Technical Lead or Assistant Manager

Appealing Points:

• Opportunity to work on cutting-edge WebRTC architecture and scalable real-time audio/video conferencing solutions, including both P2P and server-based communication models.
• Hands-on exposure to advanced networking and media technologies such as SIP, RTP/RTCP, ICE, STUN/TURN, codec optimization, and media stream performance tuning in a global Agile environment.
• A high-impact technical leadership role involving cross-platform development (Web, iOS, Android, Desktop), team mentorship, and driving engineering best practices across the full software development lifecycle.

Annual salary : 8 to 10 Million yen

Job Responsibilities:

• 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.

Job 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).

Language: Japanese (Elementary level) and English (Fluent) is highly preferred.

About Company

The largest eCommerce company in Japan, and the third-largest eCommerce marketplace company worldwide. The organization provides a variety of consumer and business-focused services including e-commerce, e-reading, travel, banking, securities, credit card, e-money, portal and media, online marketing, and professional sports. The company is expanding globally and currently has operations throughout Asia, Western Europe, and the Americas.

[Measures against passive smoking]
No smoking indoors allowed
Designated smoking area

. Skillset Required: WebRTC, P2P Communication, Server-based Communication Models, SIP, RTP, RTCP, ICE, STUN, TURN, Codec Optimization, Media Stream Performance Tuning, Agile, Cross-platform Development, Team Mentorship, Software Development Lifecycle, Scalable WebRTC Architectures, Multi-party Audio/Video Conferencing, SFU, MCU, SDP, VP8, VP9, AMR, Opus, Bandwidth Estimation, Jitter Buffering, Product Management Collaboration, Quality Assurance, Software Engineering Best Practices, Code Reviews, Unit Testing, CI/CD, WebRTC APIs, getUserMedia, RTCPeerConnection, RTCDataChannel, TCP, UDP, IP Networking, DTLS, SRTP, Wireshark, Packet Analysis, Media Stream Debugging, Coding Standards, Source Control, Automated Testing, Collaborative Mindset, Agile Environments, VoIP, Telephony Systems, VoLTE, IMS, Janus, Mediasoup, Kurento, Jitsi, Machine Learning, Audio/Video Enhancement, Noise Reduction, Echo Cancellation, Signaling Servers, Node.js, Python, Go