VoIP Engineer

? Overview

We are building a large-scale, highly available telephony platform that integrates with mobile and desktop applications, supporting both call center operations and AI-driven voice communication.

In this role, you will design, develop, and operate a redundant PBX system using Asterisk and Kamailio, as well as implement call control logic in Python.
You will be responsible for ensuring high reliability (SLA 99.99%+) and scalability (10,000+ concurrent calls) in a real-time communication environment.

? Responsibilities

1. PBX / SIP Infrastructure Design & Development

• Design, build, and operate PBX systems using Asterisk

• Implement SIP routing using Kamailio (load balancing, failover)

• Design high availability architectures (Active-Active / Active-Standby)

• Optimize RTP/media handling for large-scale traffic

• Ensure system scalability to support 10,000+ concurrent calls

2. Call Control Logic Development

• Develop call control APIs using Python (e.g., FastAPI)

• Integrate with Asterisk via ARI / AMI / AGI

• Design and implement call flows (IVR, recording, transfer, queue handling)

• Handle real-time call events (webhooks, event-driven architecture)

3. Client Integration (Mobile / Desktop / WebRTC)

• Integrate telephony features with mobile apps (iOS / Android)

• Integrate with desktop applications (e.g., Electron-based softphones)

• Design connectivity with SIP clients and WebRTC

• Improve call quality (latency, jitter, packet loss)

4. System Reliability & Operations

• Maintain SLA of 99.99%+ availability

• Monitor call quality metrics (MOS, jitter, packet loss)

• Implement logging and tracing per call/session

• Troubleshoot and resolve real-time system issues

• Perform capacity planning and scaling strategies

5. AI & Voice Integration (Nice to Have)

• Integrate AI-based voice systems (e.g., real-time AI conversations)

• Build pipelines for call recording, transcription, and analysis

? Requirements

Must Have

• Strong understanding of SIP / RTP / VoIP protocols

• Hands-on experience with Asterisk

• Experience with Kamailio or OpenSIPS for SIP routing

• Experience with Linux server environments

• Backend development experience in Python

• Solid understanding of networking (TCP/IP, NAT, firewall)

Nice to Have

• Experience with WebRTC

• Experience with ARI / AMI / AGI

• Experience with FreeSWITCH

• Cloud infrastructure experience (AWS / GCP)

• Experience with Docker / Kubernetes

• Experience with real-time / low-latency systems

• Experience in call center systems or telecom platforms

? Preferred Qualifications

• Experience designing systems for 10,000+ concurrent calls

• Experience maintaining high availability systems (99.99%+ SLA)

• Experience in distributed systems and event-driven architectures

• Experience with observability (metrics, tracing, logging)

• Experience working on AI-driven voice or conversational systems

? Tech Stack (Example)

• PBX: Asterisk

• SIP Proxy: Kamailio

• Backend: Python (FastAPI)

• Infrastructure: AWS / GCP

• Client: iOS / Android / Electron

• Protocols: SIP / RTP / WebRTC

? What We Offer

• Opportunity to design and build a large-scale telephony infrastructure from scratch

• Work on cutting-edge AI + voice communication systems

• Experience in high-load, real-time distributed systems

• Collaborative and global engineering environment

? System Scale & Requirements

• Concurrent Calls: 10,000+

• SLA: 99.99%+ availability

• Use Cases:

  • Call center systems

  • AI-driven voice communication

  • Mobile and desktop softphone integration