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Real-Time Communication

Video CallingVoice ChatScreen SharingLive Streaming

Serviços de Desenvolvimento WebRTC

Aplicativos de Vídeo, Voz e Comunicação em Tempo 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.

Onde os projetos falham

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.

Conformidade

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.

O que construímos

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.

Nosso processo

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

Perguntas frequentes

Quanto custa o desenvolvimento WebRTC customizado?

Um recurso básico de videochamada 1:1 começa em torno de $14.000. Conferências em grupo com infraestrutura SFU, gravação e suporte multiplataforma custam $25K-$50K+. Os principais drivers de custo são a quantidade de participantes, requisitos de gravação e se você precisa de SDKs nativos mobile junto com o cliente web.

Devo usar um SDK de vídeo de terceiros ou construir WebRTC customizado?

SDKs de terceiros como Twilio ou Agora o levam ao mercado rapidamente, mas as taxas por minuto se acumulam rápido. Por volta de 50.000 minutos mensais, WebRTC customizado começa a se pagar. Frequentemente recomendamos começar com um SFU gerenciado como LiveKit para velocidade, depois migrar para infraestrutura auto-hospedada conforme o uso cresce.

Como você lida com NAT traversal e problemas de firewall?

Implantamos servidores TURN relay em múltiplas regiões geográficas usando Cloudflare ou Twilio Network Traversal. Isso mantém usuários atrás de firewalls corporativos e NATs simétricos conectados. Testamos contra configurações de proxy empresarial restritivas durante QA para detectar casos extremos antes do lançamento.

Os aplicativos WebRTC podem ser compatíveis com HIPAA?

Sim. WebRTC usa criptografia DTLS-SRTP por padrão, o que cobre o requisito de criptografia em trânsito. Para total conformidade HIPAA, adicionamos gravação no servidor com armazenamento criptografado, audit logging, controles de acesso e implantamos em infraestrutura elegível para BAA. Construímos plataformas de telemedicina que passaram em auditorias de segurança de terceiros.

Qual é o número máximo de participantes em uma chamada WebRTC?

Com arquitetura SFU, chamadas de vídeo em grupo suportam 50-100 participantes com vídeo ativo de forma confiável. Para audiências maiores, alternamos para um pipeline WebRTC-to-HLS — ingerindo via WebRTC para latência sub-segundo do transmissor, depois distribuindo via CDN para milhares de espectadores.

Quanto tempo leva para construir uma aplicação WebRTC?

Um recurso de videochamada 1:1 pronto para produção leva 4-5 semanas. Conferências em grupo com gravação, compartilhamento de tela e suporte mobile levam 7-10 semanas. Entregamos incrementalmente — sinalização e chamadas básicas primeiro, depois adicionamos gravação, analytics e processamento de mídia customizado.

WebRTC Development from $14,000

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

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