SIP Developer
On-siteBengaluru, Karnataka, India
Job Summary
Design and implement telephony integrations using SIP and SIPREC, developing APIs and backend services to handle call control, recording, and session management. Work with PBX systems, SIP Servers, and Media Servers to manage call flows and media capture while integrating third-party VoIP systems with internal platforms. Analyze and troubleshoot SIP signaling and RTP media flows, collaborating with DevOps, Product, and QA to deliver scalable, secure, and resilient solutions. Create technical documentation and support material for the SBC portfolio. This role requires 6+ years of experience in JavaScript (Node.js) and SIP protocol implementation. Immediate joiners only.
Required Qualifications
- 6+ Years
- Strong experience with SIP protocol (INVITE, ACK, BYE, REGISTER, REFER, OPTIONS, etc.)
- 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
- Strong Knowledge of TCP/UDP/IP and networking concepts
- 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
- Extensive experience in large scale product development in Enterprise, webRTC, VoIP, VoLTE based products
- Primary Language: JavaScript (Node.js backend)
- Frameworks/Tools: Express.js, Socket.io (for signaling if needed), Wireshark (for debugging), Sngrep
- Must be able to fix issues for SBC Portfolio of Products
- Shall be thorough with Linux/RTOS internals and product Architecture
- Sound knowledge of telecom protocols like SIP/ICE/STUN/TURN/SRTP/DTLS/H323/Diameter/Radius
Desired Qualifications
- Experience with Media Servers (e.g., Janus, Kurento, Mediasoup)
- 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
- Shall be thoroughly analytical
- Experience with gRPC and Protobuf for building efficient and scalable communication interfaces
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