Is WebRTC free to use in a mobile app?
Yes — WebRTC is a free, open W3C/IETF standard with no licensing fee. CelloIP builds native WebRTC mobile apps with self-hosted SFU and TURN infrastructure, avoiding the per-minute fees of managed APIs like Agora.
How many participants can a WebRTC mobile call support?
Peer-to-peer WebRTC supports roughly 3–4 participants before quality degrades. Beyond that, a Selective Forwarding Unit such as LiveKit or Mediasoup is required, supporting 100+ participants per call.
WebRTC Mobile App Development
Custom real-time voice, video, and data apps on native WebRTC for iOS and Android — with self-hosted SFU and TURN infrastructure. Free, open standard; no mandatory per-minute API fee.
What We Build
Native WebRTC SDK Integration
Google's native WebRTC library integrated directly into iOS (Swift) and Android (Kotlin) apps — full control over peer connection management, ICE handling, and camera/mic pipelines.
Group Calling via SFU
Beyond 3–4 participants, peer-to-peer WebRTC breaks down. CelloIP integrates LiveKit or Mediasoup as a Selective Forwarding Unit for group calls up to 100+ participants.
Self-Hosted STUN/TURN
Dedicated coturn TURN server deployment for NAT traversal — the infrastructure managed APIs bill per-minute for, self-hosted at a flat monthly cost.
Real-Time Data Channels
WebRTC DataChannel apps for collaborative tools, file transfer, live location sharing, and low-latency game state sync — built into the same mobile app as your voice/video features.
Cross-Platform WebRTC Apps
Flutter apps using LiveKit's official Flutter SDK, or React Native with @livekit/react-native — one codebase for iOS, Android, and (for Flutter) Web.
WebRTC-to-SIP Gateway
Bridge mobile WebRTC calls to your existing SIP PBX or carrier trunk — useful when a WebRTC-based app needs to reach PSTN numbers or an Asterisk/FreeSWITCH backend.
Native SDK vs Raw WebRTC vs Managed API
WebRTC is a definitional building block, not an out-of-the-box app — it's the free, open standard that WhatsApp, Google Meet, and Discord all build their real-time calling features on top of. For a mobile app, there are three practical ways to use it: integrate Google's native WebRTC library directly (maximum control, most engineering effort), use LiveKit's official Flutter/React Native SDKs (near-native performance with dramatically less integration work), or use a managed API like Agora (fastest to a demo, most expensive at scale).
CelloIP's default recommendation for new mobile WebRTC apps is LiveKit, specifically because it eliminates the two hardest parts of raw WebRTC integration — ICE/SDP negotiation and SFU media routing for group calls — while remaining self-hostable, so the cost advantage of "own your infrastructure" isn't traded away for integration speed.
WebRTC Mobile App Development FAQ
Is WebRTC free for commercial use?
Yes — WebRTC is a free, open, W3C/IETF standard implemented natively in browsers and available as an open-source library for native mobile apps. There is no licensing fee for using WebRTC itself; the cost is engineering and infrastructure (TURN servers, SFU media servers) to run it in production.
Can you build WhatsApp with WebRTC?
Yes — WhatsApp's voice and video calling features are built on WebRTC, as are Google Meet, Discord's voice channels, and most modern real-time communication apps. WebRTC provides the media transport layer; the app-specific features (messaging, presence, contact sync) are built separately on top.
WebRTC vs Agora — which is better?
Raw WebRTC (self-hosted, or via LiveKit) is better for cost control at scale and full data ownership — no per-minute fee. Agora is faster to integrate for an early prototype but bills $3.99 per 1,000 HD minutes, which compounds significantly once an app has meaningful usage. CelloIP generally recommends starting on a managed option like LiveKit Cloud, then migrating to self-hosted WebRTC/LiveKit once usage justifies the ops investment.
How many users can a WebRTC call support?
Direct peer-to-peer WebRTC supports roughly 3–4 participants before call quality degrades, since every device must upload a separate stream to every other participant. Beyond that, a Selective Forwarding Unit (SFU) like LiveKit or Mediasoup is required, supporting 100+ participants per call.
Do I need a TURN server for a mobile WebRTC app?
Yes, for any production app. Roughly 20–40% of real-world network conditions (corporate firewalls, symmetric NATs, some carrier networks) block direct peer-to-peer connections, requiring a TURN relay server to keep the call working. Skipping TURN means a meaningful fraction of users will experience silently failed calls.
Start Your WebRTC Mobile App
Book a free technical assessment with a CelloIP WebRTC engineer.