Powerful VoIP load testing tool to simulate SIP traffic, monitor quality, and stress-test telephony infrastructure
Powerful VoIP load testing tool to simulate SIP traffic, monitor quality, and stress-test telephony infrastructure
Vote (11 votes)
Program license Free
Developer StarTrinity
Version 3.1
Works under Windows
Vote
(11 votes)
Developer
StarTrinity
Works under
Windows
Program license
Free
Version
3.1
Pros
- Handles large-scale SIP call simulations with real RTP media
- Detailed real-time analytics and reporting
- Customizable CallXML scripting for diverse scenarios
- Audio recording and playback for quality verification
- 24x7 monitoring capabilities
- Wide codec and T.38 fax support
- Free for Windows users
Cons
- Interface may feel technical for beginners
- Requires understanding of SIP protocols and scripting for advanced use
- Focuses exclusively on SIP; does not support non-SIP protocols
A comprehensive VoIP load testing tool for SIP networks and equipment.
Feature-Rich VoIP Load Testing
SIP Tester is a dedicated utility for evaluating the performance and reliability of SIP-based VoIP infrastructures. Designed for Windows environments, this application caters to system administrators, VoIP developers, integrators, and anyone managing or deploying SIP servers, PBXs, IVR systems, or call centers. With its capability to simulate a large volume of simultaneous SIP calls, SIP Tester provides extensive stress testing and monitoring functionalities.
Advanced Simulation and Analysis
One of the application's primary strengths is its ability to generate thousands of concurrent inbound and outbound SIP calls, each with real RTP media streams. Through customizable CallXML scripts, users can craft detailed testing scenarios reflecting real-world usage and edge-case failures. This scripting flexibility allows for in-depth evaluation of SIP stack robustness, call handling patterns, and fault tolerance.
Another notable aspect is the inclusion of bulk call generation for intensive load testing as well as specialized simulations such as SIP DoS (Denial-of-Service) attacks. These stress tests are essential for determining system limits and uncovering vulnerabilities in SIP implementations.
Real-Time Monitoring and Reporting
SIP Tester continuously analyzes call quality, producing real-time statistics and historical reports. Key metrics such as RTP jitter, packet loss percentage, and answer delay are monitored, ensuring a thorough assessment of audio performance. The application also records audio streams (both mixed and separated RX/TX), enabling users to listen to call samples and evaluate overall voice fidelity.
The reporting module delivers clear CDRs (Call Detail Records) with call information and RTP performance data. Additional insights, like the "least quality calls" report, assist testers in quickly identifying problematic calls for further investigation.
Comprehensive Codec and Protocol Support
Compatibility with widely used audio codecs, including G.711, G.723, and G.729, ensures accurate testing across various deployment scenarios. For those needing fax transmission tests, T.38 fax protocol support is integrated, broadening the testing scope to include more communication modalities.
Continuous Infrastructure Monitoring
Beyond stress and compliance testing, SIP Tester offers tools for ongoing 24x7 monitoring of VoIP infrastructure and RTP jitter. This is especially valuable for IT teams aiming to maintain optimal service levels and to receive early warnings of developing issues.
User Experience and Usability
While SIP Tester delivers robust functionality, its user interface maintains a focus on practicality over aesthetics. Some familiarity with SIP concepts and scripting (CallXML) is recommended to unlock the application's full potential. However, with well-documented settings and script examples, the learning curve is manageable for professionals in the field.
Conclusion
SIP Tester proves itself as a powerful, versatile tool for anyone needing to assess, monitor, or validate SIP networks and devices. By combining real-time analysis, broad protocol support, and customizable testing scenarios, it serves as a valuable asset in the maintenance and optimization of modern VoIP environments.
Pros
- Handles large-scale SIP call simulations with real RTP media
- Detailed real-time analytics and reporting
- Customizable CallXML scripting for diverse scenarios
- Audio recording and playback for quality verification
- 24x7 monitoring capabilities
- Wide codec and T.38 fax support
- Free for Windows users
Cons
- Interface may feel technical for beginners
- Requires understanding of SIP protocols and scripting for advanced use
- Focuses exclusively on SIP; does not support non-SIP protocols