Iñaki Baz Castillo

Passionate about new technologies, Open Source, WebRTC, modern Web development, Node.js, C++ and, above all, Real-Time Communications.

#webrtc #voip #nodejs #cplusplus #html5

ibc@aliax.net

Projects

mediasoup

core author

Cutting Edge WebRTC Video Conferencing.

rtp.js

main author

RTP stack for Node.js and browser written in TypeScript.

awaitqueue

main author

TypeScript utility to enqueue asynchronous tasks and run them sequentially one after another.

h264-profile-level-id

main author

TypeScript utility to process H264 profile-level-id values.

protoo

main author

Minimalist and extensible Node.js signaling framework for multi-party Real-Time Communication applications.

jssip

core author

The JavaScript SIP library.

libsdptransform

main author

Session Description Protocol C++ parser/writer.

cordova-plugin-iosrtc

main author - unmaintained project

Cordova iOS plugin exposing the full WebRTC W3C JavaScript API.

oversip

main author - unmaintained project

A powerful and flexible SIP proxy & server written in Ruby.

cpim / mimemessage / iscomposing

main author - unmaintained project

JavaScript libraries implementing CPIM, MIME and "iscomposing" IETF specifications.

em-udns

main author - unmaintained project

An async DNS resolver for EventMachine based on the udns C library.

Publications

RFC 7118

The WebSocket Protocol as a Transport for SIP

2014 • main author

❛❛The WebSocket protocol enables two-way real-time communication between clients and servers in web-based applications. This document specifies a WebSocket subprotocol as a reliable transport mechanism between Session Initiation Protocol (SIP) entities to enable use of SIP in web-oriented deployments.❜❜

ORTC

Object RTC API for WebRTC

2014 • co-author

❛❛This document defines a set of ECMAScript APIs in WebIDL to allow media to be sent and received from another browser or device implementing the appropriate set of real-time protocols. However, unlike the WebRTC 1.0 API, Object Realtime Communications (ORTC) API does not mandate a media signaling protocol or format.❜❜

draft-raymond-rtcweb-webrtc-js-obj-api-rationale

WebRTC JavaScript Object API Rationale

2013 • co-author

❛❛This document describes the reasons why a JavaScript Object Model approach is a far better solution than using SDP as a surface API for interfacing with WebRTC.❜❜

draft-ibc-websocket-dns-srv

DNS SRV Resource Records for the WebSocket Protocol

2011 • main author

❛❛This document specifies the usage of DNS SRV resource records by WebSocket clients when resolving a "ws:" or "wss:" URI. The DNS SRV mechanism confers load-balancing and failover capabilities for WebSocket service providers.❜❜

draft-sipdoc-rtcweb-open-wire-protocol

Open In-The-Wire Protocol for RTC-Web

2011 • co-author

❛❛RTC-Web clients communicate with a server in order to request or manage realtime communications with other users. This document exposes four hypothetical and different RTC-Web scenarios and analyzes the requirements of the in-the-wire protocol in each of them.❜❜

RFC 6455

The WebSocket Protocol

2011 • contributor

❛❛The WebSocket Protocol enables two-way communication between a client running untrusted code in a controlled environment to a remote host that has opted-in to communications from that code.❜❜

Experience

Miro 2022 - 2025

WebRTC Senior Software Engineer

A single, AI-powered collaboration platform that helps teams move faster from idea to outcome.

Around 2019 - 2022

WebRTC Senior Developer

Quick, high-impact video calls for a new era of work.

46 Labs LLC 2017 -2019

WebRTC Senior Developer

Leading the development of Sync: a WebRTC multiparty conference application focused on enterprises.

Self-Employed 2016 - 2017

WebRTC Freelancer

Working on my own for WebRTC related projects.

eFace2Face 2014 - 2016

WebRTC Senior Developer

Development of the eFace2Face WebRTC application. Backend (C++, Node.js) and frontend (desktop and mobile).

XtraTelecom 2008 - 2014

VoIP Senior Engineer

Responsible of the VoIP Department at XtraTelecom, a Spanish telco provider focused on enterprise market. SIP trunking services (Kamailio) and hosted PBX’s (Asterisk).

Ilimit Comunicacions 2007 - 2008

VoIP Senior Engineer

Deployment of a complete SIP provider based on OpenSer, FreeRadius, CDRTool, MySQL and SEMS. Linux Debian administrator.

irontec 2006 - 2007

VoIP Engineer

Residential/SOHO PBX solutions based on Asterisk. Linux (Debian, Ubuntu, RedHat) consultance services.

Talks

JanusCon

2019 / Naples

mediasoup: Like a Margherita Pizza for WebRTC

CommCon UK

2018 / London

Building multi-party video apps with mediasoup

voip2day

2017 / Madrid

WebRTC: Más allá de conferencias

Fosdem

2017 / Brussels

mediasoup: Powerful WebRTC SFU for Node.js

voip2day

2016 / Madrid

mediasoup: the programmable media server

ElastixWorld

2016 / Buenos Aires

mediasoup: Powerful WebRTC SFU for Node.js

voip2day

2015 / Madrid

WebRTC: Rock & Blood

GSICKMinds

2014 / A Coruña

WebRTC: do you even JavaScript?

voip2day

2013 / Madrid

Si sólo ves un softphone no entiendes WebRTC

ElastixWorld

2013 / México

Si sólo ves un softphone no entiendes WebRTC

4K Conference

2012 / Bogotá

SIP y WebRTC para Seres Humanos

SIMO

2012 / Madrid

World Wide SIP

SIMO

2011 / Madrid

SIP en la Web

SIMO

2009 / Madrid

Presente y Futuro de las Comunicaciones SIP

SIMO

2008 / Madrid

Asterisk & Carriers PSTN

SIMO

2007 / Madrid

Dialplan: el cerebro de Asterisk