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Tier-1 employer and mid-level seniority increase competition, niche WebRTC/C++ requirements moderate density.
Highly domain-specific RTC, VoIP, and C++ media-server expertise limits cross-industry transferability.
Multiple mandatory technical filters: 5+ years, C/C++, WebRTC internals, production debugging, and RTC stacks.
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Design and implement scalable real-time audio/video communication systems using WebRTC, RTP/RTCP, and WebSocket signaling.
Develop and enhance cloud media server components including signaling, TURN routing, call control, and media processing with a focus on low latency and high throughput.
Debug and resolve production issues related to call failures, media negotiation, voice quality degradation, and perform root cause analysis for system reliability improvements.
5+ years of professional experience in software engineering.
Proficiency in C/C++ programming including multithreading, memory management, and low-level networking.
Hands-on experience with WebRTC internals, SIP/VoIP signaling, RTP/RTCP protocols, audio/video pipelines, and jitter/latency optimization.
Bachelor's degree in Computer Science, Electronics & Communication, Electrical Engineering, Information Technology, or related technical discipline.
Experienced in production environments working with open-source RTC stacks like LiveKit or Janus and skilled at debugging network and media issues using packet captures and RTP analysis.
Comfortable working on cloud-native infrastructure with knowledge of GCP, Docker, Kubernetes, and distributed high-availability systems.
Has strong expertise in real-time media systems and telecom-grade reliable, ultra-low latency real-time communication platforms.