Voice & Real-Time Media Platform Engineer

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 SummaryWe 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.---Technical EnvironmentReal-Time Communications* WebRTC* SIP* RTP* LiveKit* FreeSWITCH* baresipVoice & 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 ManagementSecurity & Identity* OpenFGA* Keycloak* OIDC* Authentication & Authorization Frameworks