Posted:1 week ago| Platform:
On-site
Full Time
β Strong experience with SIP protocol (INVITE, ACK, BYE, REGISTER, REFER,OPTIONS, etc.) β Practical experience with SIPREC for recording VoIP calls. β Solid development skills in JavaScript (Node.js). β Experience working with SIP Servers (e.g., FreeSWITCH, Asterisk, Kamailio,OpenSIPS). β Hands-on knowledge of WebRTC, RTP streams, and VoIP media handling. β Experience building and consuming RESTful APIs. β Familiarity with call flows, SIP traces analysis (using Wireshark, sngrep, or similar). β Strong understanding of networking basics (UDP, TCP, NAT traversal, STUN/TURN). β Ability to troubleshoot and debug complex telephony and media issues Show more Show less
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