Telephony Developer - SIP/RTP

We are seeking a highly passionate and skilled Core Telephony Developer to join our team in Pune.

The ideal candidate will be an immediate joiner eager to build and maintain cutting-edge Real-Time Communication (RTC) systems and voice platforms.

You will be instrumental in ensuring the performance, quality, and security of our carrier-grade communication infrastructure.

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