Alberto Gonzalez Trastoy

About

Alberto is a Barcelona-born, Miami-based technologist who helps teams design, build, test, and operate real-time communications products. He holds a degree in Telecommunication Engineering from UPC and a Masters degree in Information Technology and Management from IIT. He is a consultant specializing in WebRTC, VoIP, IoT, voice AI, and low-latency audio, video, and data applications and currently works as CTO at WebRTC.ventures.

His telecom engineering background has led him to architect and deliver products for startups and enterprises, including video chat, telepresence remote control, speech analytics, broadcasting, and other real-time applications. Alberto brings hands-on architecture, implementation, testing, and technical leadership to teams that need dependable low-latency media and data systems.

Alberto has been sailing since he was 8 years old. He loves kiteboarding, skiing and, although he is not a very good dancer, he keeps trying.

Relevant Skills: WebRTC, VoIP, Voice AI, IoT, Low-Latency Audio/Video/Data, Software Architecture, Development, Tech Lead

Selected customer work

Real-time products shipped with public customer stories

A few public WebRTC.ventures stories show the type of client work Alberto helps lead: scalable telehealth, online education, and human-centered remote recording.

Sessions Health

Telehealth video infrastructure scaling for mental health professionals, including enterprise-grade Jitsi architecture on AWS.

Mathnasium

Custom online tutoring platform work with scalable video, collaborative workflows, pre-call checks, and role-based access.

StoryCorps

Remote interview application work for recording conversations from different locations with consent, checks, and cloud storage.

Voxbone / Bandwidth

Open source and demos

Top GitHub projects

Nine public repositories from Alberto's GitHub profile, ranked by public stars and focused on real-time media, WebRTC infrastructure, and voice AI experiments.

WebRTC Live Streaming with AI

One RTP video stream to many WebRTC clients with real-time image detection using Janus and OpenCV.

webrtc-agent-livekit

Quickstart template for real-time voice AI with LiveKit Agents, WebRTC, and local or third-party services.

reunitus

A quick prototype of a live video room using React and the Janus Media Server.

OpenVideoCall Web Agora

Multiparty video chat app using the Agora SDK with TensorFlow background removal.

free-whip

WebRTC broadcasting Python CLI using WHIP, the WebRTC HTTP ingestion protocol.

Janus WebRTC Server Container

A generic Janus WebRTC Media Server Docker container for repeatable deployments.

Voice AI with Pipecat

Demos for real-time AI-powered voice processing with Pipecat, OpenAI GPT, Deepseek, or Llama.

Kurento RPI Live Streaming

Kurento and Raspberry Pi WebRTC live-streaming experiments.

native-webrtc-peer-to-peer

A basic native WebRTC peer-to-peer web app implementation.

Active work and contributions

Current projects and open-source ecosystems

A focused view of active products and open-source projects connected to Alberto's current voice, WebRTC, and agent tooling work.

rtc-asr

Streaming speech-recognition lab for comparing ASR engines, latency, and real-time voice pipeline behavior.

CallAlternative

Voice-calling SaaS work around AI-assisted calling, telephony workflows, and customer communication.

PeerMetrics

Open-source WebRTC monitoring and quality tooling for diagnosing real-time communication sessions.

OpenClaw

Personal AI assistant runtime and automation ecosystem for agents, skills, tools, and workspace workflows.

Janus

WebRTC server and media infrastructure ecosystem behind many real-time communication deployments.

WebRTC.ventures

Case studies

Public stories of success from WebRTC.ventures across telehealth, edtech, contact centers, accessibility, broadcasting, and conferencing.

When Telehealth Grows Up

Video infrastructure transformation for Sessions Health using enterprise-grade Jitsi architecture on AWS.

Mathnasium's EdTech Platform

Custom remote tutoring platform with scalable WebRTC infrastructure and live learning workflows.

StoryCorps Remote Interview Application

Human-centered remote recording with video conferencing, consent forms, and cloud storage.

Advanced Call Center and Remote Video Interpreting

Contact-center and video-interpreting platform work for real-time customer support workflows.

WebRTC to SIP Gateway for ASL Interpreters

Real-time accessibility infrastructure connecting browser-based video to SIP systems.

Live Video Broadcasting and Webcast Control Room

Broadcasting and control-room tooling for live video production and managed viewing experiences.

Writing and decks

Real-time systems content

Recent blog posts and presentation decks on local speech recognition, voice AI, WebRTC architecture, SIP, and low-latency applications.

Medium

Real-Time Transcription Without a Powerful GPU

Local speech recognition and sub-200 ms partial transcript latency on everyday hardware.

vCons for Voice AI Records

SIP signaling, audit trails, disputes, and compliance evidence for voice AI teams.

Why WebRTC Matters for Real-Time Voice AI

Why low-latency voice AI needs real-time media transport, not just WebSockets.

Slideshare

Real-Time AI Voice Agents over WebRTC

A practical playbook for low-latency conversational voice systems.

Recipes for Real Time Voice AI

WebRTC, small language models, and open-source software patterns for voice AI.

Things I Wished I Knew Before Building My First WebRTC App

Practical lessons from architecture, media servers, networks, testing, and operations.

Talks and video

Conference and community sessions

Selected talks and short videos on architecture, voice AI, AWS, and AI-powered live streaming.

Geekle Official

Worldwide Software Architecture Summit 22 vol.2

Real-Time AI Voice Agents over WebRTC

WebRTC.ventures

Three Quick Thoughts from AWS reInvent 2025

Lessons Learned Building an AI Powered Live Streaming Camera

Location

Miami, FL

Email

albertogontras@gmail.com