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Tata Communications
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Manager - Customer Interaction Suite

Manager - Customer Interaction Suite

Name: Jobpe
Address: T-Hub, Plot No 1/C, Sy No 83/1, Raidurgam panmaktha, Knowledge City Rd, Hyderabad, Telangana, 500081, IN
Telephone: +91-83339-09630
Price range: Free

Tata Communications

10 - 13 years

0 Lacs

Ahmedabad

Posted:1 day ago| Platform:

Apply

Skills Required

debugging kubernetes rest systems programming restful load balancing message queues firewalls subject matter expert docker analytics restful apis gcp grafana use cases linux json prometheus aws programming azure

Work Mode

Work from Office

Job Type

Full Time

Job Description

We 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 Responsibilities
Core 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

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Tata Communications

Telecommunications

Chennai

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Manager - Customer Interaction Suite

Experience & Salary

Skills Required

Work Mode

Job Type

Job Description

highly experienced VoIP Developer and Subject Matter Expert (SME)

FreeSWITCH and Kamailio

UCaaS and CPaaS

audio/video conferencing, WebRTC, and SIP-based systems

Go, C, and C++

VoIP protocols

scalable, asynchronous, API-driven platforms

FreeSWITCH and Kamailio modules

call signaling, media flows, and RTP streams

SIP routing, load balancing, NAT traversal, and failover

audio/video conferencing systems

high concurrency and low-latency real-time traffic

SIP, SDP, RTP/RTCP, SRTP, DTLS

WebRTC internals

Go, C, and C++

asynchronous and event-driven programming models

RESTful APIs

cloud-native UCaaS and CPaaS solutions

sngrep, Wireshark, tcpdump, gdb

high availability, scalability, and fault tolerance

FreeSWITCH

Kamailio

Programming Languages:

Protocols:

Networking:

APIs:

Janus, Asterisk, OpenSIPS

SFU/MCU architectures

CI/CD pipelines

telecom security and compliance

technical authority and mentor

More Jobs at Tata Communications