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渋谷区, 日本
Onsite
Job Description:
About Organization
The LINK Department is a highly diverse and technologically advanced organization within Rakuten Symphony, Inc. Our team, comprised of members from over 10 countries, fosters a culture of open respect and creativity. Our core product, "Rakuten Link," is a comprehensive communication and ecosystem app for Rakuten Mobile customers, offering services such as calls, messaging, news, cash transfers, and point rewards. LINK Desktop further extends this communication experience to PCs and laptops.
We are seeking a skilled and experienced Technical Lead/SME (WebRTC) to join our dynamic team working on the Link application and lead the architecture, development, and optimization of our real-time communication platform. The ideal candidate will bridge the gap between traditional VoIP technology and modern web development, creating seamless peer-to-peer (P2P) or server-mediated (SFU/MCU) experiences. You will be responsible for defining technical roadmaps, mentoring engineering teams, and solving complex, low-latency streaming challenges.
Job Duties
Architecture & Design: Design and implement scalable WebRTC architectures for one-to-one and multi-party audio/video conferencing, supporting both P2P and server-mediated (SFU/MCU) models.
Protocol Implementation: Lead the expert-level implementation of core protocols, including SIP, SDP, RTP/RTCP, ICE, STUN, and TURN.
Media Optimization: Manage and optimize audio/video quality by tuning codecs (VP8, VP9, AMR, Opus) and handling media stream dynamics, such as bandwidth estimation and jitter buffering.
End-to-End Development: Collaborate with Product Managers, Tech Leads, and offshore/backend teams to translate requirements into robust, high-quality features.
Cross-Platform Integration: Build and maintain WebRTC integrations across web browsers, mobile (iOS/Android), and desktop applications.
Mentorship & Quality Assurance: Mentor engineering teams, promote best practices in software engineering (code reviews, unit testing, CI/CD), and foster a collaborative, knowledge-sharing environment.
Minimum Qualifications
Education: Bachelor’s degree in Computer Science or a related technical discipline.
Experience: 8–10 years of software development experience, including 3–5+ years specifically in WebRTC development.
Technical Expertise: Deep knowledge of WebRTC APIs (getUserMedia, RTCPeerConnection, RTCDataChannel) and network protocols (TCP/UDP, IP networking, DTLS, SRTP).
Debugging: Proficiency with Wireshark for packet analysis and media stream debugging.
Engineering Standards: Strong commitment to full SDLC best practices, including coding standards, source control, and automated testing.
Soft Skills: Collaborative mindset, openness to constructive feedback, and experience working in Agile environments.
Preferred Qualifications
End-to-end experience building commercial WebRTC-based applications.
Experience with VoIP/telephony systems (VoLTE, IMS).
Experience with open-source media servers (Janus, Mediasoup, Kurento, Jitsi).
Background in machine learning for audio/video enhancement (noise reduction, echo cancellation).
Experience with signaling servers (Node.js, Python, or Go).
Languages:
English (Overall - 4 - Fluent)