When a customer clicks 'Talk to Us' on your website and a voice call starts immediately — no app download, no plugin, no phone number required — that's WebRTC. It's reached a level of browser compatibility and tooling maturity that makes it a serious choice for enterprise communication applications.
What WebRTC Actually Is
Web Real-Time Communication is a set of browser APIs that enable peer-to-peer audio, video and data sharing. The signaling layer goes through a server, but the actual media travels directly between clients. This is why WebRTC calls have low latency and don't require your media to pass through a third-party server.
Enterprise Use Cases That Are Production-Ready
Click-to-call from web properties, in-app customer support voice chat, browser-based softphones for call center agents, video consultations for professional services. Each can be deployed without asking your customers or agents to install anything.
What to Watch For in Implementation
Network traversal is the hard problem. WebRTC requires STUN and TURN servers to handle NAT traversal. Provisioning adequate TURN capacity with geographic coverage that matches your user base is often underestimated in project planning.
How It Connects to Your Existing Stack
WebRTC integrates cleanly with SIP-based VoIP infrastructure through SIP-WebRTC gateways. A browser-based caller can talk directly to a traditional SIP endpoint without either side knowing the other's technology.
Browser support is no longer the limiting factor it once was. Chrome, Firefox, Safari and Edge all support WebRTC without plugins. Mobile browsers on both iOS and Android support it natively. For enterprise deployments, browser version management through MDM systems makes compatibility predictable.
The latency profile of WebRTC makes it suitable for use cases where delay would be noticeable. Under good network conditions, WebRTC calls achieve sub-100ms round-trip latency. For comparison, traditional PSTN calls typically have 100–250ms of latency. WebRTC's peer-to-peer architecture — bypassing intermediate servers for media — is what makes this possible.
Data channels are a lesser-known WebRTC capability that open up interesting application possibilities. Beyond audio and video, WebRTC data channels enable arbitrary data to be sent peer-to-peer: file transfers during a call, real-time collaborative document editing, screen state synchronization. For enterprise applications that combine communication with workflow, data channels are worth investigating.
Ready to modernise your voice infrastructure?
Express IVR's Cloud Contact Center supports WebRTC-based click-to-call for customer-facing applications and internal agent tooling. If you're designing a product that needs real-time voice, we'd like to understand your requirements.
