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 and optimize real-time voice, audio, and AI-agent infrastructure. Requires deep experience with WebRTC, SIP/RTP, telephony gateways, and backend languages like Go or Python.
Key Skills for This Role
Responsibilities
- Design and optimize real time voice, audio, and media systems
- Work with WebRTC internals: peer connections, media tracks, in page plumbing
- Build meeting automation for Google Meet, Microsoft Teams, and Zoom
- Debug real time media issues across browsers, backends, and telephony bridges
- Work with telephony gateways (i.e. FreeSWITCH) and SIP (baresip)
- Build low latency STT/TTS pipelines with fallback, barge in, and lipsync handling
- Work with the LiveKit Agents SDK and real time agent infrastructure
- Improve observability and tracing across media pipelines
- Lead incident response for audio quality, latency, and session reliability
Requirements
- Deep experience with low latency, real time systems
- Strong WebRTC internals and browser media handling
- SIP, RTP, telephony bridges, or media gateway experience
- Solid grasp of audio pipelines, codecs, jitter, packet loss, and synchronization
- Production experience with voice, audio, video, or RTC systems
- Strong backend skills in Go, Java, C/C++, or Python
- Distributed systems debugging and incident response
Full Job Posting
Job Summary
- Build and evolve real time voice, audio, meeting automation, and AI agent infrastructure.
- Highly specialized role focused on low latency media systems and meeting platform integrations.
Responsibilities
- Design and optimize real time voice, audio, and media systems
- Work with WebRTC internals: peer connections, media tracks, in page plumbing
- Build meeting automation for Google Meet, Microsoft Teams, and Zoom
- Debug real time media issues across browsers, backends, and telephony bridges
- Work with telephony gateways (i.e. FreeSWITCH) and SIP (baresip)
- Build low latency STT/TTS pipelines with fallback, barge in, and lipsync handling
- Work with the LiveKit Agents SDK and real time agent infrastructure
- Improve observability and tracing across media pipelines
- Lead incident response for audio quality, latency, and session reliability
Required Skills
- Deep experience with low latency, real time systems
- Strong WebRTC internals and browser media handling
- SIP, RTP, telephony bridges, or media gateway experience
- Solid grasp of audio pipelines, codecs, jitter, packet loss, and synchronization
- Production experience with voice, audio, video, or RTC systems
- Strong backend skills in Go, Java, C/C++, or Python
- Distributed systems debugging and incident response
Preferred Skills
- FreeSWITCH, baresip, SIP bridges, custom media modules
- C/cgo, native extensions, IPC for real time media
- Chrome DevTools Protocol, headless Chrome, browser pool management
- Reverse engineering meeting platforms with limited APIs
- LiveKit Agents SDK
- Custom STT/TTS plugins, barge in, lipsync timing
- Multilingual or Arabic real time NLP
- OpenTelemetry across Python and TypeScript
- Kubernetes tuning for browser automation and media workloads
- OpenFGA, Keycloak, secure authorization modeling
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