of SIP calls manually
2015-01-22 CallXML: "getstringlength" to operate with length of dialed numbers or any other strings
2015-01-21 CallXML: option to start calls from the middle of CSV file - extended $seq_sip_uri_from_csv() parameters
2015-01-19 command line: added parameters "LogLevel" and "OperationMode"
2015-01-15 CallXML: "switch", "case", "default" syntax for better routing and unit tests
2015-01-09 UAS registrations page: added "SIP Trace" button for debugging
2015-01-08 CallXML: "setcallgeneratorparams" element and "$timeOfDayInHours();" keyword to simulate variable call load
2015-01-08 Added "callerIpAddress" variable for IP address restriction logic
2015-01-06 Fixed issue with MSI installer "auto-healing" which reverted settings
2015-01-03 "UAS registration and authorization" page
2014-12-25 Web interface: developed "CDR" page
2014-12-24 Developed CDR filter
2014-12-23 Implemented SIP trace view for passive mode in SIP Tester
2014-12-14 Improved performance of CDR datagrid, added option to hide columns
2014-12-12 Web UI: created buttons to create calls, start and stop call generator
2014-12-12 Basic Web API for softswitch and SIP Tester is ready for use
2014-12-11 "Enabled" setting for UAC registrations: some customers need to disable registration without deleting it
2014-12-09 CallXML: "startCallGenerator", "movingAveragePut", "movingAverageGet", "movingAverageReset" elements
2014-12-09 Developed code to restore WinPCAP capture thread after sleep mode
2014-12-07 If there is XML error in both GUI and WebUI, CallXML script is not applied until it is fixed, so live calls are not affected
2014-12-05 Executing custom command line when MOS drops below a threshold
2014-12-05 Developed new RTCP VoIP quality CDR fields, history charts. Updated CDR table schema
2014-12-02 Simulation of RTP jitter
2014-11-29 Added an option of SIP proxy for INVITE messages
2014-11-29 CallXML: added "localRtpPort" parameter to "call" and "accept" for explicit RTP port number assignment
2014-11-28 Included "Allow" header into REGISTER message
2014-11-27 Added "auth. user name" setting for UAC registrations (for someone in San Francisco Bay Area)
2014-11-26 Added settings "UseAsCaller" and "UseAsCalled" for UAC registrations
2014-11-24 Created web interface
2014-11-20 Implemented sending of RTCP and RTCP-XR reports to third party destination for analysis and monitoring
2014-11-20 CallXML event "audioSignal" for "call" and "transfer" elements
2014-11-17 Added CDR field "ReleasedBy"
2014-11-17 "random least busy registration" call mode: select least busy extension for both caller and called side
2014-11-17 Implemented recording of SIP and RTP packets to .pcap files for further debugging
2014-11-13 Added SIP proxy setting for UAC REGISTER
2014-11-11 Moved CallXML scripts into the single settings XML file
2014-11-08 Setting "RecordSilenceWithoutRtp" to put silence into recorded WAV files when no RTP is received yet
2014-11-08 Setting "DefaultSignalDetectorThresholdDb" to change the threshold which is used for "SignalDelay" CDR field
2014-11-07 Fixed an issue related to RTCP and NAT
2014-11-06 Optional From/To host name for UAC REGISTER
2014-11-04 CallXML: "callsLeftConference" event
2014-11-03 RTCP-XR reports (G.107 MOS and R-factor fields are not ready yet)
2014-11-02 Extended freeware license: now it allows 20 concurrent calls, unlimited attempted/received calls
2014-11-01 Added CDR fields: RTP_Caller_Delay, RTP_Called_Delay - time between INVITE and first RTP packet
2014-10-30 Compiled DLLs for both 64 and 32 bit modes
2014-10-27 Moved to .NET 4
2014-10-25 CallXML: added "getcallgeneratorinfo" element
2014-10-23 Supported "audio/telephone-event" content type for SIP INFO DTMF packets
2014-10-22 CallXML: "repeat" parameter for "inputdigits"
2014-10-22 System error notifications by email
2014-10-21 Added an option to generate multiple SIP calls in a burst
2014-10-19 Added a button to make pause when generating calls in SIP Tester
2014-10-12 CallXML: "maxansweredtime" parameter for "call"
2014-10-12 CallXML: "startswith" keyword for test syntax
2014-10-11 Added "Created" and "Answered" fields to "Current calls SIP info"
2014-10-10 CallXML: added "minTimeAboveLevel" to "waitforsignal"
2014-10-09 Saving CDR records to database using ODBC driver
2014-10-07 Added "ifcallexists", "getcallinfo" CallXML elements
2014-10-06 Supported standard codecs with dynamic payload type (pt <= 96)
2014-10-03 Separated CallXML and system logs in SIP Tester's GUI
2014-09-27 "if" and "else" syntax elements
2014-09-27 Added "OR" and "AND" syntax elements into "block"
2014-09-27 Option of sending email reports at specific time of day with reset of statistics
2014-09-25 Added jitter and answer delay indicators into status panel, so they are always visible for control
2014-09-22 Removed a ":" in default User-Agent field, because it conflicts with Zultys IP-PBX
2014-09-19 Added lower threshold into email alerting module: now it sends email also when call quality becomes normal again
2014-09-18 Developed "measuresignal" CallXML element to analyse received RTP stream
2014-09-17 Added "level" parameter to "waitforsignal" to configure threshold of measured signal
2014-09-16 "substring" element for CallXML scripts
2014-09-15 Added CPU load display to the main screen
2014-09-12 Developed "function" and "runfunction" elements
2014-09-11 Added "ifconferenceexists" CallXML element
2014-09-11 Added "stopaudio" CallXML element to stop audio player which was started asynchronously
2014-09-09 Added "getstatistics", "resetstatistics" CallXML elements to access statistics programmatically
2014-09-08 Implemented call rate (CPS) threshold for email alerting
2014-09-08 Added "mathvalue" attribute for "assign" CallXML element
2014-09-07 Implemented $randchoice(option1,option2,option3,...) substitution for scripts
2014-09-07 Implemented "less than" and "greater than" conditions for "block" and "goto" CallXML elements
2014-09-07 Added "syncDestroy" mode for "conference" CallXML element
2014-09-02 Refactored GUI, added split panels: simulation and results
2014-08-27 R2S (Real Time Session Supervision Protocol) for ED-137 tests
2014-08-24 Re-register button
2014-08-23 Automatic backup of settings and CallXML scripts
2014-08-20 GUI for codec settings and RTP playback
2014-08-20 Added "maxtime" to "waitforsignal" CallXML element
2014-08-20 Renamed setting "CsvCdrDelimiter" into "CsvDelimiter", used it for both reading and writing CSV files
2014-08-10 Uniform probability distribution of interval between generated SIP calls
2014-08-10 CDR filter by SIP status code
2014-08-04 "stoptest" CallXML element
2014-08-02 Defined status code 1408 (NoResponse)
2014-07-20 "getwg67info" CallXML element
2014-07-18 Added settings for recorded WAV file names: "RecordedWavFileNamePattern" and "DebugMediaFileNamePattern"
2014-07-18 Added "contentType" attribute to "sendsipinfo" CallXML element
2014-07-16 Saving remote SDP attributes into CallXML variables (is intended for ED-137 tests)
2014-07-11 Support of RTP header extensions for ED-137 ATM VoIP testing
2014-07-08 Support of user-defined codec with audio/video playback from PCAP file
2014-06-30 Exporting/importing setings from/to XML file
2014-06-26 Export of SIP and RTP packets into .pcap files for further debugging
2014-06-02 Custom SIP headers and SDP attributes for both INVITE requests and responses
2014-06-02 Generation of email alerts and reports on call capacity overloads and low audio quality detections
2014-05-22 Extended statistics of SIP packets (for someone in Spain)
2014-05-21 Configurable "User-Agent" and "Server" header (for someone in Sweden)
2014-04-25 G.107 E-model R-factor and MOS measurement
2014-04-24 Improved RTP impairments generation
2014-04-13 History and percentiles chart for all performance indicators
2014-04-11 Configurable total attempted calls count (for someone in US, NJ)
2014-04-05 Configurable RTP TX packet time (for someone in Mexico)
2014-04-03 "waitforsignal" CallXML element (for someone in Paris)
2014-04-02 Configurable SIP timers (T1-T4)
2014-04-02 List of random options for "assign" CallXML element
2014-03-17 Support of any WAV and MP3 files
2014-03-10 Max number of calls per destination (for someone in US, TX, Dallas)
2014-03-02 Passive (non-intrusive) monitoring module (capturing and analyzing VoIP packets with winpcap)
2014-02-19 "register" CallXML element for automated tests with SIP REGISTER (for someone in Singapore)
2014-02-15 Generation and processing calls without RTP
2014-02-09 Command line interface
2014-02-03 More accurate RTP jitter measurement with winpcap library
2014-01-24 Added "skipHeaders" = "true/false" attribute to "readcsv" (for someone in Denmark)
2014-01-23 Created "replacestring" CallXML element (for someone in Austria)
2014-01-23 Created "writefile" CallXML element (for someone in Austria)
2014-01-22 Created "loopbackaudio" CallXML element to test roundtrip delay
2014-01-21 Added "probability" attribute to "verifyaudio" CallXML element
2014-01-21 Custom CDR CSV field delimiter (for someone in US, FL)
2014-01-18 Allowed asterisk (*) in SIP user name (for someone in Netherlands)
2014-01-14 Added button to save reports as HTML file
2014-01-12 Added display of least quality calls, sorted by custom CDR field like MOS
2014-01-05 Improved performance and memory usage of audio playback
2014-01-04 MOS (mean opinion score) reports for VoIP audio quality testing
2013-12-18 Allowed usage of CallXML script variables as $rand() parameters
2013-12-12 Custom called number for "Make calls via least busy UAC registration"
2013-12-12 Audio verification for testing round trip delay, 3-way conference calls, IVR menus
2013-12-09 Added "countsColumnIndex" attribute to "readcsv" CallXML element
2013-12-09 Added "mode" (="random") attribute to "readcsv" CallXML element
2013-12-07 Added "repeat" attribute to "readcsv" CallXML element
2013-12-03 Generation of high call loads from multiple distributed instances of SIP Tester, installed on multiple servers
2013-11-19 Measuring delay between simulated DTMF and IVR audio response
2013-11-19 Saving custom variables into CDR CSV file
2013-11-18 Added sorting of least quality calls by answer delay, 180/183 delay, by status code
2013-11-17 Trial and one-time license mode
2013-11-16 TCP transport for REGISTER
2013-11-12 Option to update version in GUI
2013-11-12 Time column in log
2013-11-10 'readcsv' CallXML element
2013-11-10 'Non-silence' RX/TX audio duration field in CDR
2013-11-10 More accurate calculation of lost/delayed RTP packets' percentage
2013-11-09 Option to view content of tabs in separate windows
2013-11-08 Added packet loss percentage field into CDR file
2013-11-07 Included reseller ID into installer for affiliate links
2013-11-04 History chart on statistics tab
2013-11-01 Performance chart
2013-10-20 Stepwise testing
2013-09-18 Automatic measuring of post-dial delay (PDD) by detecting dial tone in RTP stream
2013-09-03 Saving of CDR into CSV files
2013-08-25 Making calls to list of destinations in CSV file
2013-08-20 TCP transport for SIP calls
2013-08-03 Added feature of sending custom SIP messages
2013-05-06 Moved from WinForms to WPF
2012-10-25 Released public freeware version at startrinity.com
2008-02-07 Released first version for internal testing of a call center

Note: this is not full list of implemented features.

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