Aram M.

I design and deliver production-grade real-time communication systems including video conferencing, live streaming, VoIP, and messaging platforms that must perform reliably at scale. With over 15 years of experience in WebRTC and telecom technologies, I have built solutions ranging from one-to-one calling apps to large multiuser collaboration and broadcasting systems across mobile, web, and backend infrastructure. What I can help you build or fix: ✔ Low-latency audio and video platforms using WebRTC, LiveKit, Janus, or custom stacks ✔ Video conferencing and virtual collaboration applications ✔ VoIP and SIP integrations ✔ Live streaming and broadcasting systems ✔ Real-time messaging platforms and backends ✔ Cross-platform mobile apps for iOS and Android ✔ Debugging complex media issues such as quality problems, device failures, sync issues, and scaling challenges Technical expertise: • Protocols: WebRTC, SIP, RTP, SRTP, RTSP, RTMP, XMPP, HLS • Networking: UDP, TCP, IP, NAT traversal, media routing • Platforms: iOS, Android, Web, Node.js, Go, Python backends • Infrastructure: scalable real-time architectures • Performance optimization and reliability engineering • Twilio, Telnyx, DeepGram, ElevenLabs I have helped startups launch communication products from scratch and assisted established teams in stabilizing systems already running in production. If your project involves real-time audio, video, or messaging, there is a high chance I have solved a similar challenge before. Let’s build a system that works flawlessly in production, not just in demos.