1. Strong ability to troubleshoot and resolve complex SIP-related issues in a carrier-grade network.
2. Administration of Linux environments (operation, deployment, and monitoring).
3. In conjunction with senior engineering, you will work to create, implement, troubleshoot, and automate newly created services within a VoIP environment.
4. Working with a team, you will work to maintain the current infrastructure and execute a plan to make voice resources highly available and scalable.
5. Troubleshoot issues that have been escalated from other departments.
6. Work to prevent the recurring problem and implement automated detection methods upon completion.
7. Participate in an on-call rotation and conduct maintenance during the after-hours maintenance window.
1. Bachelor's degree or equivalent experience in Computer Science or a related field.
2. 2+ years of experience with SIP/RTP and other real-time communication protocols, such as WebRTC is a must.
3. Experience with open-source VoIP platforms, FreeSwitch, or Asterisk in a Linux environment is a must.
4. Experience with highly scalable SIP proxies (Kamailio and OpenSIPS) is a plus.
6. Programming language experience with C/C++, or Go.
7. Knowledge of configuration management tools such as Ansible, Puppet, or Chef.
8. Relational Databases (MySQL, Postgres).
9. End user-oriented.
10. Advanced knowledge of operating systems and networks.
11. Analytical and problem-solving skills.
12. Aptitude for learning new technology.
13. Deadline driven.
14. Good Attitude.