• May 16, 2025

•

Must Have

• Design and implement telephony integrations using SIP and SIPREC.

• Practical experience with SIPREC for recording VoIP calls.

• Experience working with SIP Servers (e.g., FreeSWITCH, Asterisk, Kamailio, Opensips).

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

• Experience – 5+ years

Responsibility:

• Telephony Integration Developer with deep expertise in SIP (Session Initiation Protocol) and SIPREC (SIP Recording)

• 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

Requirements

• Strong experience with SIP protocol (INVITE, ACK, BYE, REGISTER, REFER, OPTIONS, etc.)

• Practical experience with SIPREC for recording VoIP calls.

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

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

• 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.

• 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

Benefits

• High-impact role

• Remote-first culture.

• Exposure to global clients and cutting-edge technologies.

• Career growth, ownership, and continuous learning opportunities.

Please do not apply if you don’t meet the must-have requirements.

Job Type: Full Time

Job Location: Remote

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