Driansh Softtech

QA & Testing

VoIP testing services for SIP and WebRTC platforms

We test call quality, signaling, load and failover on your VoIP platform, find what breaks under real conditions and report exactly how to fix it.
Talk to a VoIP QA engineerSee what we build

Free consultation with our engineers

What our VoIP testing covers

VoIP testing checks that your calls connect, sound clear and keep working as traffic grows. We test SIP signaling, RTP media and WebRTC clients, measuring jitter, latency, packet loss and MOS under realistic network conditions.

We test platforms built on FreeSWITCH, Asterisk, Kamailio, OpenSIPS and SIP.js, because we build with them too. You get repeatable test cases and a report that ranks issues by impact.

Technology we work with

  • SIPp
  • Wireshark
  • JMeter
  • sngrep
  • Homer
  • Playwright
  • SIP
  • RTP / SRTP
  • WebRTC
  • Python

What we offer

VoIP testing services we offer

Call quality testing

Measure MOS, jitter, latency and packet loss, so you know how calls really sound to users.

SIP protocol testing

Check INVITE, REGISTER, BYE and error flows, so calls set up and tear down correctly.

WebRTC testing

Test browser calls across devices and networks, including codec negotiation and ICE behaviour.

Load & stress testing

Push concurrent calls and call rates with SIPp, so you find limits before your customers do.

Interoperability testing

Verify calls between Asterisk, FreeSWITCH, OpenSIPS, carriers and softphones, so mixed setups work.

Failover & NAT testing

Simulate crashes, network loss and NAT limits, so you can confirm recovery works as designed.

Worried about call quality or capacity?

Tell us about your platform and traffic. A VoIP QA engineer will suggest a test plan and what to check first.
Talk to a VoIP QA engineer

What we build

Platforms we test

Multi-tenant IP PBX

Per-tenant call quality and capacity checks for hosted PBX.
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Call center software

Dialer, queue and recording tests under real agent load.
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Class 4 softswitch

High-volume routing, CPS and carrier failover tests.
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Class 5 softswitch

Feature and call-flow tests for retail VoIP services.
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Unified communications

Voice, video and chat tests across browser and mobile.
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Mobile softphones

Registration, push and audio tests for SIP apps.
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Why Driansh

Why choose Driansh for VoIP testing

Testers who build VoIP

Our QA engineers work next to our VoIP developers, so they know where platforms tend to fail.

Reusable test suites

We hand over SIPp scenarios and automation scripts, so you can rerun tests after every release.

Clear reports

Each issue comes with evidence, impact and a suggested fix, so your team knows what to do next.

Fixes on request

If you want, the same team can fix the issues we find instead of handing you a list.

How we work

From idea to live platform in four steps

  1. 01

    Discover

    We review your requirements, traffic and existing stack, then agree scope, milestones and cost.

  2. 02

    Design

    We plan the architecture, integrations and failover so the platform fits your growth plans.

  3. 03

    Build & test

    We develop in short sprints with regular demos, then load-test and QA before go-live.

  4. 04

    Deploy & support

    We launch on your servers or cloud and stay on for monitoring, fixes and upgrades.

Frequently asked questions

Get in touch

Worried about call quality or capacity?

Tell us about your platform and traffic. A VoIP QA engineer will suggest a test plan and what to check first.

Tell us about your project

Share a few details and the right engineer will get back to you.

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