Skip to content
Now accepting Q2 projects — limited slots available. Get started →
Deutsch 한국어 中文 繁體中文 Espanol Francais 日本語 Nederlands Portugues العربية English
Real-Time Communication
Video CallingVoice ChatScreen SharingLive Streaming

WebRTC開発サービス

ビデオ、音声、リアルタイム通信アプリ

<200ms
End-to-End Latency
Global edge network
99.9%
Uptime SLA
Production reliability
1,000+
Concurrent Streams
Tested at scale
0
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.

プロジェクトが失敗する理由

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.

コンプライアンス

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.

構築する内容

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.

私たちのプロセス

01

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
02

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
03

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
04

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
05

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

よくある質問

カスタムWebRTC開発にはいくらかかりますか?

基本的な1対1ビデオ通話機能は約14,000ドルから始まります。SFUインフラストラクチャ、録画、クロスプラットフォーム対応を備えたグループ会議は25,000~50,000ドル以上です。主なコスト要因は参加者数、録画要件、Webクライアントに加えてネイティブモバイルSDKが必要かどうかです。

サードパーティのビデオSDKを使用すべきか、カスタムWebRTCを構築すべきか?

TwilioやAgoraなどのサードパーティSDKを使用すれば素早い市場展開が可能ですが、分単位の料金が増加します。月間50,000分程度で、カスタムWebRTCが採算が取れるようになります。多くの場合、速度のためにLiveKitなどのマネージドSFUから始めることをお勧めし、使用量が増えるにつれて自ホストインフラストラクチャに移行します。

NATトラバーサルとファイアウォールの問題にはどう対処しますか?

CloudflareまたはTwilio Network Traversalを使用して複数の地理的地域にTURNリレーサーバーをデプロイしています。これにより、企業ファイアウォールと対称NATの背後にいるユーザーを接続状態に保ちます。QA中に制限的なエンタープライズプロキシ設定に対してテストし、起動前にエッジケースをキャッチします。

WebRTCアプリはHIPAA準拠にできますか?

WebRTCはデフォルトでDTLS-SRTP暗号化を使用しており、転送中の暗号化要件をカバーしています。完全なHIPAA準拠のために、暗号化されたストレージ、監査ログ、アクセス制御を備えたサーバー側記録を追加し、BAA適格インフラストラクチャにデプロイします。第三者セキュリティ監査に合格した遠隔医療プラットフォームを構築した実績があります。

WebRTC通話の最大参加者数は?

SFUアーキテクチャでは、グループビデオ通話は50~100のアクティブビデオ参加者を確実にサポートします。より大きなオーディエンス向けには、WebRTC-to-HLSパイプラインに切り替えます — ブロードキャスター側の1秒未満のレイテンシーのためにWebRTC経由で取り込み、次にCDN経由で数千のビューアーに配信します。

WebRTCアプリケーションの構築にはどのくらい時間がかかりますか?

本番対応の1対1ビデオ通話機能は4~5週間かかります。録画、画面共有、モバイルサポート付きのグループ会議は7~10週間です。インクリメンタルに提供します — シグナリングと基本通話をまず提供し、その後記録、分析、カスタムメディア処理を積み重ねます。

WebRTC Development from $14,000
Fixed-fee. 30-day post-launch support included.
See all packages →
Next.js DevelopmentCore Web Vitals OptimizationCore Web Vitals Complete Guide 2026

Get Your Free WebRTC Assessment

Describe your use case. We'll deliver an architecture recommendation and quote within 24 hours.

Get a Free Assessment
Get in touch

Let's build
something together.

Whether it's a migration, a new build, or an SEO challenge — the Social Animal team would love to hear from you.

Get in touch →