Lead VoIP & Telephony Engineer

About the role

A financial services institution is scaling its high-volume tele-sales and digital contact center, moving from legacy telephony to a proprietary in-house calling and predictive dialing platform with over 130 concurrent seats and 1,000+ simultaneous outbound channels. It is looking for an experienced Lead VoIP & Telephony Engineer to own the end-to-end design, development and deployment of the calling engine, from media server clustering and SIP load balancing to predictive pacing, WebRTC integration and banking/payment API bridges.

Key responsibilities

• Design, configure and scale a high-concurrency telephony cluster using FreeSWITCH (or Asterisk 20+ LTS with ARI) and Kamailio/OpenSIPS as SBC/SIP proxy
• Develop proprietary outbound dialing algorithms (predictive, progressive, preview) that optimise line ratios, reduce wait times and cap call abandonment
• Implement browser-based softphones with WebRTC (SIP.js) and unify agent session states
• Manage SIP trunking, media gateways, codec negotiation (G.711, Opus) and RTP routing with Pakistani telecom operators (PTCL, Jazz, Telenor)
• Build microservices and webhooks connecting the dialer to CRM workflows, agent dashboards and banking partner APIs
• Set up non-blocking call recording, trans-coding, local archival and compliance reporting under PTA, SECP and SBP rules
• Deploy Prometheus, Grafana and Homer/HEP dashboards for SIP jitter, packet loss, ASR and system load

Required profile

• 5+ years designing and administering high-throughput VoIP, SIP trunking and enterprise contact center architectures
• Solid understanding of Pakistani telecom topology, PRI/SIP trunks, CLI whitelisting and PTA requirements for corporate call centers

Required skills

• FreeSWITCH (mod_sofia, ESL) and/or Asterisk (PJSIP, ARI, AMI)
• Kamailio or OpenSIPS for carrier routing and SIP security
• WebRTC calling via SIP.js or Verto in React/TypeScript
• Go, Node.js or Python for telephony control daemons and APIs
• PostgreSQL and Redis