Senior Manual QA Engineer (VoIP / Voice / Real-Time Communications) at MMDSmart Ltd

, , Bosnia and Herzegovina -

Full Time

Start Date

Immediate

Expiry Date

24 May, 26

Salary

0.0

Posted On

23 Feb, 26

Experience

5 year(s) or above

Remote Job

Yes

Telecommute

Yes

Sponsor Visa

Skills

VoIP, Voice, Real-Time Communications, SIP Protocol, SIP/RTP Traffic Analysis, Call Center Systems, WebRTC, Voice Quality Metrics, TCP/IP Networking, Linux Environments, Scripting, REST API Testing, RTP Media Flow, Event-Driven Systems, Agile, Root Cause Analysis

Industry

IT Services and IT Consulting

Description

Would you like to build a career in a leading multinational IT company? We are looking for a QA engineer who behaves more like a voice system investigator than a checklist tester to join our team. Since 2007, MMDSmart has been a leading supplier in cloud communications, delivering game-changing IT products to businesses in over 100 countries. Our success is fueled by the inspiration and innovative ideas of our team members, encouraging their growth and impacting our Messaging and Voice solutions. These empower our clients to communicate more effectively with their customers. Don’t miss this chance to excel in a rapidly expanding tech company, where your potential and ambitions are not just recognized but celebrated. What will you do at MMDSmart? You will analyze business requirements and technical specifications for both web interfaces and backend, proactively challenging architectural assumptions to prevent issues before they reach production. You will act as a "Voice System Investigator" performing deep-dive analysis of complex call flows and media streams using packet captures (PCAPs) and server logs to identify the root cause of signaling and audio defects. You will design and execute advanced test strategies that cover the entire communication lifecycle, ensuring seamless integration between WebRTC components, IVR logic, ACD routing, and call recording systems. You will investigate the intersection of web application states and real-time media, debugging issues like UI/UX synchronization with agent states, call control race conditions, and WebRTC connectivity hurdles. You will proactively simulate network impairments (latency, jitter, packet loss) and use voice quality metrics (MOS) to ensure the system remains resilient and high-performing under sub-optimal conditions. You will drive quality improvements within an Agile environment, participating in architectural reviews and brainstorming sessions to advocate for testability and system stability. You will mentor the team and improve QA processes, sharing your expertise in troubleshooting stateful, event-driven systems and helping to refine estimation and testing methodologies. You will maintain technical documentation and RCA (Root Cause Analysis) reports, ensuring that complex "hard-to-reproduce" bugs are well-documented and that the team learns from every critical leak. You are a perfect match if you have: 5+ years of software testing experience, with at least 2+ years in VoIP / Voice / real-time communication systems Strong hands-on understanding of SIP protocol (call flows, registrations, INVITE flows, SDP, RTP) Practical experience analyzing SIP/RTP traffic Experience testing call center / CCaaS systems (IVR flows; ACD / call routing logic; Call recording; Agent states & call lifecycle) Familiarity with WebRTC technologies and real-time communication challenges Deep understanding of voice quality metrics (latency, jitter, packet loss, echo, MOS) Solid knowledge of TCP/IP networking, NAT, firewalls, ports, STUN/TURN Experience working in Linux environments and reading server logs Basic scripting skills in Python / JS / TS for test tooling and log analysis Experience with REST API testing (Postman, curl, or similar) Understanding of RTP media flow vs SIP signaling Ability to debug one-way audio, no audio, delayed media, early media Experience testing systems that depend on timing, network conditions, and concurrency Understanding how telephony behaves differently from web applications (stateful, event-driven, race conditions). Nice to have: Experience with test automation using Playwright or similar Experience with API automation frameworks (pytest, REST Assured, etc.) Familiarity with performance / load tools: (k6, JMeter, Locust) Experience simulating SIP traffic using SIPp Experience working with PBX / SBC / softswitches (Asterisk, FreeSWITCH, Zoiper, etc.) Experience testing under network impairment (packet loss / delay simulation) Why do talents choose and stay at MMDSmart? Embrace the opportunity to work in a company that cares about: Your Wellness: Access to mental health programs, comprehensive medical insurance, and a sports card*. Your Happiness: Enjoy the company of cool teammates, a friendly atmosphere, a strong mentorship, and a variety of bonuses & corporate gifts. Your Development: Benefit from free educational events & courses, and English classes with native speakers. Your Career: Focus on your professional development in a supportive environment. Work-Life Balance: Choose your working format - remote, office-based, or hybrid and engage in team-building activities. Think this role's got your name on it? Go ahead and apply, we'd love to see what you've got! Equal-Opportunity Employment We are an equal opportunity employer committed to creating a diverse and inclusive work environment. All qualified applicants will receive consideration for employment without regard to race, color, religion, gender, gender identity or expression, sexual orientation, national origin, genetics, disability, age, or veteran status. *Medical Insurance and Sport Card perks are available for employees located in Bulgaria.

Responsibilities

The engineer will act as a 'Voice System Investigator,' performing deep-dive analysis of complex call flows and media streams using packet captures and server logs to find the root cause of signaling and audio defects. They will also design advanced test strategies covering the entire communication lifecycle, ensuring seamless integration between WebRTC, IVR, ACD, and call recording systems.