WebRTC Explained

WebRTC is one of the most important technologies behind modern real-time communication on the web. It allows browsers, mobile apps, and embedded devices to exchange audio, video, and arbitrary data with very low latency, often without requiring users to install plugins or specialist software.

For businesses, WebRTC can power video consultations, live support, remote collaboration, telehealth platforms, online education, multiplayer experiences, secure file transfer, IoT control panels, and real-time operational dashboards. Used well, it creates fast, interactive digital products that feel immediate and natural.

What is WebRTC?

WebRTC, short for Web Real-Time Communication, is a collection of open standards and browser APIs that enable peer-to-peer communication. Instead of sending every packet through a central media server by default, WebRTC can connect users directly when network conditions allow it.

The technology is built into major browsers and is also available for native mobile and desktop applications. This means developers can build real-time communication features using JavaScript, native SDKs, and backend services without relying on proprietary plugins.

Core WebRTC capabilities

WebRTC is usually discussed in the context of video calls, but it is broader than that. Its main capabilities include:

• Audio streaming: low-latency microphone capture and playback for calls, voice chat, support desks, and conferencing.
• Video streaming: camera, screen-sharing, and media track delivery for meetings, remote assistance, education, and live collaboration.
• Data channels: peer-to-peer transfer of arbitrary data, useful for chat, file transfer, gaming state, device telemetry, and collaborative editing.
• Encryption: secure media and data transport using protocols such as DTLS and SRTP.
• Network traversal: support for connecting users across NATs, firewalls, and different network environments using ICE, STUN, and TURN.

How WebRTC works

A WebRTC connection involves several moving parts. The browser or app captures media, negotiates capabilities, discovers network routes, secures the connection, and then streams audio, video, or data between endpoints.

1. Media capture

The getUserMedia API lets an application request access to a user's microphone and camera. The browser controls permission prompts, helping protect user privacy while giving applications access to real-time media streams.

2. Signalling

WebRTC does not define a single signalling protocol. Applications must provide their own signalling layer to exchange session descriptions, call state, participants, and connection candidates. This is often implemented using WebSockets, HTTP APIs, server-sent events, or a messaging system.

During signalling, clients exchange SDP offers and answers. These describe supported codecs, media tracks, transport options, and encryption parameters. Signalling is not the media path itself; it is the coordination layer that helps peers find and agree how to communicate.

3. ICE, STUN, and TURN

Real-world networks are messy. Users may be behind routers, mobile carriers, corporate firewalls, VPNs, or restrictive NAT configurations. WebRTC uses ICE, or Interactive Connectivity Establishment, to test possible connection paths.

• STUN servers help clients discover their public-facing network address.
• TURN servers relay traffic when direct peer-to-peer connectivity is not possible.
• ICE candidates represent possible routes between peers, which are tested until the best working path is found.

TURN infrastructure is especially important for production systems. Without reliable TURN capacity, some users will fail to connect, particularly on locked-down enterprise networks or mobile networks.

4. Peer connection

The RTCPeerConnection API manages the secure connection between endpoints. It handles media transport, codec negotiation, encryption, congestion control, bandwidth adaptation, packet loss recovery, and connection state.

5. Data channels

The RTCDataChannel API allows applications to exchange non-media data with low latency. Data channels can be configured for reliable or partially reliable delivery, making them useful for everything from messaging to multiplayer interactions.

Common WebRTC architectures

The right architecture depends on the number of users, media quality requirements, compliance needs, cost constraints, and whether sessions need to be recorded or moderated.

Peer-to-peer

In a simple one-to-one call, two clients may connect directly. This can reduce server bandwidth costs and latency, but it depends on network conditions and still requires signalling and STUN/TURN services.

Mesh

In small group calls, each participant can connect to every other participant. This is easy to understand but does not scale well because each user must upload multiple streams. Mesh architectures are usually unsuitable for larger meetings.

SFU

A Selective Forwarding Unit receives media streams from participants and forwards selected streams to others. SFUs are widely used for video conferencing because they scale better than mesh while keeping latency relatively low.

MCU

A Multipoint Control Unit mixes or transcodes media streams on the server. MCUs can simplify client requirements and produce a single composed stream, but they are more compute-intensive and can increase cost and latency.

Where WebRTC is used

WebRTC is suitable for many real-time digital services, including:

• Video consultations and appointment platforms
• Live customer support and co-browsing
• Remote technical assistance
• Virtual classrooms and tutoring platforms
• Team collaboration tools
• Telehealth and wellbeing products
• Field-service communication apps
• Low-latency live streaming workflows
• Browser-based voice and video chat
• Real-time IoT monitoring and control
• Secure peer-to-peer file transfer
• Multiplayer games and shared interactive spaces

Benefits of WebRTC

WebRTC is popular because it combines user convenience with technical power. Key benefits include:

• Low latency: designed for interactive communication where delay matters.
• Browser support: works in modern browsers without plugins.
• Secure by design: encryption is built into the transport layer.
• Flexible media handling: supports audio, video, screen sharing, and data.
• Cost efficiency: direct connections and SFU architectures can reduce unnecessary media processing.
• Cross-platform reach: usable across web, mobile, desktop, and embedded environments.

Challenges in production

Building a demo WebRTC application can be quick. Building a reliable production service is harder. Teams need to consider network resilience, observability, security, scaling, user experience, infrastructure cost, and browser differences.

Connectivity failures

Some networks block direct peer-to-peer connections. Production systems need robust TURN infrastructure, sensible retry behaviour, and clear connection-state handling in the user interface.

Scaling and bandwidth

Group calls and high-quality video can consume significant bandwidth. Architecture choices such as SFU deployment, simulcast, adaptive bitrate, and regional routing have a major effect on user experience and cloud costs.

Device and browser variation

Cameras, microphones, mobile browsers, permissions, audio routing, and codec support vary across devices. Applications need careful testing across real user environments.

Monitoring and debugging

WebRTC issues are often intermittent and network-specific. Good telemetry should include connection state, ICE candidate types, packet loss, jitter, round-trip time, bitrate, resolution, codec, device information, and user-impacting errors.

Security and privacy

WebRTC applications handle sensitive media and identity flows. Teams should design secure authentication, authorization, room access, token expiry, recording controls, data retention, audit trails, and compliance processes from the start.

WebRTC and cloud infrastructure

Production WebRTC often requires more than application code. A reliable service may include signalling APIs, media servers, TURN relays, regional deployment, load balancing, autoscaling, observability, CI/CD pipelines, and infrastructure-as-code.

Cloud choices matter. AWS, containers, Docker, Kubernetes, Terraform, and automated deployment pipelines can all play a role in making WebRTC platforms scalable and maintainable. The correct setup depends on traffic volume, geography, reliability targets, security requirements, and budget.

Best practices for WebRTC projects

• Start with a clear use case: one-to-one calls, group rooms, streaming, remote support, or data transfer.
• Choose the architecture early: peer-to-peer, SFU, MCU, or a hybrid approach.
• Plan TURN capacity before launch, not after users report failed calls.
• Test on real networks, including mobile, office Wi-Fi, VPNs, and restrictive firewalls.
• Design the signalling layer as a reliable product component, not a throwaway demo service.
• Implement metrics and logging from the beginning.
• Use secure room access, short-lived tokens, and careful permission handling.
• Optimise media quality dynamically using simulcast, adaptive bitrate, and sensible defaults.
• Make reconnection, mute state, device switching, and permission errors clear to users.
• Automate deployment and infrastructure changes to reduce operational risk.

When to use WebRTC

WebRTC is a strong fit when users need to interact live. If your product depends on conversation, shared context, immediate feedback, or low-latency device communication, WebRTC is often the right foundation.

However, it is not always the best choice for one-way broadcast video, long-form media hosting, or situations where several seconds of delay are acceptable. In those cases, HLS, DASH, or managed streaming platforms may be more appropriate. A good architecture review can prevent expensive rework later.

How Eight Mile can help

Eight Mile can support WebRTC projects across the full delivery lifecycle: product discovery, system architecture, custom software development, web and mobile applications, backend APIs, cloud infrastructure, AWS deployment, Docker and Kubernetes environments, Terraform infrastructure, CI/CD pipelines, workflow automation, technical consultancy, and legacy modernisation.

Whether you are building a new video platform, adding live support to an existing application, modernising an older communications system, or scaling a real-time product for production traffic, the right technical decisions will affect reliability, cost, security, and user experience.

WebRTC is powerful, but production success depends on the surrounding engineering. Signalling, media infrastructure, network traversal, observability, security, deployment, and support processes all need to work together.

Conclusion

WebRTC enables real-time communication directly inside web and mobile products. It can deliver high-quality audio, video, screen sharing, and data exchange with low latency, but it requires thoughtful architecture to perform reliably in the real world.

For organisations planning a real-time communication product, the best results come from treating WebRTC as part of a complete system: frontend experience, backend APIs, media infrastructure, cloud deployment, monitoring, security, and ongoing operations.

Need help planning, building, or scaling a WebRTC service? Contact Eight Mile to discuss your project.