Senior VoIP Engineer

🎧 Senior VoIP Engineer Synergy Beam Solutions is at the forefront of IT and cybersecurity, currently building a state-of-the-art telecommunications platform designed for high-performance call centers worldwide. We are seeking a Senior VoIP Engineer to be a foundational member of our engineering team, driving the architecture, scaling, and reliability of our core SIP-based SaaS platform. Join us if you are passionate about tackling complex challenges and redefining communication and emergency systems globally. 🎯 Your Impact: You will lead the design and maintenance of mission-critical, real-time voice systems. Your focus will be on ensuring unparalleled reliability, scalability, and security while enhancing voice quality and observability across our global cloud infrastructure. Key Responsibilities: Architect & Optimize: Design, develop, and maintain high-performance SIP and RTP-based VoIP systems , serving as the subject matter expert on core VoIP protocols. Deep Troubleshooting: Utilize expert-level skills to diagnose and resolve complex routing, codec, and signaling issues using packet captures (Wireshark), logs, and SIP traces . Protocol & Network Tuning: Proactively optimize network performance across critical protocols ( SIP, RTP, TCP/IP, DNS, NAT, and firewall rules ). Ecosystem Integration: Integrate, support, and ensure seamless interoperability between PBX systems, SBCs, and modern cloud VoIP environments. Quality Assurance: Define, monitor, and aggressively improve call quality metrics ( MOS, jitter, latency ) through proactive performance tuning. Feature Leadership: Lead the technical architecture and deployment of advanced voice applications, including real-time transcription, AMD/VMD (Answering Machine Detection) , and next-generation AI-driven voice features. Escalation Point: Act as the final point of technical escalation for intricate VoIP incidents, effectively coordinating with internal teams and external vendors. ⚙️ Technical Requirements: Experience: 6+ years in software engineering, with a minimum of 4 years dedicated to VoIP/SIP development . Core Platforms: Deep, hands-on experience with core VoIP infrastructure components, specifically Freeswitch and Kamailio . Programming: Proficiency in scripting and development with Lua and Python . (Experience with Go or C/C++ is a significant plus). Networking Expertise: Expert-level command of network protocols and advanced debugging tools ( Wireshark, SIPp, libpcap, Linux/BSD sockets ). Cloud & WebRTC: Practical experience with WebRTC , configuring and managing TURN/STUN servers, and deploying solutions on major cloud platforms (AWS, Google Cloud, Cloudflare). Hardware Familiarity: Understanding of traditional telecom hardware, including GSM, E1 gateways, and serial-connected devices . Scalability: Proven track record in building highly scalable systems and implementing effective automated testing frameworks. Desired Skills (Nice-to-Have): Hands-on knowledge of call flow simulation tools and SIP testing frameworks. Experience designing and implementing RESTful APIs and scalable microservices. Familiarity with global network routing principles and optimization techniques. Experience with embedded systems or SIP hardware devices. Background in implementing AI/ML-driven features within VoIP solutions.