About The Company
Tata Communications Redefines Connectivity with Innovation and IntelligenceDriving the next level of intelligence powered by Cloud, Mobility, Internet of Things, Collaboration, Security, Media services and Network services, we at Tata Communications are envisaging a New World of CommunicationsWe are looking for a
highly experienced VoIP Developer and Subject Matter Expert (SME)
with deep hands-on expertise in
FreeSWITCH and Kamailio
. The ideal candidate will design, develop, debug, and optimize carrier-grade real-time communication platforms supporting
UCaaS and CPaaS
use cases, including
audio/video conferencing, WebRTC, and SIP-based systems
.This role requires strong systems-level programming skills in
Go, C, and C++
, deep understanding of
VoIP protocols
, and experience building
scalable, asynchronous, API-driven platforms
.Key ResponsibilitiesCore VoIP & RTC Development
- Design, develop, and maintain FreeSWITCH and Kamailio modules
- Debug and optimize call signaling, media flows, and RTP streams
- Implement advanced SIP routing, load balancing, NAT traversal, and failover
- Work on audio/video conferencing systems (mixing, recording, SFU/MCU concepts)
- Handle high concurrency and low-latency real-time traffic
Protocols & Media
- Deep expertise in SIP, SDP, RTP/RTCP, SRTP, DTLS
- Strong experience with WebRTC internals (ICE, STUN, TURN, DTLS-SRTP)
- Troubleshoot interoperability issues across softphones, browsers, SBCs, and carriers
- Optimize codecs (Opus, G.711, G.729, VP8/VP9, H.264)
Backend & Systems Programming
- Develop high-performance components in Go, C, and C++
- Implement asynchronous and event-driven programming models
- Build and integrate RESTful APIs for call control, provisioning, analytics, and billing
- Work with message queues, state machines, and distributed systems
UCaaS / CPaaS Platforms
- Design and enhance cloud-native UCaaS and CPaaS solutions
- Implement features like:
- Call routing and IVR
- Call recording and transcription
- Conferencing and collaboration
- Messaging and notifications
- Integrate with third-party carriers, SBCs, and CPaaS providers
Debugging & Performance
- Analyze logs, SIP traces, RTP dumps, and core dumps
- Use tools like sngrep, Wireshark, tcpdump, gdb
- Identify and resolve memory leaks, race conditions, and latency issues
- Tune systems for high availability, scalability, and fault tolerance
Required Skills & Qualifications
Technical Skills
- FreeSWITCH (core, modules, dialplan, ESL, mod_sofia, mod_conference)
- Kamailio (routing logic, modules, scripting, dispatcher, registrar)
- Programming Languages:
- Go (preferred for backend services)
- C / C++ (VoIP core and module development)
- Protocols: SIP, SDP, RTP, RTCP, SRTP, DTLS, WebRTC
- Networking: TCP/IP, UDP, NAT, firewalls, load balancers
- APIs: REST, JSON, async APIs, webhook-based systems
Platform & Infrastructure
- Linux system internals
- Docker / Kubernetes (nice to have)
- Cloud platforms (AWS, GCP, Azure)
- Monitoring and observability (Prometheus, Grafana, ELK)
Nice to Have
- Experience with Janus, Asterisk, OpenSIPS
- Experience building SFU/MCU architectures
- Familiarity with CI/CD pipelines
- Knowledge of telecom security and compliance
- Contributions to open-source VoIP projects
Soft Skills
- Strong troubleshooting and analytical skills
- Ability to act as a technical authority and mentor
- Excellent communication and documentation skills
- Experience working with globally distributed teams
