Telephony Developer - SIP/RTP

<p><p>We are seeking a highly passionate and skilled Core Telephony Developer to join our team in Pune.<br/><br/>The ideal candidate will be an immediate joiner eager to build and maintain cutting-edge Real-Time Communication (RTC) systems and voice platforms.<br/><br/>You will be instrumental in ensuring the performance, quality, and security of our carrier-grade communication infrastructure.</p><br/><p><b>Key Responsibilities : </b><br/><br/>- Design and Development : Architect, develop, test, and maintain robust, scalable, and high-performance RTC systems and core voice platforms.<br/><br/>- System Ownership : Take full ownership of key layers, including SIP signaling, media handling (RTP), and session control.<br/><br/>- 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.<br/><br/>- 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.<br/><br/>- Collaboration : Work cross-functionally with QA, Product Managers, and other engineering teams to define requirements, troubleshoot issues, and deliver innovative communication features.<br/><br/>- Troubleshooting : Act as an escalation point for complex production issues related to telephony core components, ensuring rapid resolution.</p><br/><p><b>Required Skills (Must-Have) : </b><br/><br/>- Deep Expertise in SIP : Extensive hands-on experience with the Session Initiation Protocol (SIP), including message flow, various headers, error handling, and related protocols.<br/><br/>- FreeSWITCH Mastery : Expert-level knowledge and development experience with FreeSWITCH (or similar platforms like Asterisk), including module development, configuration, dialplan scripting, and application integration.<br/><br/>- 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.<br/><br/>- Core Telephony Concepts : Solid understanding of telephony fundamentals, including codecs, QoS, NAT traversal, and security protocols (SRTP/TLS).<br/><br/>- Networking : Strong foundation in TCP/IP networking and common routing protocols relevant to real-time communications.</p><br/><p><b>Highly Recommended Skills : </b><br/><br/>- Kamailio/OpenSIPS : Proven experience with Kamailio (or OpenSIPS) for high-scale SIP proxying, load balancing, and routing.<br/><br/>- RTP Engine : Hands-on experience with RTP Engine (or similar media proxy/relay solutions) for efficient media handling and firewall traversal.<br/><br/>- Database : Experience with relational (e.g., PostgreSQL, MySQL) and/or NoSQL databases.</p><br/><p><b>Nice to Have : </b><br/><br/>- Additional Languages : Experience with Python and/or Go (Golang) for tooling, scripting, or service development.<br/><br/>- Cloud Infrastructure : Familiarity with deploying and managing telephony components on cloud platforms (AWS, Azure, or GCP).<br/><br/>- Containerization : Experience with Docker and Kubernetes.</p><br/><p><b>Education and Experience : </b><br/><br/>- Experience : 5 to 8 years of dedicated experience in developing and deploying core telephony or real-time communication systems.<br/><br/>- Education : Bachelors or Masters degree in Computer Science, Engineering, or a related field, or equivalent practical experience</p><br/></p> (ref:hirist.tech)