VoIP · SIP · RTP · WebRTC · AI Voice

Voice systems for production teams.

VoIP system architect and developer focused on SIP, RTP, FreeSWITCH, Asterisk, Kamailio, OpenSIPS, WebRTC, PBX platforms, SIP trunks, and AI voice automation.

I work across call routing, signaling, media behavior, softswitches, browser voice, carrier trunks, and debugging workflows that make voice systems understandable and reliable.

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Sip can happen

SIP routingRTP mediaFreeSWITCHAsteriskOpenSIPSKamailioWebRTCPBX platformsSIP trunksAI voice

Focus

Systems, signaling, media paths, and real-world voice debugging.

SIP and VoIP troubleshooting

Call-flow debugging across registration, routing, codec negotiation, rejected calls, NAT behavior, RTP paths, and one-way audio.

Softswitch and routing systems

Practical work around FreeSWITCH, Asterisk, OpenSIPS, Kamailio, gateway behavior, carrier failover, and dialplan logic.

WebRTC and AI voice experiments

Notes and implementation work around browser voice, SIP over WebSocket, realtime media, voice agents, and automation flows.

Projects

Recent builds, labs, and implementation notes.

OpenSIPS routing lab

Dispatcher, carrier failover, SIP routing decisions, and call-flow debugging notes.

WebRTC voice experiments

Browser media, SIP over WebSocket, ICE behavior, realtime voice, and AI voice paths.

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