Custom WebRTC apps for real-time video, audio and data streaming on mediasoup, LiveKit, Janus and Pion. P2P, SFU and MCU patterns. Sub-300 ms latency, multi-region SFU, HIPAA and SOC 2.
Fora Soft — 625+ real-time products shipped since 2005, including BrainCert (500M+ minutes), Nucleus (600M+ call minutes / month, HIPAA + HITRUST) and TransLinguist (NHS UK).
We design & develop end-to-end WebRTC solutions: live video apps and video & audio streaming that just work. We use STUN/TURN, SFU/MCU, and AI-driven analytics to ensure stable performance across devices and networks.
No matter the size or complexity of your project, we’ll take it on and get it done. No excuses.
20+ years on real-time media. From one-on-one calls to broadcasts for 10,000+ concurrent users. mediasoup, LiveKit, Janus, Pion in production.
Sub-300 ms glass-to-glass latency on simulcast SFU. Edge POPs (Hetzner / AWS / Cloudflare), congestion control, TURN fallback, FEC — same stack we run for Nucleus (600M+ minutes / month).
Client-side noise suppression (RNNoise / Krisp), background blur (MediaPipe), live captions (Whisper / Deepgram), moderation, voice agents (LiveKit Agents, OpenAI Realtime). Side-bus, never blocks the live RTP path.
We build real-time communication apps that send video, audio, and data directly between users with minimal delay. Under the hood, WebRTC handles peer-to-peer media transfer, while our backend keeps everything secure, scalable, and reliable.
Here’s how a typical WebRTC system we build works:
When a user starts a call or stream, the app connects to a signaling server.
This server helps participants find each other and exchange connection details.
Many users are behind firewalls or NAT.
We use STUN and TURN servers to establish a stable media path even on restricted networks.
For group calls or live streams, media flows through an SFU (Selective Forwarding Unit) or MCU (Multipoint Control Unit):
Audio and video are encoded using codecs like VP8, VP9, H.264, and Opus.
We tune bitrate, resolution, and network adaptation to keep latency low and quality high.
We can add AI-powered features such as noise suppression, background blur, speaker detection, live transcription, or moderation – all processed in real time.
We integrate analytics to track packet loss, jitter, latency, and device performance.
This allows automatic adjustments and long-term quality improvements.
Result: smooth, low-latency video and audio that works across devices, browsers, and network conditions.
Our WebRTC architecture is designed for low latency, high reliability, and horizontal scalability.
A typical system includes the following layers:
Result: This modular architecture allows your WebRTC platform to grow from small MVPs to systems with thousands of concurrent users.
* Client Applications
Web, iOS, Android, or desktop apps built with WebRTC SDKs. These handle media capture, encoding, and playback on the user’s device.
* Signaling Server
A secure backend service that manages session setup, user presence, and call control. This is where peers exchange connection data before media starts flowing.
* STUN / TURN Servers
Infrastructure that ensures users can connect even behind strict corporate or mobile networks. TURN relays traffic when direct peer-to-peer is not possible.
* Media Servers (SFU / MCU)
Core real-time media layer for multi-user sessions: SFU for scalable video conferencing and live classes; MCU for stream mixing, recording, or legacy device support.
* Application Backend
Handles business logic: user accounts, permissions, payments, scheduling, chat, and integrations with third-party systems (CRM, EHR, LMS, etc.).
* AI & Media Processing Services (Optional)
Separate services for speech-to-text, content moderation, noise removal, or computer vision tasks such as face tracking or activity detection.
* Monitoring & DevOps Layer
We deploy logging, metrics, and alerting to track call quality, server load, and uptime. Infrastructure is typically hosted on AWS, GCP, or Azure with autoscaling.
Custom WebRTC apps for every case. Secure, scalable, and packed with smart features – built by the pros who pioneered it.
Have an idea? We’ll turn it into a fully working app – from design and backend to launch and support.
Got a product that needs more speed, stability, or features? We’ll make it stronger and ready to scale.
Struggling with unfinished or broken code? We’ll step in, clean it up, and get your project back on track.
Startup 💡
WebRTC MVP — 1:1 video / audio call, basic SFU on a single Hetzner box (mediasoup or LiveKit), STUN, TURN, recording to S3, simple admin. Right for validating an idea or replacing Zoom inside a niche product.
$8,000
from 4-5 weeks
Growth 🚀
Multi-party SaaS — multi-region SFU, TURN, simulcast / SVC, recording, transcription (Whisper, Deepgram), mobile SDKs (iOS / Android), RBAC, SOC 2-ready logging, custom UI on web + iOS + Android.
$25,000
from 2-3 months
Enterprise 🏢
Enterprise / production — HIPAA + HITRUST / SOC 2 hardening, on-premise or VPC, AI hooks (live captions, moderation, summaries), white-label SDKs, 99.9%+ SLO, on-call support.
$80,000
from 3-5 months
Perfecting WebRTC and real-time apps since 2005. From CirrusMED telehealth to ProVideoMeeting conferencing — we’ve shipped real-time apps used by millions.
Senior developers, QA, UI/UX, analytics – all in-house. No outsourcing, no handoffs. We think like product owners, not just coders.
Over 600 completed projects, 100% Upwork Success rate, and 400+ client reviews. Trusted by startups and enterprises across 30+ countries.
Quick answers about WebRTC development costs, timelines, AI integration, and what makes our approach different
We build the full range: 1-on-1 video chat, group conferencing (up to 10,000+ participants), telehealth platforms, e-learning classrooms, live streaming apps, remote collaboration tools, and real-time customer support systems. Every app is custom-built for your use case.
Absolutely. We integrate AI-powered features like real-time transcription, live translation, noise suppression, background blur, speaker detection, and content moderation — all processed in real time during calls or streams.
MVPs start around $8K (4–5 weeks), scaling products around $25K (2–3 months), and enterprise systems from $50K+ (3–5 months). We provide free estimates upfront so you know exactly what to expect before committing.
We deploy STUN/TURN servers for NAT traversal, adaptive bitrate streaming, intelligent load balancing, and codec optimization (VP8/VP9, H.264, Opus). Every build is stress-tested across real-world network conditions to guarantee sub-second latency.
600+ completed projects, 100% Upwork Success rate, and 20 years of experience. We combine senior engineers, in-house QA, real-time analytics, and proven architecture patterns to deliver apps that are scalable, stable, and production-ready from day one.
