Telephony Developer - SIP/RTP

Key Responsibilities

• Design and Development : Architect, develop, test, and maintain robust, scalable, and high-performance RTC systems and core voice platforms.
• System Ownership : Take full ownership of key layers, including SIP signaling, media handling (RTP), and session control.
• Optimization : Continuously monitor and optimize system performance, focusing on minimizing latency, maximizing call quality, and ensuring reliable service delivery across both cloud and on-premise deployments.
• Security & Reliability : Implement and enforce rigorous security measures for communication infrastructure and collaborate closely with Site Reliability Engineering (SRE) and Product teams to achieve carrier-grade reliability and availability.
• Collaboration : Work cross-functionally with QA, Product Managers, and other engineering teams to define requirements, troubleshoot issues, and deliver innovative communication features.
• Troubleshooting : Act as an escalation point for complex production issues related to telephony core components, ensuring rapid resolution.
  

Required Skills (Must-Have)

• Deep Expertise in SIP : Extensive hands-on experience with the Session Initiation Protocol (SIP), including message flow, various headers, error handling, and related protocols.
• FreeSWITCH Mastery : Expert-level knowledge and development experience with FreeSWITCH (or similar platforms like Asterisk), including module development, configuration, dialplan scripting, and application integration.
• Node.js Proficiency : Strong development skills in Node.js for building high-concurrency, event-driven back-end applications and microservices that interface with telephony core.
• Core Telephony Concepts : Solid understanding of telephony fundamentals, including codecs, QoS, NAT traversal, and security protocols (SRTP/TLS).
• Networking : Strong foundation in TCP/IP networking and common routing protocols relevant to real-time communications.
  

Highly Recommended Skills

• Kamailio/OpenSIPS : Proven experience with Kamailio (or OpenSIPS) for high-scale SIP proxying, load balancing, and routing.
• RTP Engine : Hands-on experience with RTP Engine (or similar media proxy/relay solutions) for efficient media handling and firewall traversal.
• Database : Experience with relational (e.g., PostgreSQL, MySQL) and/or NoSQL databases.
  

Nice To Have

• Additional Languages : Experience with Python and/or Go (Golang) for tooling, scripting, or service development.
• Cloud Infrastructure : Familiarity with deploying and managing telephony components on cloud platforms (AWS, Azure, or GCP).
• Containerization : Experience with Docker and Kubernetes.
  

Education And Experience

• Experience : 5 to 8 years of dedicated experience in developing and deploying core telephony or real-time communication systems.
• Education : Bachelors or Masters degree in Computer Science, Engineering, or a related field, or equivalent practical experience