Senior Telephony / VoIP Engineer (Asterisk / FreeSWITCH)

Location : Remote / India (Nagpur preferred)

Experience : 4–8 years

Employment Type : Full-time

About the Role

We are building an advanced AI-driven voice automation platform, and our internal team

already has strong expertise in AI, LLMs, cloud engineering, and backend development.

The primary skill gap we want to fill is telephony infrastructure.

We are hiring an experienced Asterisk / FreeSWITCH Engineer who can design, deploy, and

maintain our VoIP and SIP-based telephony layer, and integrate it with our real-time AI backend.

If you have built production telephony systems and enjoy solving SIP / RTP / media challenges,

this role is for you.

Key Responsibilities

1. Telephony Infrastructure (Core Mission)

 Deploy, configure, and maintain Asterisk or FreeSWITCH servers in production.

 Set up and manage SIP trunks (Airtel, Jio, Tata, Exotel, Route, etc.).

 Design and implement dialplans, IVRs, call flows, inbound / outbound routing.

 Handle SIP signaling, SDP negotiation, and RTP media streams.

 Configure and optimize codecs, transcoding, jitter buffers, NAT traversal.

 Set up call recording, failover routes, call bridging, and basic CTI flows.

2. Integration with AI Backend (We Will Train You)

 Work with our engineering team to connect telephony with our AI voice gateway

(WebSocket / gRPC audio streaming).

 Forward RTP or PCM audio from Asterisk / FreeSWITCH to the AI engine.

 Send AI-generated audio back into call flows.

 No prior AI / LLM knowledge required — our internal team handles that layer.

3. Application Control & Automation

 Use ARI / AGI / EAGI (Asterisk) or ESL / FS API (FreeSWITCH) for programmatic call

control.

 Implement call events, state transitions, and real-time call orchestration.

 Collaborate with backend engineers on API-based workflows (CRM, booking, case

creation).

4. Performance, Monitoring & Reliability

 Ensure high availability, low-latency call paths, and call quality stability.

 Monitor SIP / RTP performance using tools like :

o sngrep, tcpdump, Homer / HEP, CDRs

 Troubleshoot call drops, one-way audio, codec mismatch, registration issues.

 Optimize system load, media handling, and concurrency scaling.

Required Skills & Experience

Telephony Expertise (Must-Have)

 4–8 years hands-on experience with Asterisk or FreeSWITCH in production.

 Strong understanding of :

o SIP, SDP, RTP

o Codecs (G.711, G.729, Opus)

o NAT traversal, STUN / TURN

o Dialplans, IVRs, bridging, DID routing

 Experience integrating SIP trunks with Indian carriers or CPaaS providers.

Technical Skills

 Scripting / programming familiarity with Python or Node.js.

 Strong Linux (Ubuntu / CentOS) knowledge.

 Ability to read SIP traces and debug issues quickly.

Nice-to-Have Skills (Not Required)

 Experience with CPaaS platforms : Twilio, Exotel, Route Mobile, Kaleyra, Plivo,

Knowlarity.

 Exposure to : o WebSocket / gRPC

o STT / TTS or conversational AI

o Kamailio / OpenSIPS

o Contact center systems (ACD, queues, CTI)

 Knowledge of GStreamer / FFmpeg and general audio processing.

(We will train you in AI, LLM, and real-time voice integration — not required upfront.)

What We Offer

 Opportunity to build one of India's most advanced agentic AI voice platforms.

 Work alongside a highly skilled AI / Cloud engineering team.

 Ownership of the entire VoIP / telephony layer.

 Exciting challenges in real-time audio, SIP, and next-gen voice automation.

 Flexible work environment and competitive compensation

Please include links to any relevant VoIP / GitHub projects or past telephony implementations.

Skills Required

Sdp, freeswitch, Node.js, Rtp, Ubuntu, Bridging, Linux, Sip, Centos, Asterisk, stun, Python