VoIP Engineer

We’re looking for a VoIP Engineer to help build and stabilize a SIP-based calling system.

Requirements:

• Strong experience with SIP, RTP, WebRTC
• SIP servers, call routing, NAT traversal
• Debugging call quality, latency, drops
• Experience with real-time voice systems

Nice to have:
Asterisk, FreeSWITCH, Kamailio, cloud deployments

Type: Contract or Full-time

Apply with: brief summary + relevant VoIP projects

Job Types: Full-time, Contract

Application Question(s):

• Have you created a SIP server before?
• Have you worked with Asterisk or Jambonz or LiveKit or similar tools before? Please specify.

Work Location: Remote