Build custom WebRTC apps for real-time video, audio, and data streaming.
Fora Soft – top WebRTC developers for startups and enterprise platforms since 2005.
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.
14+ years hands-on with WebRTC, from one-on-one calls to massive live broadcasts for 10,000+ users.
Deliver video and audio in real-time (as little as 0.5s delay) for telehealth, e-learning, and live events.
Boost your platform with AI-driven features – noise suppression, background effects, or real-time transcription.
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 💡
Perfect for MVPs: fast launch, core features, and a foundation to test your idea.
~$8,000
from 4-5 weeks
Growth 🚀
Ideal for scaling products: advanced functionality, integrations, and performance tuning.
~$25,000
from 2-3 months
Enterprise 🏢
Built for mission-critical systems: heavy traffic, complex infrastructure, and robust security.
~$50,000
from 3-5 months
Perfecting WebRTC and real-time apps since day one – reliable solutions that deliver real value.
Senior developers, QA, UI/UX, analytics – all in-house. We think like product owners, not just coders.
Over 600 completed projects, 100% Upwork Success rate, and 400+ honest clients' reviews. Results you can trust.
Get the scoop on video, chat, and building with WebRTC – straight talk from the top WebRTC devs
Everything from one-on-one video chat to telehealth platforms, conferencing tools, live streaming apps, e-learning systems, and beyond.
You bet! We bake in AI for live captions, translations, and smart video – making your app a genius.
It depends on complexity, but we provide free ballpark estimates upfront. Pricing usually falls into startup, growth, or enterprise tiers.
We optimize with STUN/TURN servers, adaptive bitrate streaming, load balancing, and codec tuning (VP8/VP9, Opus). We also test across real-world network conditions.
We combine senior engineers with in-depth analytics, in-house QA, and proven best practices, ensuring apps are scalable, stable, and user-friendly.
