Voice & Real-Time Media Platform Engineer

Company Description

Job Summary

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.

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++

• Chrome DevTools Protocol

• Headless Chrome
• Browser Automation

Infrastructure & Observability

• Kubernetes
• OpenTelemetry
• Distributed Tracing

Security & Identity

• OpenFGA
• Keycloak
• OIDC