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APKCombo Apps Communication SIP API Tester Download APK(38 KB) Play On Windows PC 2 W8 for it Download APK(38 KB) Enjoy millions of the latest Android apps, games, music, movies, TV, books, magazines & more. Anytime, anywhere, across your devices. Description Communication Latest Version Version 2 (2) Update Nov 16, 2015 Developer W8 for it Category Communication Google Play ID sipapi.tester Installs 1,000+ App APKs SIP API Tester APK SIP API Tester APP Use this app to see if your device supports the android SIP API. You will receive a True or False output as to whether the API is supported, VoIP is supported, and if SIP is only supported on Wifi. Read more Advertisement Download APK(38 KB) Trending Searches starlink g camera awax zapya google voice king hydra bbc iplayer gcam 6.1 google play services kleinanzeigen vivamax netmod animesaturn kuku fm duolingo chat gpt firefox line deviantart lite See more

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Opens a directory with log files. The files contain debugging information; by default error logs and GUI activity reports are sent to StarTrinity developers via encrypted channel. If you want to disable error reports, please set "DisableAnonymousReports" = "1" in settings tab. License information This screen displays information about license for current installation. For information about prices please contact sales manager by email. Command line interface (CLI) The SIP tester has a command line interface which is available with StarTrinity.SIPTester.CLI.exe. This executable runs in command prompt, it has following parameters: (if a parameter is not specified, it is taken from XML settings file) TotalCalls (required for active mode when generating calls) - total number of outgoing calls to generate, or 'infinite'. After exceeding this number, SIP Tester waits for current calls to complete TotalIncomingCalls (optional) - total number of calls to receive before exiting. After exceeding this number, SIP Tester waits for current calls to complete MaxCurrentCalls (optional) - max number of concurrent calls MaxCurrentCallsLimitMode (optional) - mode to limit max number of concurrent calls: 'incoming+outgoing' (default), 'outgoing', 'incoming' MaxTestDurationSec (optional) - max duration of test in seconds. If the "MaxTestDurationSec" is set, SIP Tester waits for all current calls to complete before exiting. Warning: please limit call duration by "maxansweredtime" to avoid hang calls IncomingCallsExitTimeoutSec (optional) - max time between incoming calls to stop test. Is used when you expect incoming calls to arrive during the test. If the "IncomingCallsExitTimeout" fires, SIP Tester waits for current calls to complete and exits with status -1 OutgoingCallsPeriodMs (optional) - interval between simulated calls in milliseconds NumberOfCallsPerBurst (optional, default is 1) - number of calls to generate per burst (tick) OutgoingCallsMode (optional) - mode of interval between simulated calls: fixed or random OutgoingCallXmlFile (optional) - CallXML file name to generate outgoing calls.. About Sip Tester Album. Sip Tester is a English album released on . Sip Tester Album has 1 song sung by PlanBfrmCVS. Listen to Sip Tester song in high quality download Sip Tester song on Gaana.com.

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"MailSenderServer", "MailSenderFrom", they are configured in SIP Tester on settings screen. 3 and $global.available; != false"> Simulating RE-INVITE (put SIP call on hold) Script for call generator: Script for call receiver: Simulating REFER (call transfer) Script for call generator: Script for call receiver: Identification of IVR messages This script recognizes audio signal in RTP stream, compares it with list of predefined WAV/MP3 files. The resuts are saved into variables and to CDR report. Audio verification is used to test IVR menus and check performance of IVR servers. PESQ MOS (mean opinion score) audio quality measurement This script measures audio quality of IVR message which is played by destination server Client-server-client audio path verification If SIP Tester is installed on both server side and on client side, it could be configured to test IP network for ability to pass RTP audio correctly. Client-side SIP Tester generates SIP calls, transmits RTP audio to the server-side SIP Tester Server-side SIP Tester accepts SIP calls, receives RTP audio and plays exactly same audio back to client (creates loopback connection) If there is a NAT, you should set settings "SymmetricSIP" = "1" and "SymmetricRTP" = "1" on server-side SIP Tester Client-side SIP Tester verifies the received audio if it equals to what it transmitted Checking availability of SIP server using command line interface (CLI) This script makes SIP call to a server and returns SIP status code into .bat file. CallXML file (c:\srcipt.xml) test.bat file launches SIP Tester via CLI, passes server SIP URI as

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In NIC drivers. You can find more details about SIP Tester's performance here. The SIP Tester is able to run as Windows Service - run in background with web interface, start automatically with Windows. To set up it as Windows Service, run StarTrinity.SIPTester.Service.install.bat, to uninstall service - StarTrinity.SIPTester.Service.uninstall.bat Main Window The main window is divided into 3 sections: simulation, reports and status: Simulation section is divided into following tabs: Registrations (UAC) - here you can configure SIP Tester to register at SIP server(s) as client. Registration is needed to receive incoming calls by SIP Tester Registrations (UAS) - here you configure how SIP Tester simulates IP PBX register server: edit list of extensions - users and passwords Outgoing calls simulation - configure script for simulated outgoing calls Incoming calls handling - configure script for incoming calls Manual tests - here you can manually send DTMF to current SIP calls, send REFER or start fax operation Impairments generation - allows you to simulate RTP packet loss and/or jitter, also to terminate SIP Tester and simulate hang calls Stepwise testing - is used to run tests with increasing number of concurrent calls Reports section is divided into following tabs: Current calls - SIP info - displays SIP information about current calls: SIP Call ID, Status Code, Answer Delay etc. Current calls - RTP info - displays RTP information about current calls: jitter, packet loss, G.107 MOS etc. Calls history (CDR) - displays both current and completed calls Lowest quality calls - shows SIP calls ordered by various quality indicators, e.g. calls with largest packet loss Reports/Statistics - shows summary statistics for measured SIP calls, e.g. max. answer delay, mean jitter etc. Performance Chart - visual representation of quality indicators on 2D chart, e.g. jitter vs. number of channels Log - system and

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Call-ID - SIP Call-ID header, unique identifier of the REGISTER request. Is used to trace the request Custom Call-ID - optional field to overwrite default random SIP Call-ID header Use as caller, use as called, single channel - boolean flags, used when generating SIP calls via registrar. In this mode SIP Tester makes calls from one registered SIP user agent to another Enable - allows you to temporarily disable an item without deleting it. Applies to both REGISTERs and INVITEs associated with the item Use for calls only - turns off sending REGISTER for the item, and keeps it usable for INVITEs in mode "make calls via least busy registration" Local IP address - IP address for Contact header Contact header format - allows you to set custom Contact header for REGISTER SIP messages and for INVITE messages (when calling "via least busy registration"). If empty, the software uses a default format for the Contact header. Example: "Eran%20Younger" ;reg-id=1;+sip.instance="" Host, port for calls - used to set a different destination host name and port for SIP calls (INVITE messages) for the UAC registrations in call mode "via least busy registration". When you set custom port for calls, host for calls can not be empty Setting up user agent server (UAS) registrations SIP Tester simulates SIP server (IP PBX) by accepting multiple SIP REGISTER messages from SIP phones (extensions). The registered contact addresses of the extensions can be accessed in CallXML script via substitution $ext(x); UAS registration settings are configured on this screen: Configuring generation of outgoing calls (SIP INVITE sessions) SIP tester acts like a stress call generator. It has a timer which triggers execution of CallXML script. A default pre-installed script generates a call to specified destination(s), plays WAV file and aborts call. The parameters are configured on this

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You need to: Connect your Windows server to a mirroring port on network switch and configure the port to receive all SIP and RTP traffic or install SIP Tester on same Windows server with your PBX Install winpcap, .NET Framework 4.5 and SIP Tester Set operation mode to "Passive" on settings tab Monitor current SIP calls and RTP calls in real time Analyse measured call quality indicators in CDR report and charts (optional) Turn on recording of mixed RTP streams into WAV files in settings -> "pcap export" section (optional) Analyse recorded PCAP files for problematic SIP calls, export audio to WAV files for listening (optional) Analyse sound volume indicators, measured from captured G.711 and G.729 RTP streams (optional) Save CDRs to your database, use your own web interface to show CDRs and recorded WAV files (optional) Configure email alerts and reports for call capacity overloads or call quality drops on settings screen (optional) Configure setting 'SaveSipPacketsToMemory' or 'SaveSipPacketsToDisk' if you want to view SIP trace in UI Alternatively you can run SIP Tester in windows service mode (using StarTrinity.SIPTester.Service.install.bat) or in command-line mode: StarTrinity.SIPTester.CLI.exe OperationMode passive SaveSipAndRtpPacketsToDisk true SaveSipPacketsToDisk true Installation The program is installed with web installer which copies program files into specified directory. Multiple instances can be installed to multiple directories on same server, each instance having different SIP and HTTP ports. This directory is also used to store settings, backups, CallXML scripts, recordings, logs, .pcap and CDR files. SIP Tester can be installed on any Windows OS. It needs .NET Framework and winpcap library to be installed. Windows Server is preferred over Windows 7 or Windows XP because it allows much better performance of UDP/IP stack. Also, real servers or laptops are preferred over virtual machines. A virtual machine may have poor performance and have delays. About Sip Tester Album. Sip Tester is a English album released on . Sip Tester Album has 1 song sung by PlanBfrmCVS. Listen to Sip Tester song in high quality download Sip Tester song on Gaana.com.

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The IVR playback delays are saved to CDR in a custom field Test audio quality (PESQ MOS) in a conference server The conference server should be tested by 2 separate instances of SIP Tester. They can run on the same server on different SIP ports ("LocalSIPPort" setting). The first instance simulates call load of about 30 concurrent calls, it plays silence to the conference. The second instance generates pairs of calls, verifies audio signal and measures PESQ MOS. A pair of calls is generated if you set "number of calls to generate at a time" = "2" Here is a script to simulate the conference in SIP Tester: Test audio quality (PESQ MOS) in a conference server (long-duration calls) SIP Tester should be configured to make N calls in burst, and to limit number of concurrent calls to N. First call in burst plays audio to the conference, secondary calls listen to audio signal from the conference server and measure audio quality Here is a script to simulate the conference in SIP Tester: Test audio quality (PESQ MOS) and DTMF capability of a VoIP route (SIP trunk) in 2 directions The 2 scripts (incoming and outgoing) are used to make test calls via a SIP route to check its audio quality and DTMF passability in 2 ways (from A to B and from B to A) Test outbound dialer The script processes incoming calls which are generated by dialer. It makes random delays, accepts or rejects call. block probability="0.05" >

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Software architecture Basic steps for active testing Basic steps for passive testing (monitoring) or VoIP recording Installation Main window UAC registrations UAS registrations Outgoing SIP calls Incoming SIP calls Current calls report: SIP information Current calls report: RTP information Call Detail Record (CDR) report Lowest quality calls Reports/Statistics Performance chart Stepwise testing Manual tests Impairments generation Settings Log License information Command line interface Web API Web API: UAC registrations Web API: Jobs Screen videos, training sessions Configuring SIP Tester with Cisco Unified Call Manager (CUCM) Report unclarityBasic steps for active testing Basic steps to run a simple active SIP stress test using graphical user interface (GUI) are: Install winpcap .NET Framework 4.5 and SIP Tester Configure UAC registrations Configure outgoing call stress parameters Configure CallXML scripts (test scenarios) via GUI or XML (optional) Configure audio files for playback and/or IVR audio verification, configure other settings Run a test (optional) If you make calls via internet, simultaneously run internet latency test, continuous speed test and VoIP readiness test to double-check internet connection Watch measured VoIP quality indicators in real time: Current calls - SIP indicators Current calls - RTP indicators Reports and statistics like "max jitter", "max packet loss" per call, etc. CDR report History charts (optional) configure email alerts and reports for call capacity overloads or call quality drops on settings screen (optional) listen to recorded audio files, or export results into .pcap files Alternatively you can use command line interface (CLI) with .bat scripts or windows service mode (StarTrinity.SIPTester.Service.install.bat) with Web API to run automated tests Basic steps for passive testing (monitoring) or VoIP recording In passive mode SIP Tester monitors all UDP packets on all network adapters like wireshark. It tries to interpret packets as SIP and RTP. There is no UDP port filter. To monitor SIP calls. About Sip Tester Album. Sip Tester is a English album released on . Sip Tester Album has 1 song sung by PlanBfrmCVS. Listen to Sip Tester song in high quality download Sip Tester song on Gaana.com.

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"CsvDelimiter" (default ";") Send SDP in INVITE - turns on/off sending of RTP session description in INVITE message. If both caller and called parties don't send SDP, SIP call will not initiate RTP stream. This can be used to test SIP signaling without RTP. Forced codec, custom SDP attributes - specify parameters of RTP media session. G.711, G.723, G.729 and user-defined codecs are supported Terminate calls if not answered - specifies timeout before answering (receiving 200 OK). If timeout expires, SIP Tester aborts a call by sending CANCEL. Terminate calls after answering - specifies timeout after answering (receiving 200 OK). If timeout expires, SIP Tester aborts a call by sending BYE. Record mix of RX and TX audio streams - turns on recording into PCMA WAV files. By default files are recorded into program's folder. Play RTP audio from file - selects an audio file to play via RTP stream. WAV, MP3, PCAP formats are supported Record audio streams - turns on recording of RTP streams into PCMA WAV files. RX, TX and mixed recording modes are available. Recordings are used to check audio quality of SIP call and for debugging. Configure jitter buffer settings - shows Settings screen where you can configure parameters of simulated jitter buffer. By adjusting these parameters and checking quality of call you can find the optimal jitter buffer length for your VoIP system. Text editor of CallXML script is used by advanced users who need complex test scenarios. Please refer CallXML documentation and example CallXML scripts to write your own script. Configuring processing of incoming calls (SIP INVITE sessions) The SIP Tester is able to receive multiple SIP calls and simulate IVR servers by executing CallXML script. A default pre-installed script makes a delay, accepts or rejects a call, plays a WAV file and