Job Description

Note : We want immediate Joiner Only.

Experience : 2 - 5 Years

Location:

Mohali, Punjab

Work Mode:

On-site

Timings:

10:00 AM - 7:00 PM (Day Shift)

Interview Mode:

Face-to-Face ( On-Site)

Contact:

+91-9872993778 (Mon-Fri, 11 AM - 6 PM)

Key Responsibilities

Design, configure, and maintain

real-time voice systems

using

Asterisk

,

FreeSWITCH

, or similar PBX platforms. Develop and manage

SIP trunking

,

call routing

,

IVRs

,

auto-dialers

, and

call queue systems

. Integrate telephony systems with

Node.js

or

Next.js

-based microservices and third-party systems like

CRMs

or

Hospital Information Systems (HIS)

using

REST

and

WebSocket APIs

. Build

browser-based voice calling applications

using

WebRTC

and real-time signaling protocols. Implement features such as

call recording

,

call monitoring

,

analytics

, and ensure

Quality of Service (QoS)

including low latency and jitter control. Diagnose and resolve issues related to

SIP signaling

,

RTP streams

, and network behavior using tools like

Wireshark

,

sngrep

, and

SIP traces

. Deploy and manage

secure

,

scalable

, and

highly available

telephony systems using

Docker

,

Kubernetes

, and

cloud telephony providers

.

Must-Have Technical Skills

Strong understanding of

SIP

,

VoIP protocols

,

RTP/RTCP

, and

WebRTC

. Hands-on experience with

Asterisk

,

FreeSWITCH

, or other open-source telephony platforms. Programming experience in

Node.js

or

Python

for building telephony logic and backend APIs. Familiarity with

STUN/TURN servers

,

NAT traversal

, and use of

network debugging tools

such as

Wireshark

and

sngrep

. Knowledge of

telephony system scaling

,

clustering

,

failover mechanisms

, and

performance tuning

.

Nice-to-Have / Bonus Skills

Experience with

CPaaS platforms

such as

Twilio

,

Plivo

,

Telnyx

, or similar. Familiarity with

Kafka

,

RabbitMQ

, or other messaging systems for real-time event handling. Understanding of

VoIP security best practices

, including

TLS/SRTP

,

SIP authentication

, and

fraud prevention techniques

.

Education & Experience

Bachelor's or Master's degree in

Computer Science

,

Telecommunications

,

Engineering

, or a related field.

2 to 5 years

of hands-on experience working with

VoIP systems

,

real-time communication

, and

telephony infrastructure

.

Soft Skills

Strong

troubleshooting

and

analytical

skills. Ability to work independently and as part of

Agile/Scrum

teams. Good communication and documentation skills, especially for cross-functional collaboration.
Job Types: Full-time, Permanent

Pay: Up to ₹40,000.00 per month

Work Location: In person