SIP Developer

Jobskey Search and Selection; Bengaluru; Hybrid; INR 1,200,000 - 1,500,000; Full time

Job summary

An advanced technology recruitment firm is seeking a SIP Developer (Associate Architect) with over 6 years of experience. You will design, develop, and integrate telephony systems, focusing on VoIP communication and SIP signaling. Ideal candidates will have extensive knowledge in telephony integration and APIs, as well as strong analytical skills for troubleshooting SIP flows. This position is hybrid, requiring two days a week in the office, located in India, Karnataka, Bengaluru. Competitive salary and perks are available.

Qualifications

• Deep expertise in SIP and SIPREC essential.
• 6+ years of experience in telephony systems development.
• Strong knowledge of networking concepts and TCP/UDP/IP.

Responsibilities

• Design and implement telephony integrations using SIP.
• Develop APIs and backend services for call management.
• Collaborate across teams to ensure scalable solutions.

Skills

SIP

VoIP

Telephony Integration

APIs

VoIP Security Practices

Tools

Express.js

Wireshark

Socket.io

Job description

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.

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.
• Analytical skills for fixing issues in SBC Portfolio of Products.
• Familiarity with Linux/RTOS internals and product architecture.
• 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 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.

Frameworks/Tools

Express.js, Socket.io (for signaling if needed), Wireshark (for debugging), Sngrep.

Apply/Contact

WhatsApp: [phone]

Email: [e-mail]