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Real-Time Communication

Video CallingVoice ChatScreen SharingLive Streaming

Servicios de Desarrollo WebRTC

Aplicaciones de Video, Voz y Comunicación en Tiempo Real

<200ms

End-to-End Latency

Global edge network

99.9%

Uptime SLA

Production reliability

1,000+

Concurrent Streams

Tested at scale

Vendor Lock-In

You own the stack

What Is WebRTC Development?

WebRTC (Web Real-Time Communication) is browser-native and mobile technology that lets devices exchange video, voice, and data peer-to-peer without any plugins. A real WebRTC engagement covers the full stack — STUN/TURN server configuration, Selective Forwarding Unit (SFU) architecture, front-end media controls, recording, and adaptive bitrate streaming — and ships production-ready real-time communication features, not a proof of concept.

Dónde fallan los proyectos

Off-the-shelf video SDKs charge per-minute fees that explode at scale Monthly costs can hit $50K+ once you pass a few thousand daily active users.

WebRTC works fine in the browser until NAT traversal breaks behind corporate firewalls Without proper TURN infrastructure, 20-30% of enterprise users run into connection failures — and they blame your product, not their IT department.

Media quality falls apart under packet loss if you haven't built adaptive bitrate logic Users blame your app for choppy video — not their network — and they churn fast.

Group calls need SFU architecture, not simple peer-to-peer Peer mesh topologies fall apart past 4 participants, which makes your app unusable for teams.

Recording and compliance are afterthoughts in most WebRTC tutorials Healthcare and finance clients require server-side recording — and retrofitting it later costs far more than building it right the first time.

Cross-browser and mobile inconsistencies cause silent failures Safari and iOS handle getUserMedia differently, which leads to blank screens and a flood of support tickets.

Cumplimiento

End-to-End Encryption

DTLS-SRTP encryption is built into every stream by default. For industries that need zero-trust architectures, we add optional application-layer encryption on top of that.

HIPAA-Ready Infrastructure

Server-side recording with encrypted storage and audit logging for telehealth applications. BAA-compatible deployment on compliant cloud infrastructure.

Adaptive Bitrate Streaming

Simulcast and SVC encoding adjust video quality in real-time based on bandwidth, CPU, and network conditions. Every user gets the best quality their connection can actually support.

Server-Side Recording

Composite or individual track recording with webhook notifications on completion. Recordings go straight into your S3 bucket with configurable retention policies.

Connection Analytics

Real-time dashboards surface call quality metrics — jitter, packet loss, round-trip time, and resolution. You'll catch connection issues before users ever think to report them.

Firewall Traversal

Globally distributed TURN servers keep connectivity alive behind corporate NATs and restrictive firewalls. We test against enterprise proxy configurations during QA so edge cases don't surface in production.

Qué construimos

1:1 and Group Video Calls

SFU-based architecture supporting up to 100 participants with dynamic layout switching, dominant speaker detection, and bandwidth-aware quality scaling.

Screen Sharing & Co-Browsing

Full-screen and application-window sharing with annotation overlays and remote cursor tracking for collaborative workflows.

Voice-Only Channels

Low-bandwidth voice rooms with noise suppression, echo cancellation, and spatial audio support for gaming and social apps.

Live Streaming (WebRTC to HLS)

Sub-second latency broadcasting from WebRTC ingestion to HLS/DASH output for audiences of 10,000+ concurrent viewers.

Chat & Data Channels

WebRTC DataChannels handle in-call text chat, file transfer, and real-time data sync — no separate WebSocket server needed.

Custom Media Pipelines

Background blur, virtual backgrounds, real-time transcription, and AI-powered noise cancellation integrated at the media track level.

Nuestro proceso

Architecture & Protocol Design

We map your use case to the right topology — peer-to-peer, SFU, or MCU. We define the signaling protocol, TURN strategy, and recording requirements before writing a line of code. Deliverable: a technical architecture document.

Week 1

Signaling & Media Server Setup

We stand up your signaling server (WebSocket or HTTP), configure MediaSoup or LiveKit SFU, and deploy TURN/STUN infrastructure across edge regions.

Weeks 2-3

Client SDK & UI Development

We build the front-end media controls — camera/mic selection, layout switching, screen share, and in-call chat. Everything gets cross-browser tested on Chrome, Firefox, Safari, and mobile.

Weeks 3-5

Load Testing & Network Simulation

We simulate 500+ concurrent sessions with packet loss, jitter, and bandwidth throttling. Then we tune adaptive bitrate, reconnection logic, and failover recovery paths until they hold.

Week 6

Launch & Monitoring

We deploy to production with call quality dashboards, error alerting, and 30-day post-launch support. SRTP metrics and TURN utilization stay monitored so nothing quietly degrades at scale.

Week 7+

WebRTCNext.jsNode.jsSocket.ioMediaSoupLivekitSupabaseVercelCloudflare TURN

Preguntas frecuentes

¿Cuánto cuesta el desarrollo personalizado de WebRTC?

Una funcionalidad básica de videollamada 1:1 comienza alrededor de $14,000. Las conferencias grupales con infraestructura SFU, grabación y soporte multiplataforma cuestan $25K-$50K+. Los principales impulsores de costos son la cantidad de participantes, los requisitos de grabación y si necesitas SDKs móviles nativos junto al cliente web.

¿Debo usar un SDK de video de terceros o construir WebRTC personalizado?

Los SDKs de terceros como Twilio o Agora te permiten llegar al mercado rápido, pero sus tarifas por minuto se acumulan rápidamente. Alrededor de 50,000 minutos mensuales, el WebRTC personalizado comienza a pagarse por sí solo. A menudo recomendamos comenzar con un SFU administrado como LiveKit para velocidad, luego migrar a infraestructura auto-hospedada a medida que crece el uso.

¿Cómo manejas la traversía NAT y problemas de firewall?

Desplegamos servidores de retransmisión TURN en múltiples regiones geográficas usando Cloudflare o Twilio Network Traversal. Esto mantiene a los usuarios detrás de firewalls corporativos y NATs simétricos conectados. Probamos contra configuraciones restrictivas de proxies empresariales durante QA para detectar casos límite antes del lanzamiento.

¿Pueden las aplicaciones WebRTC cumplir con HIPAA?

Sí. WebRTC usa encriptación DTLS-SRTP por defecto, lo que cubre el requisito de encriptación en tránsito. Para cumplimiento total de HIPAA, agregamos grabación del lado del servidor con almacenamiento encriptado, registro de auditoría, controles de acceso e implementamos en infraestructura elegible para BAA. Hemos construido plataformas de telesalud que han pasado auditorías de seguridad de terceros.

¿Cuál es el número máximo de participantes en una llamada WebRTC?

Con una arquitectura SFU, las videollamadas grupales soportan de forma confiable 50-100 participantes de video activos. Para audiencias más grandes, cambiamos a un pipeline de WebRTC a HLS — ingiriendo vía WebRTC para latencia sub-segundo desde el difusor, luego distribuyendo vía CDN a miles de espectadores.

¿Cuánto tiempo lleva construir una aplicación WebRTC?

Una funcionalidad de videollamada 1:1 lista para producción toma 4-5 semanas. Las conferencias grupales con grabación, compartir pantalla y soporte móvil toman 7-10 semanas. Entregamos de forma incremental — señalización y llamadas básicas primero, luego agregamos grabación, análisis y procesamiento de medios personalizado.

WebRTC Development from $14,000

Fixed-fee. 30-day post-launch support included.

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