Tata Communications
Website:
tatacommunications.com
Job details:
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 Communications
We are seeking a highly skilled
VoIP Architect with deep expertise in
SIP, WebRTC, RTP, and open-source telephony platforms. The ideal candidate will design, build, optimize, and troubleshoot large-scale real-time communication systems, while also contributing to and extending open-source VoIP stacks such as
FreeSWITCH and
Kamailio.
Key Responsibilities
Architecture & Design
- Design scalable, high-availability VoIP and real-time communication systems.
- Architect end-to-end call flows using SIP, WebRTC, RTP, and related protocols.
- Design and optimize media handling, NAT traversal, and signaling flows.
- Define best practices for low-latency, high-quality audio/video delivery.
Protocol Expertise
- Deep understanding and hands-on experience with:
- SIP (Session Initiation Protocol)
- WebRTC (ICE, STUN, TURN, DTLS-SRTP)
- RTP / RTCP / SRTP
- Analyze SIP traces and debug signaling issues (call setup, teardown, routing failures).
- Troubleshoot media issues such as:
- One-way audio
- Jitter, packet loss, latency
- Codec negotiation issues
Open Source Telephony Platforms
- Strong hands-on experience with:
- Kamailio / OpenSIPS (SIP proxy, routing logic, scalability)
- FreeSWITCH / Asterisk (media servers, call control)
- RTPengine / MediaProxy
- Customize and extend:
- FreeSWITCH modules (C/C++)
- Kamailio modules (C)
- Contribute to open-source or maintain internal forks.
Debugging & Troubleshooting
- Perform deep-dive debugging using:
- SIP logs (pcap, sngrep, Wireshark)
- RTP analysis tools
- Diagnose and resolve:
- Call drops and setup failures
- Media path issues
- Performance bottlenecks
- Root cause analysis (RCA) and preventive improvements.
Performance & Load Testing
- Design and execute load tests using SIPp.
- Simulate real-world traffic scenarios and stress conditions.
- Optimize system performance for high CPS (Calls Per Second) and concurrent calls.
Backend & API Development
- Develop high-performance backend services using Golang.
- Build and maintain REST APIs for telephony services and integrations.
- Integrate VoIP systems with external platforms (CRM, billing, analytics).
Data & Messaging Systems
- Work with databases and caching layers:
- Use message brokers:
- Design data models for call records, signaling data, and analytics.
DevOps & Infrastructure
- Deploy and manage VoIP systems in cloud/on-prem environments.
- Work with containers (Docker, Kubernetes preferred).
- Implement monitoring, alerting, and logging systems.
- Ensure high availability, failover, and disaster recovery.
Core Skills
Required Skills & Qualifications
- Strong expertise in SIP, RTP, WebRTC protocols.
- Deep experience with:
- Kamailio / OpenSIPS
- FreeSWITCH / Asterisk
- RTPengine
- Proficiency in debugging VoIP call flows and media issues.
Programming
- Strong experience in:
- C/C++ (for FreeSWITCH/Kamailio module development)
- Golang (backend services & APIs)
- Good understanding of networking (TCP/IP, UDP, NAT, firewalls).
Tools & Technologies
- SIPp (load testing)
- Wireshark, sngrep, tcpdump
- REST API design
- Linux system programming
Databases & Messaging
- MongoDB, MySQL
- Redis (caching/session handling)
- RabbitMQ (event-driven architecture)
Preferred Qualifications
- Experience with large-scale VoIP deployments (carrier-grade systems).
- Familiarity with telecom standards and interoperability.
- Experience with video (WebRTC SFU/MCU architectures).
- Contributions to open-source VoIP projects.
- Knowledge of security in VoIP (TLS, SRTP, fraud prevention).
Soft Skills
- Strong analytical and problem-solving skills.
- Ability to handle production incidents and critical debugging.
- Excellent communication and documentation skills.
- Ability to mentor engineers and lead technical initiatives.
Nice to Have
- Experience with Kubernetes-based VoIP deployments.
- Exposure to AI-driven voice systems or speech analytics.
- Experience integrating PSTN gateways / SIP trunk providers.
Click on Apply to know more.