Freeswitch Engineer

2+ years experience with SIP/RTP and other real-time communication protocols such as WebRTC is a mustExperience with open source VoIP platforms, FreeSwitch, or Asterisk in a Linux environment is a mustStrong ability to troubleshoot and resolve complex SIP related issues in a carrier grade networkAdministration of Linux environments (operation, deployment, and monitoring)Experience with highly scalable SIP proxies (Kamaillo, OpenSIPS) a plus Scripting experience with Perl, Python, Erlang, JavaScript or PHPProgramming language experience with C/C++, or GoKnowledge of configuration management tools such as Ansible, Puppet, or Chef Relational Databases (MySQL, Postgres)In conjunction with senior engineering, you will work to create, implement, troubleshoot, and automate newly created services within a VoIP environment.Working with a team, you will work to maintain current infrastructure, as well as execute a plan to make voice resources highly available and scalableTroubleshoot issues that have been escalated from other departments. Upon completion, work to prevent the issue from recurring and implement automated detection methods.Participate in an on-call rotation and conduct maintenance during the after-hours maintenance windowDiploma or Bachelor degree in computer science, informatics or a related discipline.VoIP platforms-FreeSwitch experience.End user-oriented.Advanced knowledge of operating systems and network. Analytical and problem-solving skills.Aptitude for learning new technology. Deadline driven.Good Attitude.