Cymetrix Software - SIP Developer - PBX System

Responsibilities :

- Design and implement telephony integrations using SIP and SIPREC.

- Develop APIs and backend services to handle call control, call recording, and session management.

- Work with PBX systems, SIP Servers, and Media Servers for SIP call flows and media capture.

- Integrate third-party VoIP systems with internal applications and platforms.

- Analyze and troubleshoot SIP signaling and RTP media flows.

- Collaborate with cross-functional teams including DevOps, Product, and QA to deliver scalable solutions.

- Create technical documentation, diagrams, and support material.

- Ensure systems are secure, resilient, and scalable.

Must-Have Skills :

- Practical experience with SIPREC for recording VoIP calls.

- Solid development skills in JavaScript (Node.js).

- Experience working with SIP Servers (e.g., FreeSWITCH, Asterisk, Kamailio, OpenSIPS).

- Hands-on knowledge of WebRTC, RTP streams, and VoIP media handling.

- Experience building and consuming RESTful APIs.

- Familiarity with call flows, SIP traces analysis (using Wireshark, sngrep, or similar).

- Strong understanding of networking basics (UDP, TCP, NAT traversal, STUN/TURN).

- Ability to troubleshoot and debug complex telephony and media issues.

Good to Have Skills :

- Knowledge of Call Recording Systems architecture and compliance standards (PCI-DSS, GDPR).

- Experience with Cloud Telephony Platforms (Twilio, Genesys Cloud, Amazon Chime SDK, etc.).

- Familiarity with Session Border Controllers (SBCs).

- Prior experience with SIP trunking and carrier integrations.

- Exposure to Protocol Buffers or gRPC for real-time messaging.

- Understanding of security practices in VoIP (TLS, SRTP, SIP over WebSockets).

- Knowledge of Docker and Kubernetes for deploying SIP services at scale.

- Sound knowledge of telecom protocols like SIP/ICE/STUN/TURN/SRTP/DTLS/H323/Diameter/Radius.

- Shall be thoroughly analytical and fix issues for SBC Portfolio of Products.

- Shall be thorough with Linux/RTOS internals and product Architecture is preferred.

- Strong Knowledge of TCP/UDP/IP and networking concepts is a must.

- Knowledge of IP telephony, SIP, Call Routing Techniques of ARS, AAR on Trunk config environment.

- Prior Experience on working with FreeSwitch, Kamailio & RTP Proxy, etc.

- Strong understanding of Audio streaming/websockets and their application in real-time communication systems.

- In-depth knowledge of audio codecs and their impact on voice quality and bandwidth utilization.

- Experience with gRPC and Protobuf for building efficient and scalable communication interfaces.

- Extensive experience in large scale product development in Enterprise, webRTC, VoIP, VoLTE based products.

Base Language/Framework :

- Frameworks/Tools : Express.js, Socket.io (for signaling if needed), Wireshark (for debugging), Sngrep.