Voice & Real-Time Media Platform Engineer Full Time NEW

Company Description TestCrew | Quality Engineering & Software Testing is a Saudi-born leader in Quality Engineering, Digital Assurance, and Digital Engineering, helping enterprises build and scale technology with confidence. With a team of 700+ experts across KSA, UAE, Jordan, Egypt, India, and Europe, the company delivers end-to-end solutions grounded in global best practices. TestCrew serves critical sectors including banking, government, telecom, aviation, retail, and SportsTech, supporting high-availability and complex digital environments. The company works on large-scale digital transformation, Quality Engineering Centers of Excellence, and cloud and DevOps modernization initiatives. Team members contribute to mission-critical programs for major ministries, regulators, banks, giga-projects, aviation operators, retail brands, and global enterprises.

Job Summary

We are seeking a highly skilled Voice & Real-Time Media Platform Engineer to design, build, and optimize real-time voice, audio, meeting automation, and AI-agent infrastructure. This is a specialized engineering role focused on low-latency media systems, telephony integrations, browser-based communications, and real-time AI interactions.

The ideal candidate will possess deep expertise in WebRTC, SIP/RTP, telephony gateways, audio processing, and distributed systems. You will play a critical role in developing scalable voice and media platforms, integrating with major meeting providers, and ensuring exceptional audio quality, reliability, and performance across production environments.

Key Responsibilities

* Design, develop, and optimize real-time voice, audio, and media systems with a focus on low latency and high reliability.

* Build and maintain infrastructure supporting AI-powered voice agents and real-time communications.

* Work extensively with WebRTC internals, including peer connections, media tracks, signaling, and browser media pipelines.

* Develop meeting automation solutions for platforms such as Google Meet, Microsoft Teams, and Zoom.

* Troubleshoot and resolve complex media-related issues across browsers, backend services, telephony systems, and real-time communication platforms.

* Integrate and manage telephony gateways and media servers, including SIP and RTP-based communications.

* Design and optimize low-latency Speech-to-Text (STT) and Text-to-Speech (TTS) pipelines with support for fallback mechanisms, barge-in functionality, and lip-sync synchronization.

* Develop and extend capabilities using the LiveKit Agents SDK and real-time agent infrastructure.

* Improve monitoring, observability, tracing, and diagnostics across voice and media processing pipelines.

* Lead incident response activities related to audio quality, latency, connectivity, media synchronization, and session reliability.

* Collaborate with product, AI, infrastructure, and backend teams to deliver scalable and resilient real-time communication solutions.

Required Qualifications

* Proven experience designing and operating low-latency, real-time communication systems in production environments.

* Deep understanding of WebRTC architecture, browser media handling, peer connections, and real-time media transmission.

* Strong experience with SIP, RTP, telephony integrations, media gateways, or voice communication platforms.

* Solid knowledge of audio processing concepts, including codecs, packet loss, jitter buffering, synchronization, and media quality optimization.

* Hands-on experience supporting production voice, audio, video, or real-time communication (RTC) platforms.

* Strong software development skills in Go, Java, C/C++, or Python.

* Experience troubleshooting distributed systems and leading incident response for critical production services.

* Strong understanding of networking protocols and performance optimization techniques for real-time media delivery.

Preferred Qualifications

* Experience with FreeSWITCH, baresip, SIP bridges, media gateways, or custom telephony integrations.

* Knowledge of C, cgo, native extensions, inter-process communication (IPC), and low-level media processing.

* Experience with Chrome DevTools Protocol, headless Chrome, browser automation, and browser-pool management.

* Ability to reverse-engineer or integrate with meeting platforms that provide limited public APIs.

* Hands-on experience with LiveKit and the LiveKit Agents SDK.

* Experience building custom STT/TTS integrations, barge-in capabilities, speech interruption handling, and lip-sync timing systems.

* Familiarity with multilingual speech processing and Arabic real-time NLP solutions.

* Experience implementing OpenTelemetry and distributed tracing across Python and TypeScript applications.

* Knowledge of Kubernetes performance tuning for browser automation, media processing, and real-time workloads.

* Familiarity with OpenFGA, Keycloak, identity management, and secure authorization models.

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Technical Environment

Real-Time Communications

* WebRTC

* SIP

* RTP

* LiveKit

* FreeSWITCH

* baresip

Voice & AI Technologies

* Speech-to-Text (STT)

* Text-to-Speech (TTS)

* Real-Time AI Agents

* Barge-In Handling

* Lip-Sync Processing

Programming Languages

* Go

* Java

* Python

* C/C++

Browser & Automation Technologies

* Chrome DevTools Protocol

* Headless Chrome

* Browser Automation

* Browser Pool Management

Infrastructure & Observability

* Kubernetes

* OpenTelemetry

* Distributed Tracing

* Monitoring & Incident Management

Security & Identity

* OpenFGA

* Keycloak

* OIDC

* Authentication & Authorization Frameworks