Freeswitch/Asterisk VOIP Consultant

Responsibilities

• Owner of Freeswitch VOIP Domain (System Design, Development, Technical Support, System Integrator)
• Develop high-performance and low latency systems for telecommunications
• Help the team build solutions e.g., call, chat, audio, video, etc
• Developing and debugging open source telephony technologies
• Identify and drive areas of improvement

Requirement

• Bachelor’s degree in Engineering (B.E./B.Tech or MCA) with 7+ years of experience
• 3+ years of experience in Freeswitch/Asterisk or other related VoIP technologies.
• Experience in building unified communication
• Familiarity with open source network tracing tools such as Wireshark/tshark and Ngrep.
• Experience with VoIP library for iOS, Android & Windows
• Designed/developed a number of VoIP, Telecom Solutions like IVR Systems, Autodialers, Conferencing Solutions, and Voice Mail solutions
• Hands-on experience in SIP, RTP, and RTCP.
• Knowledge of RTP Proxy and routed audio conferences
• SDP Protocol offer/answer.
• FreeSWITCH – Listening conference events using esl socket (i.e mod_event_socket).
• Worked on any tool for load testing of free switch for an audio conference.
• Experience in deploying and handling multiple instances of FreeSWITCH using a load balancer.
• Experience with telecom protocols like SIP, RTP, and SMPP.
• Open-source telephony technologies like FreeSWITCH, WebRTC, etc.
• Well-versed in SIP servers like SER/Open SER, and proxy servers.
• Knowledge of SBC, FreeSWITCH, and SIPX will be an additional advantage.
• Proficiency in operating in a pure Linux environment.
• Good knowledge of TURN, STUN, NAT, and TLS.
• Knowledge of H323
• Understanding of codecs like PCMU, PCMA, G729, Opus, etc
• Proficiency in operating in a pure Linux environment.
• Ability to understand basic to intermediate SQL.
• Familiarity with open source network tracing tools such as Wireshark/tshark and Ngrep.
• Excellent technical problem-solving skills.