Jarvis Technology and Strategy Consulting
Website:
jarvis.consulting
Job details:
Position: Telephony Engineer
Department: Engineering – Voice / Communications Infrastructure
Experience: 3+ years
Employment Type: Full-time
Location: Noida
About Jarvis Technology & Strategy Consulting:
Jarvis Technology & Strategy Consulting is one of India’s leading political and policy consulting organizations, founded by socio-entrepreneurs from premier institutions such as IIT, ISB, and NIT. With a strong presence across the country, we leverage data and technology to drive large-scale political campaigns, design action-oriented community outreach programs, enable targeted communications, and improve the effectiveness of government initiatives.
About the Role:
We are looking for an experienced Telephony Engineer to design, build, and scale our voice
communications infrastructure. You will own the backbone that powers thousands of concurrent
calls daily, working on systems that demand high reliability, low latency, and a seamless user
experience. If you have built telephony platforms that just work — even under heavy load — we
would love to talk to you.
Key Responsibilities:
• Design, deploy, and maintain Asterisk-based telephony systems supporting 2000+
concurrent calls.
• Architect and optimize SIP-based calling flows, including inbound, outbound dialing
(OBD), and click-to-call features.
• Build and manage WebRTC-based voice solutions for browser and mobile clients.
• Develop and scale audio bridge and conferencing capabilities for multi-party calls.
• Troubleshoot call quality issues such as jitter, latency, packet loss, codec negotiation,
and NAT traversal.
• Integrate telephony systems with CRMs, dialers, and internal applications via APIs.
• Monitor system health, set up alerting, and ensure high availability across the voice
stack.
• Collaborate with backend, DevOps, and product teams to ship reliable voice features.
• Document architecture, runbooks, and post-incident learnings.
Required Skills & Qualifications:
Must-Have-
• 3+ years of hands-on experience as a Telephony / VoIP Engineer.
• Strong expertise in Asterisk — dialplan, AGI, AMI, ARI — with production-grade
deployments.
• Deep working knowledge of the SIP protocol — registration, INVITE flows, headers, and
debugging with sngrep or Wireshark.
• Solid experience with WebRTC — signaling, ICE/STUN/TURN, and media negotiation.
• Proven track record of building SIP calling, click-to-call, and outbound dialing (OBD)
systems.
• Experience designing and operating audio bridges and conferencing solutions at scale.
• Demonstrated ability to handle 2000+ concurrent calls in production.
• Experience with highly scalable, distributed infrastructure — load balancing, failover, and
horizontal scaling.
• Comfort with Linux, networking fundamentals (RTP, UDP, codecs like G.711, Opus), and
shell scripting.
Good-to-Have:
• Experience with Kamailio, OpenSIPS, or FreeSWITCH.
• Familiarity with cloud platforms (AWS, GCP) and containerization (Docker, Kubernetes).
• Knowledge of SBCs (Session Border Controllers) and carrier integrations.
• Scripting or programming in Python, Go, or Node.js.
• Experience with call recording, transcription pipelines, or IVR design.
• Exposure to telephony security — SIP TLS, SRTP, and fraud prevention.
What Success Looks Like:
• Voice infrastructure runs reliably at scale with minimal downtime and consistent call
quality.
• Click-to-call and OBD flows ship on time and integrate cleanly with product systems.
• Capacity planning and monitoring proactively catch issues before they reach customers.
• Knowledge is documented and shared so the team can operate the stack confidently.
Why Join Us:
• Own a mission-critical system from architecture to production.
• Work on problems that scale — real traffic, real impact.
• Collaborative team that values clean engineering and pragmatic solutions.
• Competitive compensation and growth opportunities.
How to Apply:
Interested candidates can share their updated resume, along with a brief note on relevant
projects — especially scale (concurrent calls handled), Asterisk/SIP/WebRTC work, and any
production systems they have owned.
Click on Apply to know more.