Kamailio NAT Traversal Specialist

“We are looking for a VoIP / real-time media engineer with hands-on experience in Kamailio + RTPengine, RTP/Opus audio, and low-latency networking. This is not a UI or PBX admin role. Deep understanding of SIP/RTP media flow and NAT traversal is required 80% of components are open source available on github

Build Gateway v1 (device sip client Audio Tunnel → SIP Backend)
Objective
Build a cloud Gateway/Relay service that allows device sip client devices to send/receive real-time audio over TLS (TCP 443) and bridge that audio to a Kamailio + RTPengine SIP backend.
The device sip client must not use public SIP/RTP directly.

Scope (Gateway v1)
• Public interface:
◦ WebSocket Secure (WSS) on TCP 443
• Device side:
◦ Accept authenticated device sip client connections
◦ Receive Opus audio frames from device sip client
◦ Send downlink Opus audio frames to device sip client
• Call control:
◦ Handle START_CALL / HANGUP commands from device
• Media side:
◦ Exchange RTP (Opus) with RTPengine
◦ Bridge audio between WSS tunnel ⇄ RTP
• Backend:
◦ Trigger SIP calls via Kamailio
• No UI, no Android code

Must Use (Open Source)
• Kamailio (SIP routing)
• RTPengine (media anchoring)
• libopus
• Any open-source WebSocket server framework

Acceptance Criteria
• device sip client connects via WSS :443 from restricted networks
• Gateway receives uplink audio frames and sends downlink frames
• Gateway can place a SIP call via Kamailio
• Two-way audio works for 10+ minutes
• No SIP/RTP required on device sip client public network

Deliverables
• Gateway source code (Go/Node preferred)
• Build/run instructions
• Simple test log proving audio flow