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SipT: Revolutionizing Automated Call Testing for Call Centers

Date
SipT v1.0.0

We are thrilled to announce the official launch of SipT, a cutting-edge automatic call testing tool based on the SIP protocol, designed specifically for call center systems. SipT offers an innovative approach to improving system quality with ease, flexibility, and power that sets it apart from traditional testing tools.

SipT enables effortless simulation of extension device registration, outbound and inbound calls, and advanced call handling actions like transfers, holds, and hangups—all through simple HTTP API calls. Its intuitive design allows even users with no SIP protocol knowledge to conduct robust call testing with minimal setup, allowing focus on test scenarios rather than complex configurations.

Key features include:

  • Support for Multiple Protocols: SipT supports UDP, TCP, TLS, WSS, and WS, giving it broad versatility across different environments.
  • Customizable Call Behavior: With SipT, you can simulate custom audio input for outbound calls, set call duration limits, and define dynamic or pre-configured inbound call strategies, allowing for unmatched flexibility.
  • Industry-Leading WSS Simulation: SipT is the first in the industry to support batch WSS device simulation, ideal for high-concurrency environments.

One of SipT’s key advantages is its real-time control over call interactions. Unlike traditional tools like SIPp, where complex scenario files are required, SipT allows users to modify call behaviors on the fly, significantly reducing logical complexity while providing callback features to ensure precise test assertions.

Why Choose SipT Over SIPp?

  • Flexibility & Ease of Use: No more writing intricate scenario files—simply use HTTP API to trigger traffic functions. This API-first approach makes it easy to integrate into your existing test framework.
  • WSS/WS Support: SipT can simulate WebSocket devices, handling high-concurrency tasks with superior performance.
  • Real-Time Control: Adjust answering strategies and interact during the call process, something scenario-based tools like SIPp cannot offer.
  • Comprehensive Debugging: SipT offers detailed logging of SIP SDP, RTP, API calls, and runtime logs, making it easier to diagnose call issues and pinpoint failures.
  • Port Automation: Unlike SIPp, which requires manual port allocation, SipT automates port handling, reducing resource conflicts and improving call success rates.
$ curl -H 'Content-type:application/json' -d '[{"user":"1000","pwd":"temp123","scheme":"digest","realm":"client.com"}]' http://127.0.0.1:11111/account/add

$ curl -H 'Content-type:application/json' -d '[{"server":"sip:1000@8.8.8.8:6060","from":"1000","to":"1002","user":"1000"}]' http://127.0.0.1:11111/MakeCall

[{
    "call_id":"ebdf2673-d23b-418a-b2e9-c9f085b6818f",
    "from":"1000",
    "to":"1002"
}]

What’s more, SipT’s zero learning curve ensures that developers can use their preferred programming languages to control call logic without needing to understand the SIP protocol or learn new scripting languages.

Backed by real-world production experience, SipT has been rigorously tested across various scenarios, making it not just another testing tool, but a proven solution for improving the efficiency and accuracy of call center testing.

Experience the Future of Call Testing with SipT—available now!

$ ./SipT config/config.json