Measuring quality

StarTrinity SIP Tester™ (call generator, simulator) - VoIP monitoring and testing tool

Windows server(s) or laptop(s)
StarTrinity SIP Tester performance:
Intel Core i7-3700 4x3.9GHz
8000 concurrent G.711 calls
per 1 server
VoIP network: corporate LAN, WAN, internet, VPN, wireless LAN
RTCP, T.38

cloud IP-PBX, IVR, call center

SIP gateway, trunk, VoIP provider

in-house IP-PBX, IVR, call center

SIP phones
  • WAV, MP3, PCAP playback and recording
  • RTP audio verification
  • Jitter, packet loss, RTT measurement

StarTrinity SIP Tester™ is a VoIP load testing tool which enables you to test and monitor VoIP network, SIP software or hardware. It is able to simulate and passively monitor thousands of simultaneous incoming and outgoing SIP calls with RTP media, analyze call quality and build real time reports. Call flow is specified by CallXML script where one can design various situations that can cause failure of tested SIP stack. The SIP Tester runs on any Windows PC without special hardware and simulates application server, media server, SIP phone or register server. Freeware license of SIP tester allows 50 actively simulated concurrent calls and unlimited passively monitored or recorded calls. For extended number of calls commercial license is available. The software is licensed and protected by law (see license agreement for details). The unlimited license for the SIP Tester is free for non-profit medical organizations (hospitals, research institutes), charity, and nature protection organizations. An alternative to SIP Tester for VoIP wholesalers is "dialer campaigns" module in StarTrinity Softswitch - you can use it to generate VoIP traffic from our server or your server, using a web interface.

Key features

  • Free unlimited non-intrusive monitoring of existing VoIP deployment, capturing, recording and analyzing SIP/RTP packets with realtime reports and charts (passive testing, free)
  • Outgoing and incoming SIP calls simulation (active testing, limited in freeware license. commercial licenses are available)
  • High performance: one i7 server can simulate 5400 simultaneous G.729 calls or 2600 G.711 calls (SIP+RTP)
  • Operating system: Windows XP, Windows 7, Windows 8, Windows 10, Windows Server 2003, Windows Server 2008, Windows Server 2012, Windows Server 2016, Windows Server 2019
  • G.107 E-model and PESQ (P.862.1) MOS and R-factor measurement
  • CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex
  • Custom WAV/MP3/PCAP file RTP audio and video playback
  • SIPREC: generate SIP calls with 2 RTP streams and play stereo WAV file to test recorders
  • Custom SIP headers and SDP attributes
  • Export of SIP and RTP packets into separate (per call) pcap files
  • Support of any user-defined audio/video codec: RTP playback from .PCAP file
  • Command line interface for SIP unit tests
  • Detection of ringback tone audio signal from received RTP packets; post-dial delay (PDD) measurement
  • Audio verification: IVR Tests, conference audio path tests (CallXML element verifyaudio)
  • Jitter buffer simulation with configurable settings
  • Real time analysis of voice quality for all RTP streams, calculation of global min MOS, global max jitter, etc
  • RTP jitter, packet loss percentage, answer delay measurement
  • Real time reports and charts for every individual call or for all calls
  • Email alarms (alerts) and reports on SIP trunk call capacity overloads and low audio quality detections
  • Audio recording: mixed and separate RX/TX streams. One can listen recordings and check voice quality
  • Audio connection with sound card: speak+listen to simulated SIP calls and check voice quality in real time
  • CDR reports: basic call information, RTP statistics, recorded file. Export to CSV files or database. Comprehensive filters allowing searching in CDR database by telephone numbers, qualitative parameters (loss/delay/MOS), codecs
  • Lowest quality calls report: easy to see the worst call in a test
  • Supported audio codecs: G.711, G.723, G.729.
    PCAP pass-through mode (RTP from/to PCAP): G.711, G.723, G.729, G.722, GSM, iLBC30, iLBC30, Opus, Silk, Speex, custom
  • Supported video codecs: H.264, H.263+, VP8, any other from pcap file
  • Support of RTP header extensions for ED-137 air traffic management (ATM) VoIP tests. Measurement of round-trip delay
  • R2S (Real Time Session Supervision Protocol) for ED-137 tests
  • T.38 fax support
  • DTMF generation and detection: RFC2833 and SIP INFO
  • Bulk generation of SIP calls: manually and on timer
  • Receiving calls with and without registration at SIP Tester
  • Impairments simulation: SIP and RTP packet loss
  • UAC and UAS registration
  • SRTP (encrypted media): RFC3711, RFC4568
  • SIP over TCP, TLS and UDP transports
  • SIP authentication: digest security scheme
  • Penetration testing
  • False answer supervision (FAS): simulation of fake FAS and detection of FAS for simulated and live calls
  • Optional early media for incoming and outgoing calls
  • Reading list of destination numbers and SIP accounts from CSV file for both INVITEs and REGISTERs
  • Integration with databases via ODBC: saving CDR reports.measurements, retrieving list of numbers for dialing. MSSQL, MySQL, MariaDB, PostgreSQL
  • Simulation of putting on hold (RE-INVITE) and transfer (SIP REFER)
  • Loopback audio connection (between RX and TX RTP audio streams)
  • RTP audio signal level measurement in dB
  • Free VoIP recording: capturing G.711 and G.729 RTP streams into WAV files via mirroring port, saving CDRs to your database via ODBC driver
  • Supported specifications: RFC2833, RFC2976, RFC3261, RFC3262, RFC3264, RFC3362, RFC3515, RFC3550, RFC4028, RFC6341

Use cases

You can use SIP Tester for:
  • Performance and stability load testing of SIP servers and IP networks. Detection of memory leaks and CPU peak overlaods (gaps) with millions of calls.
  • Testing IVR servers, PBX, UC servers, call centers and other applications when upgrading to new version or moving to new SIP platform
  • Monitoring VoIP quality of live IP networks and servers, generating alerts and reports about system's performance: high jitter or packet loss, outage, reach of trunk's capaciuty limit
  • Troubleshooting call center setup issues when customers can't hear agents in some calls, resulting in dropped calls (one-way audio issues). Full 2-way audio path tests with audio verification to detect cases of bad audio quality
  • Testing of telephony billing systems: from simple tests to full 2-way audio tests
  • Testing and monitoring of SIP-GSM, SIP-PSTN gateways, SIP trunks
  • Showing demos of VoIP software and hardware with live audio connection between simulated SIP calls and laptop's sound card
  • Bulk SIP call generation
  • IVR Testing with audio verification
  • Penetration testing
  • IP PBX emulation
  • SIP DoS (flood) and brute force attack simulation and testing: view CallXML scripts for REGISTER and INVITE scanning
  • VoIP audio/voice quality measurement
  • 24x7 monitoring of RTP jitter and VoIP infrastructure availability (see script for active monitoring and sending alerts by email)
  • VoIP network diagnostics test, IP network assessment before implementing SIP services. Estimation of network's load capacity.
  • Checking QoS in the network by testing quality of calls with prioritized RTP packets (see Сhanging IP DSCP/TOS field)
  • Multichannel VoIP auto dialer: making calls to list of destinations with custom call processing
  • ED-137 performance and stability tests of mission-critical air traffic management (ATM) VoIP networks in airports
  • SIPREC SRS testing (SRC simulation): generate SIP calls with 2 RTP streams and play stereo WAV file
  • Debugging huge PCAP files, converting the PCAP files with VoIP traffic into CDRs and WAV files (G.711 and G.729 codecs are supported). It works faster than wireshark.
View diagrams for various use cases in context of different environment scenarios

Compatible software and hardware

Our customers reported successful results of testing following software and hardware

  • 2600Hz SIP server
  • 3CX Phone System 11.0.26995.0
  • Aastra MX-ONE TSE 3.2, 5.0
  • Addpac AP200D FXS-SIP gateway
  • Alcatel-Lucent OmniPCX Enterprise SIP Server 11.0.1
  • Asterisk PBX
  • Audiocodes Mediant gateways: 1xxx, 2xxx, 4xxx
  • AudioCodes MPxxx FXS/FXO SIP gateway
  • Avaya Communication Manager 6.0, 6.3
  • Avaya IP Office
  • Avaya Message Application Server (MAS)
  • Bria softphone
  • Brekeke SIP server
  • Broadvox/Fusion VoIP provider
  • Cirpack 4.56
  • Cisco Unified Communications Manager (CUCM) 8.6, 10.5
  • Cisco SIP Gateway (IOS v15.4.2)
  • Contivio SIP Server 3.3.02
  • Digitalk VPM
  • Eltex TAU-xx VoIP gateway
  • Excendia platform (SpeechMobility)
  • eXosip library
  • Fastwire OpenCA SS7-SIP signalling gateway
  • FreePBX (FPBX-2.8.1, FPBX-2.11.0)
  • FortiVoice PBX
  • FreeSWITCH
  • Genesys SIP server
  • Huawei eSpace 7820
  • Huawei SoftX3000 V300R010
  • Interactive Intelligence (ININ) Server 4.0005.0017.422
  • iTel Switch Plus softswitch platform
  • KolmiSoft MOR softswitch
  • LinkSys hardware SIP phone
  • MERA VoIP Transit Softswitch MVTS3G v.4.4.0-26b
  • Metaswitch SIP stack (DC-SIP 2.0)
  • miniSIPServer v17
  • Mitel 3300 Controller
  • NetSys NVP 4.0.2 FXS/FXO gateway
  • Nortel SESM
  • NovaAlert alerting system
  • OpenSIPS SIP server
  • OpenVox VoxStack Series GSM Gateway
  • Planet VIP-28x FXS-SIP gateway
  • Polycom RMX conference platform
  • Polygator G20 VoIP-GSM gateway
  • Quescom VoIP gateway
  • Quintum VoIP gateway
  • Saleem Telecom VoIP provider
  • Samsung OfficeServ PBX
  • Sangoma Vega 400G VoIP gateway
  • Shoretel ShoreGear voice switch
  • SI3000 Compact Call Server
  • SIP Express Router
  • gateway
  • Sippy B2BUA, softswitch
  • Sipwise NGCP SIP application server
  • Sipwise NGCP SIP proxy
  • StarTrinity softswitch
  • Telesis Stilllink ISDN-SIP gateway
  • Telviva PBX
  • VocalCom OnNet SIP server
  • VocalTec Essentra CX SS7-SIP signalling gateway
  • VOS softswitch VOS2009 V2.1.2.0
  • X-Lite SIP softphone
  • Yealink VP530 SIP phone
  • Zoiper SIP softphone

Detailed test reports can be found here

Our customers

StarTrinity SIP Tester has been purchased by more than 600 customers all over the world so far, and they are satisfied with the software and support we provide. Additionally, more than 600 customers have used the free version of SIP Tester. Our customers list includes:

BT Global Services Telefonica Digital LTD TELUS Portugal Telecom
Cisco VIDEOTRON KPN Telstra Corporation Limited
view testimonial

Airbus DS Communications Motorola Solutions, Inc Avaya Media5 Corporation SmarTone Telecommunications Netcall VoiceTrust GmbH

TELES Communication Systems GmbH Polycom Verint Systems - Contact Solutions Aspect Software Nuance Communications NEC UK LTD

NETSCOUT ZyXEL Communications

Cigna University of Houston Leidos Bloomberg LP Woodforest National Bank Merlin Entertainments Plc Insight Enterprises

Mitel Absolute Contacts GmbH Speech Mobility

FREQUENTIS E2M Technologies Sytel Tango Networks 8x8

Sangoma FaxLogic Next Generation Network ESTRACOM Systems

ConnexCS cloud softswitch provider 7G Network, a leading provider of voice services and broadband internet

If you are our customer and you don't want to have your logo displayed here, please contact us to remove your logo from the list

Our partners - individual VoIP troubleshooting specialists

We have partners who can help you to fix VoIP issues using our software. We work with following persons to troubleshoot various issues using the SIP Tester.
  • Gary Greenberg
  • Lisa Helmbacher. 20+ years experience in telecommunications industry, entrepreneur, small business owner, consultant and software developer and system integrator specializing in PSTN TDM (SS7, ISDN, MFR1, MFR2) and IP VOIP SIP products. Extensive experience in telecom testing and Cybersecurity, Toll Fraud and DDOS attack mitigation software applications.
  • Jim Hendry

Success stories

Most of customers test their SIP software, servers and network, and we don't know details. Here are details of using SIP Tester which have been shared to us

Ditto VR needed a robust SIP platform that could handle UDP, TCP and TLS+SRTP calls. Combining the StarTrinity SIP Tester with Asterisk, OpenSIPS and Cisco we were able switch call scenarios quickly. The CallXML language was critical for us when stress testing and as a result, we were able to dimension the footprint needed in AWS per 4000 concurrent calls.
Wavefront was recently commissioned to loadtest a client IVR platform and started researching tools that could provide SIP load with media support. I was reluctant to use a Windows based product since I knew I would have to integrate with a Linux based custom load generator for SMS along with a reporting tool. We came across SIPTester and quickly became comfortable with scripting in the CallXML language for creating complex inbound and outbound call scenarios. The SIPTester code is very efficient with a small memory/CPU footprint. Personally I never experienced a single crash, which was my biggest concern using a Windows based loadtest tool. Using the SIPTester command line mode and returned exit codes, we were able to integrate testing across platforms and tie in reporting tools using Windows batch scripts.

The bulk of my previous SIP load testing experience was with SIPp for Linux. Fortunately the open source SIPp project does not support media very well and I was forced to look for another tool. It was fortunate because if SIPp had supported media I may not have discovered SIPTester. We never came close to the limit of complexity of interactions that can be scripted with SIPTester. For example, SIPTester can listen to inbound media, compare the received audio with reference files and branch accordingly. This means two way conversations can be achieved very easily and the CallXML scripting language makes using these types of RTP aware features very intuitive. I experimented with these features but their use was out of scope of our project.

The list of features supported by SIPTester is very impressive but equally impressive is the comprehensive documentation available for each feature and the including examples. The documentation is freely available to study on the developer’s website and is constantly growing and improving as features are added. Our team found development of SIPTester CallXML scripts very easy. The tool itself is like an IDE in that it does real-time syntax checking, and call scenarios can be created via GUI or directly via CallXML scripts. The tool has detailed performance reporting based not just on signalling but also on RTP metrics such as levels, jitter and loss. The tool also support WAN emulation such as impairment generation (arbitrary packet loss etc.).

The value for such a powerful and mature SIP loadtest platform is extremely good and the way SIPTester can be evaluated in demo mode before purchasing, with all functionality enabled, makes it a risk free investment. The developer team at StarTrinity is very responsive to support and feature requests. I found the developers to be very knowledgably, professional and pleasant to work with.

I look forward to working with the StarTrinity team and products in the future and have been recommending the SIPTester product whenever appropriate.

Greg Toews, P.Eng
Manager, Engineering
Vancouver BC
Customer#51 in North America used SIP Tester to run VoIP tests in a satellite IP network. They have been facing some voice quality issues in the network and their vendor was unable to find solutions. With SIP Tester customer#51 assured that the satellite segment was working properly and demonstrated to vendor that the cause of the voice quality issues was be in vendor's side (Huawei IP Phones, Voice Core: SBC, Softswitch, Media Gateway, LAN switches etc). They aligned the teams and reviewed the procedures and best actions for a more effective analysis, diagnosis and troubleshooting. They used SIP Tester for the call tests using the satellite environment (RTT around 600 msec), adjusted RFC3261 T1 timer and RTP TX packet size to have a better picture of performance. SIP Tester was installed on multiple laptops and servers in both active and passive modes. For passive mode server with SIP Tester was connected to mirror port and collected performance of the live traffic. The customer was happy with quick support and releasing new versions to support their specific configurations. Based on measurements of SIP Tester, also with help of wireshark, customer discovered that there was network latency due to the satellite link plus the queuing, serialization and processing. Additionally, received RTP traffic sometimes started around 700ms and up to 1.2s (post-dial RTP delay). Conclusion was to review the configuration of the central site equipment to improve the voice quality and the total delay. For some calls SIP Tester discovered incorrect audio codec, it was solved with configuration of SIP phones. The worst SIP phones (with high packet loss) have been identified. Customer addressed every site to mitigate this problem. Overall traffic RFC3550 jitter was with average 25ms (from the remote phone, all the way through the satellite to the central site). Cases with high jitter impacted by the worst sites. The actions have been being taken to correct 31 sites wite the wrong codec and to correct some links with high packet loss. Additionally, packet loss was detected from the Huawei Core. The client verified Huawei LAN Switches, and discovered that:
  • switch statistics showed collisions and deferred packets
  • switch memory usage was 69% while it should have been 5-10 times smaller
  • switches only supported 100BaseT, all interfaces were in half duplex
They requested Huawei to replace the LAN switches with better equipment to operate at 1000BaseT and 0 packet loss, full duplex. Customer was pleased with realtime reports generated by SIP Tester:
"This kind of reports are not available in tools like Wireshark and Pilot. You must go step by step and take a lot of time to analyze and generate some statistical data. Your tool is allowing to create real time data/graphs that will help to speed up the data collection, data analysis, diagnostic, troubleshooting, and optimization."
Customer#25 used SIP Tester to simulate calls from Europe to few remote locations in Caribbean region. The calls were made through customer's softswitches, gateways and PSTN network between 2 instances of SIP Tester installed on both ends. SIP Tester was configured with custom CallXML scripts to access list of numbers from CSV file or MSSQL database, generate SIP call, make random delays and save successful and failed calls in CSV CDR files with custom format.
Customer#44 in North America used SIP Tester to test their VoIP recorder. SIP Tester was installed on 2 servers, connected via network switch. Customer's VoIP recorded was connected to mirroring port and stressed with SIP and RTP traffic generated between 2 instances of SIP Tester. SIP Tester simulated 200-800 concurrent G.711 SIP calls on i5 servers. Custom CallXML scripts were used to simulate non-standard SIP behaviour like call transfers (REFER) and call parking (re-INVITE).
Before SIP Tester: customer did not have enough information about bottlenecks and load capacity of their software. They tried to simulate high call load with Freeswitch, but it crashed.
After SIP Tester: customer optimized his code to achieve better performance. Additionaly, they discovered that with 400 concurrent calls few SIP and RTP packets become lost in spite of the fact that it was LAN environment with 1GBit ethernet. After some investigation with our help they discovered that packets were lost in NIC driver and in Windows 7 IP stack. Solution was to use Windows Server operating system and a better NIC.
Customer#7 in Eastern Asia used SIP Tester to check quality of SIP trunks provided by end suppliers. SIP Tester generated calls to list of destinations from CSV file and measured KPIs including post-dial delay and "-24dB delay" (delay of non-silent audio tones in RTP stream). The customer has successfully identified SIP trunks with long post-dial delay which reached up to 1 minute with small SIP answer (200 OK) delay.
Customer#35 in Australia used SIP Tester to continuously check availability of server and IP connectivity. SIP Tester generated test calls every 30 minutes and sent alert emails in case of call failure.
Customer#55 in Africa used SIP Tester to generate calls on daily schedule to list of destination numbers from CSV file. SIP Tester played a random (one of 10) WAV file to recipient if outgoing call was answered.
Customer#387 We are using this software to load test our phone systems. Our load test requirement was to simulate agents on the phones during a normal day. When our agents log in to the system, the phone system establishes a call to the agent’s phone and they are connected all day until they log out for lunch or leave for the day.


Sergey's SIP Tester is a really good product. I've done load tests with it up to half a million calls. It helped flush out some bugs in the deployment we did, and really saved us. Since then, he's also added in the ability to match .wav files against samples you provide the tool, and combined with the CallXML scripting language you can make a script that calls in, navigates a IVR (making sure that each wav file matches) and reach an agent. Some possibility's that I haven't completed yet that are really enticing as well, is it can branch the script on DTMF tones, so I could play a simple message to a agent such as "is the call quality good, press 1 for yes, 2 for no" and have it kick off a email on a DTMF of 2, combine that with the ability to schedule a kick off of this, and I could practically have this run against a deployment every hour as a back-check of their environment. I'm really very happy with the product.
Seriously is a very good product. I really recommend that you check out the demo, and consider getting a copy. Sergey is also pretty responsive to any emails I've shot him, and is pretty easy to work with. My biggest problem is finding the time to actually work on this stuff, and Sergey is never a hold back in that regard.
Someone in US

SIP Tester helped me a lot to qualify my switch and its configuration that the switch can do 1000 concurrent G.711 calls per port with no problems
Someone in Canada

We're using SIP Tester to load test Genesys and Avaya Communication Manager. SIP Tester is very intuitive and easy to use. I am impressed with how stable it is and the flexible CallXML functionality makes it possible for us to test a wide range of scenarios. The support we have received from Sergey is outstanding. I always get prompt replies on my emails and it is a pleasure to receive help and support directly from the guy who really knows about the inner workings of the software. I give Sergey and the software my best recommendations.
Jacob Thorbjorn, Denmark

We are very satisfied with StarTrinity SIP Tester and we will definitely recommend your product. In our case, we used it for testing the access of our Inbound Numbers and for emulating connection in our SIP Registrar.
Scott Bossler, Switzerland

I work for a VoIP and TDM service provider based in Portland, OR. A portion of my job is to test and certify IP PBX systems that customers want to use with the company's network. In addition, I assist the NOC with troubleshooting real world problems in a Lab environment. I use the StarTrinity SIP Tester in many ways. The first use is to set up a baseline verification for the VoIP switch in the Lab's Central Office. In other words, I set up the switch to communicate with a new PBX under test. To ensure the switch is configured properly, I use the StarTrinity SIP Tester. It's a known good configuration. I know when I connect up the PBX to the switch, any problems are 'in' the PBX because I confirmed all is OK in the switch. Second, we often need to simulate traffic to a SIP end point that responds with a certain message. For example, recently we needed to look at another provider's handling of a SIP 404 Not Found message. The StarTrinity SIP Tester was set up to respond to 100% of incoming calls with a 404 message. In the real world, a 404 message is something you troubleshoot and avoid. That's the beauty of the StarTrinity SIP Tester; it allows the user to specify call handling. Another use is to confirm QoS handling within and outside of the network. Bulk call generation permits pushing QoS to the maximum to see exactly what is happening to SIP signaling and the actual RTP packets. The real value of the StarTrinity SIP Tester is the intuitive GUI. I'm not a programmer and dislike CLI-based SIP test applications because of the extended learning curve. I don't have the time or ability to download and install four programs just to allow me to run some SIP CLI test. Something always fails to install and the mission is aborted. The StarTrinity SIP Tester application is running on a Windows XP notebook with no additional add-on programs needed! The StarTrinity SIP Tester is the every-man's test application! It's very easy to configure and to experiment with CallXML tweaks (to see what all that stuff means). A SIP Registration configuration and subsequent call processing can begin in about three minutes. It's that intuitive and powerful.
Ken Wells, Portland, OR, USA

We are using SIP Tester to check our system. We are developing a VoIP system and we need to confirm that the calls that we generated, they haven´t got problems about: Echo, noise and add two calls on the same SIP call (it sound terrible, I know it, but we had this problem one month ago). Your tool give us the posibility to check this and record audio at point as if it is final client. Our system must work 24 hours per day, so we leave SIP Tester working and check it every day.
Javier Estrella

Thank you very much for making the free SIP Tester tool! It is fantastic and very intuitive to use! I work for Acme Packet and have been searching for a simple SIP testing tool that allows me to load different call-type scenarios easily.
Mark Holloway

We like the simplicity of your product. We use the SIP Tester for:
  • Testing the bandwidth and quality of voice while under stress
  • Testing fax
  • Testing voip equipment
  • Testing new interconnects - for basic test we simply route calls to the tester.
Mitja Sosic

Your SIP Tester is really good piece of work! Thank you! I use it just when deploying big range of extensions or inter-connection between servers, extension on one, extension on second and check how it works, if passwords are okay, etc. But it’s simple and robust in same time.
Jan Pavlik

I need SIP Tester to make billing tests for interconnections with other operators. Usually, when i need to build new interconnect we must do the tests to confirm that both sides has the same call time in call records. For this - we make usually 5000 - 15000 calls with different call times and at the end - we compare the results. In analogue technology there are special generator, but for SIP - there is no such tool in Internet. Your tool is now very helpful for me.
Maciej Grędysa,

Your tool provides a lot of very useful features and helped us already to find bugs!

Here is what I like in SIP Tester:
  • Layout and clarity of documentation.
  • Ease of scripting thanks to documentation with clear examples.
  • Abstraction of SIP flow or ability to work with raw SIP messages if necessary.
  • Flatness of configuration/log/script/audio files.
  • Ease of upgrades, single click of link within application.
  • Light weight of application (Consumes about 150MB).
  • Performs well and is responsive, self contained.
  • Fantastic, dedicated and responsive support.
  • Value is very high. Freeware model allows for testing of all features before purchasing.
  • Ability to run on many different Windows varients (including virtual).
  • Knowledgeability of developer.
someone in North America

With StarTrinity SIP Tester, we were able to find that our QoS settings on our PBX weren't right and was causing packet loss on our LAN. We have addressed that last night and so far no dropped packets
someone in US

With the help of your application, we have identified a RFC non-compliance issue with our SBC! I want to thank you very much for the hard work!
someone in US

I really like the Star Trinity app, my engineers not so much as I have found a couple of hidden issues in our platform which they are now having to fix.
So win for me ;-)
someone in UK

Our customers like:
  • Easy to use for a non-programmer like me
  • User friendly and easy to configure and test
  • Easy interface and a lot of CallXML examples
  • Running on Windows, plain and simple
  • Straight forward settings in the gui / good functionality / (split) recording
  • Easy to setup and test
  • Love the easyness of GUI and audio quality measurements
  • Flexibility, ease of use with CSV files and DTMF ability
  • Ease of layout and details
  • Evaluation as freeware
  • A very feature rich application
customers in in US, Canada, UK, France

StarTrinity solutions help us to enhance our Engineering and Test Plan portfolio adding new cost effective capabilities. We could implement a Lab environment to test our VoIP designs simulating different load and stress tests conditions. StarTrinity arsenal of tools and flexibility, have helped us to achieve better tested VoIP solutions and accelerate the roll out in the field.
Mauro Pérez Santos, Mexico, NGN Estracom Systems, S.A. de C.V.

Cisco Enterprise Networking Business Unit is responsible for the whole Enterprise products portfolio including routing, switching, voice and wireless.
In a strategic research project there was a need, to simulate VoIP calls between clients and UCaaS (Unified Communication as a Service) across multiple IP networks and to measure quality parameters like Mean Opinion Score (MOS), round-trip time, jitter and packet loss. Traffic path was established through several LAN and SD-WAN networks including public internet, other wide area networks and two public clouds (AWS and Azure).
Network Address Translation (NAT) was used multiple times on the way from sender to receiver.
Standard test tools measure MOS score and other quality parameters like jitter based on simple UDP packets, which are sent without a real SIP emulation. Such measurements based on simple UDP streams and not on real SIP calls may give inaccurate results.
We have chosen SIPTester for the project because of almost unique ability to work in complex NAT environments and to simulate real VoIP calls. Some other tools were failing in our NAT test scenarios, while others provided unreliable data or simply were not working on public cloud.
StarTrinity SIPTester Software ran smoothly on-prem and in the public cloud. It was able to run tests during several days and to provide needed test data. StarTrinity Software Developers provided excellent support as we requested, to provide additional calculation formula for MOS score in order to compare results with some old similar tests from another departments.
Nikolai Pitaev
Senior Technical Marketing Engineer, Enterprise Networks, Cisco

What we have to improve

Our customers ask for improvements in SIP Tester. We are going to implement following new features in the SIP Tester or in a new product
  • Support of IPv6
  • Linux support
  • Not easy to edit SIP messages (not like in SIPP). Currently there are CallXML elements "sendsiprequest", "sendsipmessage", and there are parameters to insert custom SIP headers and SDP attributes. Also it is possible to access SIP trace in scripts. But the entire SIP call flow is not fully customizable.

We would like to thank all our current customers for purchasing the SIP Tester and encourage them to give more feedback. We need to know the details of your experience to make better decisions about our future development.

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