Voip Network Engineer

We are a leading financial services institution expanding our tele-sales and digital contact center operations, currently supporting over 130 concurrent seats and more than 1,000 simultaneous outbound channels. As we transition from legacy telephony systems to a proprietary, high-performance in-house calling and predictive dialing platform, we seek an experienced Lead VoIP & Telephony Engineer. This role involves full ownership of the design, development, and deployment of our next-generation calling engine, including core architecture elements such as media server clustering, SIP load balancing, predictive pacing algorithms, WebRTC integration, and third-party banking/payment API connectivity.

Key Responsibilities

Design and scale a high-concurrency telephony cluster using FreeSWITCH (or Asterisk 20+ LTS with ARI) alongside Kamailio or OpenSIPS as SIP proxies and session border controllers. Develop proprietary outbound dialing algorithms—including predictive, progressive, and preview modes—that optimize line utilization, minimize agent wait times, and control call abandonment rates. Implement and maintain browser-based softphones leveraging WebRTC (SIP.js) to replace desktop softphones and ensure unified agent session management.

Manage telecom interconnects and carrier routing by handling SIP trunking, media gateways, codec negotiation (G.711, Opus), and RTP traffic with Tier-1 Pakistani telecom operators such as PTCL, Jazz, and Telenor. Build microservices and webhook integrations to connect the dialer engine with internal CRM workflows, real-time agent dashboards, and banking partner transaction APIs.

Design automated, non-blocking call recording pipelines that handle transcoding, compression, local archival, and compliance reporting in line with PTA, SECP, and SBP regulations. Implement comprehensive system monitoring and reliability frameworks using telemetry and metrics dashboards (Prometheus, Grafana, Homer/HEP) to track SIP jitter, packet loss, Answer-Seizure Ratios (ASR), and overall system load in real time.

Required Qualifications

A minimum of 5 years of hands-on experience designing and managing high-throughput VoIP, SIP trunking, and enterprise contact center telephony architectures. Deep technical expertise with media engines such as FreeSWITCH (mod_sofia, Event Socket Library - ESL) and/or Asterisk (PJSIP, ARI, AMI). Proven ability to deploy and script Kamailio or OpenSIPS for carrier routing, failover, and SIP security.

Practical experience implementing WebRTC-based browser calling using SIP.js or Verto within modern frontend frameworks like React and TypeScript. Strong backend development skills in Go (Golang), Node.js, or Python for building telephony control daemons, pacing algorithms, and RESTful APIs. Proficiency with PostgreSQL for schema design and indexing of high-volume call detail records (CDRs), as well as Redis for in-memory state locking and real-time queue management.

Solid understanding of Pakistani telecom infrastructure, including PRI/SIP trunk configurations, CLI whitelisting, and compliance with PTA regulations relevant to corporate call centers.

Preferred Qualifications and Benefits

While not explicitly stated, candidates with experience working in financial services, FinTech, or InsurTech sectors will find this role aligned with their background. The position offers the opportunity to lead cutting-edge telephony platform development within a high-scale environment, contributing directly to the modernization of critical contact center infrastructure.

This full-time role is ideal for professionals seeking to apply their VoIP and telephony engineering expertise in a dynamic, growth-oriented financial institution focused on innovation and regulatory compliance.