Why you would fit well in our team: You have a strong background in WebRTC/VOIP related technologies and full-stack development experience with client-server architectures, micro-services, databases, cloud-based technologies, API design and more.
Essential Job Functions:
- Design/develop a wide-range of audio and video solutions;
- Work on software development teams, building and deploying full stack web applications;
- Utilize common software development practices such as version control, unit testing, and CI/CD;
- Focus on scale and reliability;
- Choose the appropriate technology based if needed;
- Ability to work independently on individual tasks but also work in a team.
- 5+ years of relevant Software Development experience;
- Experience in JsSIP and SIPJs;
- Linux Systems knowledge;
- Deep knowledge of networking protocols (IP, TCP, UDP, SIP, H323, RTP, RTCP, STUN, TURN);
- Knowledge of video codecs and protocols H264/H.265, VP8/VP9, WebRTC, RTMP, HLS, CMAF, and DASH;
- Familiarity with developing Video Conferencing solutions (SFU, MCU);
- Relational (MySQL, Postgres) and NoSQL (Redis, Mongo, DynamoDB) database technologies;
- Strong troubleshooting skills;
- Experience with Agile Development methodologies;
- Experience with version control systems like Git, SVN etc.
- Design Video and Audio solutions leveraging WebRTC standards;
- Cloud-based technologies such as AWS;
- Kubernetes, Docker, Packer, and Terraform;
- ffmpeg - including libavformat/libavcodec;
- Open-source technologies such as FreeSWITCH, Kamailio, OpenSIPs, Kurento and/or Mediasoup;
- Experience with DevOps practices.