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.
Tasks
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.
Requirements
⚙️ 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.