Job Title: SIP Developer (Associate Architect)
Location: Mumbai/Bangalore - Hybrid (2 days in a week from office)
Only Immediate Joiners please!
Experience: 6+ Years
Job Description:
We are seeking a highly skilled Telephony Integration Developer with deep expertise in SIP (Session Initiation Protocol) and SIPREC (SIP Recording) to join our growing team. You will be responsible for designing, developing, and integrating telephony systems with a strong emphasis on VoIP communication, call recording, and SIP signaling.
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:
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.
Good to Have Skills:
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.
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:
Primary Language: JavaScript (Node.js backend)
Frameworks/Tools: Express.js, Socket.io (for signaling if needed), Wireshark (for debugging), Sngrep.
Apply/Contact:
WhatsApp: +91 7013663419
Email: [email protected] 