VoIP Testing Services
VoIP testing is the process of evaluating and validating the performance, reliability, and audio quality of a Voice over IP (VoIP) communication system. Through VoIP testing, you can detect and resolve issues such as jitter, delay, packet loss, and poor audio quality before they affect your customers.
Our end-to-end VoIP testing solutions ensure every call is clear, consistent, and reliable, so you can deliver the best voip audio quality without delays or disruptions.
Why VoIP Testing Matters
Unclear voice, dropped calls, or poor WebRTC connectivity aren’t just technical glitches they hurt user experience, team productivity, and brand trust. In today’s real-time communication world, quality is non-negotiable.
At Inextrix, we go beyond basic voice tests. Our VoIP testing services proactively identify and eliminate performance bottlenecks across SIP and WebRTC environments ensuring your system is battle-ready.
What We Test
We evaluate your entire VoIP infrastructure from signaling to media using real-world simulation and advanced diagnostics:
VoIP & WebRTC Call Quality
Jitter
Fix distorted audio caused by uneven packet delivery
Latency
Minimize delays in transmission
Packet Loss
Eliminate audio gaps and voice cutouts
MOS Score
Quantify user-perceived call clarity
One-way Audio / Echo
Detect routing or codec mismatches
VoIP Bandwidth Testing
Validate capacity under load
Protocol-Level SIP & WebRTC Testing
IP Signaling (INVITE, REGISTER, BYE, etc.)
It will test the overall call flows and SIP behaviour.
RTP/Media Handling
Will analyze and validate the codec negotiation, media path, QoS
WebRTC Testing
It is the most important process where the proper evaluation is done based on browser-based calls using SIP.js, WebRTC libraries.
NAT Traversal & Firewall Handling:
It will detect all call drops due to network constraints.
Interoperability Tests
Asterisk ↔ FreeSWITCH ↔ OpenSIPS ↔ SIP.js
Failover Scenarios
Simulate service crashes, check recovery
Our VoIP Testing Process
We combine deep domain expertise with automated tools and manual validation to deliver reliable, actionable results:
Who Needs VoIP Quality Test & WebRTC Testing?
Our services are essential for any business where call quality, system reliability, and real-time communication directly impact customer experience and operations
VoIP & UCaaS Providers
Ensure your multi-tenant VoIP or UCaaS platform delivers HD voice and scales under load without jitter, packet loss, or signaling issues.
Call & Contact Centers
Maximize agent efficiency and customer satisfaction with consistent voice clarity, minimal call drops, and optimized routing.
Telecom Product Vendors
Test and fine-tune your telecom solutions - softphones, PBX, SIP gateways, or SBCs under various environments and edge cases.
Healthcare, Insurance, & BPOs
Maintain regulatory-grade voice reliability for critical conversations and ensure high uptime for remote agents or patient interactions.
IT & MSPs Managing Voice Platforms
Gain deep visibility into voice performance, proactively detect issues, and reduce support escalations across client systems.
Remote & Hybrid Teams
Guarantee seamless communication for distributed teams using browser-based VoIP/WebRTC tools regardless of their network conditions.
Why Choose Inextrix for VoIP Testing?
Inextrix stands as a trusted partner in delivering reliable, scalable, and secure VoIP solutions. Our VoIP Testing services are backed by years of telecom expertise, robust infrastructure testing capabilities, and real-world experience across global deployments.
Proven VoIP Expertise
With 15+ years in VoIP development and testing, our team understands the intricacies of SIP, RTP, WebRTC, and complex VoIP protocols across platforms like FreeSWITCH, Asterisk, OpenSIPS, Kamailio, and more.
End-to-End Testing Coverage
From SIP signaling and media stream analysis to NAT traversal, failover testing, and QoS monitoring, we cover every layer of your VoIP environment—ensuring complete readiness for production.
Real-World Simulation
We simulate real-world scenarios—packet loss, jitter, codec negotiation, security breaches, load stress, and device/browser compatibility—to validate performance under all conditions.
Custom Test Frameworks
Our team builds tailored test cases and automation frameworks using tools like SIPp, Wireshark, JMeter, and custom scripts, aligned to your infrastructure and goals.
Faster Time to Market
By detecting issues early and reducing back-and-forth QA cycles, our testing services help you launch faster with confidence and deliver crystal-clear calling experiences to your users.
Transparent Reporting
We deliver detailed, actionable reports with insights on bugs, performance metrics, and compliance gaps—so you can optimize and scale your VoIP systems efficiently.
Here's why industry leaders trust us
Inextrix stands as a trusted partner in delivering reliable, scalable, and secure VoIP solutions. Our VoIP Testing services are backed by years of telecom expertise, robust infrastructure testing capabilities, and real-world experience across global deployments.
A highly skilled team of VoIP testers, QA professionals, and telecom engineers with hands-on platform experience.
Our solutions have been successfully tested and deployed across global networks with varied latency and compliance needs.
We’ve worked with telecom operators, enterprises, healthcare, BPOs, and UCaaS providers delivering satisfaction at scale.
From startups to Tier-1 providers, we’ve tested complex VoIP and WebRTC systems under real-world stress and security scenarios.
Industries We Serve
Top reasons to choose us as your development partner for FreeSWITCH projects
Healthcare
Insurance
Education
E-commerce
Telecommunications
Finance
Real Estate
BPO
Our VoIP Services
Our commitment is to develop software that aligns with the specific requirements of our clients’ business.
Multi Tenant PBX
Tailor your telephony solutions with custom features that augment the communication experience of the participants with our custom SIP.js development services. .
Call Center
SIP.js experts will help you choose the right libraries and integrate them into your existing VoIP products and solutions to let you use the blended power of SIP and WebRTC.
Class 4 Softswitch
Our SIP.js consultants provide consulting services to help you push the limits of your existing manpower and resources and use SIP.js libraries at their full potential.
Class 5 Softswitch
Tailor your telephony solutions with custom features that augment the communication experience of the participants with our custom SIP.js development services. .
UCaas Platform
SIP.js experts will help you choose the right libraries and integrate them into your existing VoIP products and solutions to let you use the blended power of SIP and WebRTC.
Mobile Softphone
Our SIP.js consultants provide consulting services to help you push the limits of your existing manpower and resources and use SIP.js libraries at their full potential.
Our Case Studies
Conexo Technologies Leveraged Competitive Edge & Strengthened Communication With A Tailored UCC Platform
We have developed a completely tailormade UCC (Unified Communications and Collaboration) platform..
Custom Multi Tenant Call Center Software Developed In FreeSWITCH With PBX Features For A Solution Provider Company
Inextrix has delivered an all-inclusive platform to a communication software provider company.The company..
Frequently Asked Questions
VoIP testing is the process of evaluating how well your network supports Voice over Internet Protocol (VoIP) communication. It measures key parameters like bandwidth, latency, jitter, and packet loss to ensure high-quality, reliable voice calls.
Several tools are available to measure and analyze VoIP performance, including PingPlotter, Wireshark, and online VoIP speed tests (Nextiva, RingCentral, Vonage).
‘Jitter’ refers to the variation in the timing of data packet delivery. High jitter causes choppy, distorted, or delayed audio during calls. Ideally, jitter should be below 30 ms to maintain smooth and clear communication.
VoIP quality is often measured by Mean Opinion Score (MOS), which ranges from 1 (bad) to 5 (excellent).
3.5 to 4.4 MOS = Acceptable to good quality; above 4.0 MOS = Clear, reliable voice experience. An MOS closer to 5 is rare but indicates excellent VoIP performance.
VoIP latency can be tested used using: ping, traceroute, or VoIP testing tools. Ideally, it should be under 150 ms.
VoIP quality can be affected by bandwidth, latency, jitter, packet loss, network congestion, and router/ISP issues.
What Our Client Say
"We were looking for a reliable IP PBX solution as our conventional system had a lot of limitations. I’m happy with the solution and service provided by Inextrix. They gave us scalable IP PBX software and also integrated our CRM system and automated billing solution. This system increased productivity and ROI significantly."
Vice President, 8com inc
"Amazing group of guys to work with. After the quality of work I saw in the first project completed, we have decided to move all our development over to Inextrix. All work was completed on time and as expected. Would not think twice to recommend these guys."
Platformity
"Hi, I just wanted to send you guys a quick message on how great its been working with you guys on our voip development needs, in the last 3-4 years, you have been very responsive and always care about delivering the work as much as we do. Im very impressed with the high level of talent that you guys have and am grateful that I could find such dependable voip professionals that always do what they say they will. take care -Albert"
CleverMoto
