Voice & Real-Time Media Platform Engineer
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Key skills for this role
About the Role
TestCrew seeks a Voice & Real-Time Media Platform Engineer to design, build, and optimize real-time voice, audio, meeting automation, and AI-agent infrastructure. Requires deep expertise in WebRTC, SIP/RTP, telephony gateways, and backend languages like Go or Python.
Key Skills for This Role
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
Requirements
- 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
Full Job Posting
Company Description
- TestCrew | Quality Engineering & Software Testing is a Saudi born leader in Quality Engineering, Digital Assurance, and Digital Engineering.
- With a team of 700+ experts across KSA, UAE, Jordan, Egypt, India, and Europe, the company delivers end to end solutions.
- TestCrew serves critical sectors including banking, government, telecom, aviation, retail, and SportsTech.
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.
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 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
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