September 25, 2025
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WebRTC has been enabling video and audio communication directly in your browser without any plugins for 10 years now. Even services like Google Meet and Discord use WebRTC to provide crystal-clear voice and video calls in real-time. This powerful technology has revolutionized how we connect online, but getting started with it can seem daunting. 

In this WebRTC tutorial, we will break down the core WebRTC concepts, show you how communication is established, and outline what it takes to build your own WebRTC application. 

What is WebRTC?

The goal of WebRTC is to enable two or more devices, known in this context as peers, to establish a direct, secure connection to exchange video, audio, and arbitrary data in real-time.

Prior to WebRTC, this was only possible through proprietary plugins or specific software solutions that weren’t universally available. Recognizing this limitation, industry leaders collaborated to create an open standard that brings real-time communication to every modern browser.

This led to two key things:

  1. An open standard that describes the protocols and guidelines for establishing these connections.
  2. An open-source implementation, libWebRTC, maintained mainly by Google, which provides the nuts and bolts to make it work.

This is the implementation you’re using when you join a Google Meet call or use the voice mode in ChatGPT on a web browser. There are also other implementations that either fork libWebRTC or are written from scratch to provide these capabilities outside the browser, like in native mobile apps. Some examples of such implementations are aiortc for Python and pion for Go.

As a developer, your job is to leverage the APIs provided by these implementations -either directly or through a high-level SDK that abstracts them- to power real-time communication functionality in your application. 

The three main APIs you’ll work with are:

  • getUserMedia()/getDisplayMedia(): Grants access to a device’s camera, microphone, and screen.
  • RTCPeerConnection: Manages the entire process of connecting with another peer.
  • RTCDataChannel: Allows you to send any kind of data, not just media.

So, the first step to getting started is to familiarize yourself with the WebRTC implementation or SDK you’ll be using and the syntax of these essential APIs. But before your first line of code, it’s important to understand how communication works and how media is exchanged in such connections.

How is WebRTC Communication Established?

One of the main perks of WebRTC is its ability to establish direct peer-to-peer (P2P) connections, which allows media to be sent from one device to another without passing through an intermediary server.

For this connection to take place, the peers first need to negotiate terms. This process is called Signaling. It’s like a pre-call handshake where peers exchange information about themselves. WebRTC doesn’t define a specific signaling protocol; you’re free to implement it however you like, whether through WebSockets, a REST API, or even a carrier pigeon (or raven if you’re in Westeros).

The information exchanged includes:

  • Session Description Protocol (SDP) documents, known as an Offer and an Answer. These contain metadata such as what codecs are available, encryption details, and more.
  • Interactive Connectivity Establishment (ICE) candidates. These are potential network paths (IP address and port pairs) that the peers can use to connect.

The whole process generally looks like this:

  1. Peer A creates an SDP Offer and sends it to Peer B through the signaling channel.
  2. Peer B receives the Offer and creates an SDP Answer, which it sends back to Peer A.
  3. As this happens, both peers gather ICE candidates and exchange them through the signaling channel. The ICE framework then tests these paths to find the best one for a direct connection.
Depiction of WebRTC signaling process between two peers.
Depiction of signaling process between two peers

Navigating NAT with ICE Servers

In an ideal world, every device would have a unique public IP address. But in reality, most devices are behind a Network Address Translation (NAT) device (like your home router), which complicates direct connections. To solve this, the ICE framework uses special servers:

  • STUN (Session Traversal Utilities for NAT) servers help a peer discover its own public IP address and port. The peer can then share this information as an ICE candidate.
  • TURN (Traversal Using Relays around NAT) servers are a fallback. When a direct connection isn’t possible (often due to strict corporate firewalls), the TURN server acts as a relay, forwarding media between the peers. It doesn’t process the media; it just passes it along.

The Role of Media Servers

While P2P is great for one-on-one calls, it doesn’t scale well for group sessions. If ten people are in a call, your device would need to manage nine separate connections, which consumes a lot of CPU and bandwidth.

This is where Media Servers come in. A media server is a specialized peer that acts as a central hub. Instead of connecting to everyone directly, each participant connects only to the media server. The server then distributes the media to everyone else. 

Choosing a media server architecture is crucial for scaling sessions to support many participants, and enabling advanced features like recording, transcoding, and real-time analysis.

A fascinating new use case for media servers is allowing AI agents to join WebRTC sessions. Users connect to the media server, and the AI agent joins as another participant, processing the media stream to provide intelligent responses or perform tasks.

Media Servers work in one of two ways:

MCU (Multipoint Conferencing Unit)

An MCU acts like a virtual video mixer. It receives the individual media streams from every participant in the call, decodes them, and combines them into a single, composite stream. It then re-encodes this single stream and sends it back to each participant.

Think of traditional video conferencing systems where everyone appears in a grid layout that’s determined by the server. The main advantage is that clients only have to handle one incoming stream, which reduces the load on their device.

However, this process is very CPU-intensive for the server and offers less flexibility for the client to customize the layout.

SFU (Selective Forwarding Unit)

An SFU is a much more lightweight and scalable approach, and it’s the dominant architecture for modern WebRTC applications.

Instead of mixing streams, an SFU acts like a media router. It receives an incoming stream from one participant and simply forwards it, untouched, to all other participants in the session. This means each client receives multiple individual streams (one from each other participant) and is responsible for arranging and displaying them.

While this requires more downstream bandwidth and client-side processing than an MCU, it’s far less than a full mesh network. This model is significantly less demanding on the server and gives developers complete control over the user interface and layout on the client side.

How is WebRTC Media Exchanged?

Once a connection is established, peers start exchanging data. Audio and video must be compressed using a codec (encoder/decoder) to be sent efficiently over the network. It’s crucial that both peers support the same codec. 

The mandatory codecs for WebRTC are:

  • Video: VP8 and H.264
  • Audio: Opus and G.711

At a lower level, WebRTC uses two different protocols for transporting packets of compressed media:

  • SRTP (Secure Real-time Transport Protocol) is used for sending audio and video packets. It’s a battle-tested protocol from the VoIP world, perfect for media transmission. It’s paired with RTCP (RTP Control Protocol) to handle things like packet loss and congestion control.
  • SCTP (Stream Control Transmission Protocol) is used for the RTCDataChannel. Its flexibility is key—you can configure it to be reliable and ordered like TCP or fast and unordered like UDP, depending on your application’s needs.

At this point you have the basic knowledge to know your way around WebRTC-based applications. But how do you actually get started?

What Does a WebRTC Development Project Look Like?

Building a WebRTC application involves more than just writing code. It’s a complete endeavor, and the most critical decision you’ll make is how to handle the complex infrastructure required for real-time communication. This choice conditions your entire development approach.

The Architecture of a WebRTC App: The Build vs. Buy Decision

At the core of your application is the WebRTC Platform—the specialized infrastructure that handles signaling, connectivity, and media routing. Your first decision is whether to build this yourself or buy it as a service.

The “Buy” Approach: CPaaS (Communications Platform as a Service)

This approach involves outsourcing the WebRTC infrastructure to a third-party provider. For a monthly fee based on usage, they manage the global fleet of servers (Signaling, STUN/TURN, SFU/MCU), ensuring reliability and scale, while you focus on building your application’s unique features.

Some of the leading CPaaS Providers you can choose from are:

You can also “buy” only the ICE servers and build the rest of the platform. In such case, some well-known STUN/TURN providers are:

To get started with these NAT traversal services, check out our Selecting and Deploying Managed STUN/TURN Servers post.

2. The “Build” Approach: Self-Managed with Open Source

This approach gives you maximum control and can be more cost-effective at a very large scale, but it requires deep expertise in real-time networking and DevOps. You will be responsible for deploying, managing, and scaling your own signaling, ICE and media servers.

Some popular open-source Media Servers you can leverage are:

For STUN/TURN there are also open source solutions that includes:

To get started with these open source ICE solutions check out our How to Set Up Self-Hosted STUN/TURN Servers for WebRTC Applications post.

How “Build vs. Buy” Shapes Your Frontend and Backend

Your decision directly impacts how your frontend and backend teams will work.

Frontend Development:

With a CPaaS: Your developers will use the provider’s client-side SDKs (e.g., amazon-chime-sdk-js, daily-js, etc). These SDKs abstract away the low-level complexities of RTCPeerConnection, simplifying tasks like creating rooms, publishing streams, and handling events.

With a Self-Managed Platform: Your developers will work directly with the browser’s native WebRTC APIs or use the specific client library provided by your platform and/or chosen open source media server (e.g., LiveKit’s client-sdk-js, react-native-webrtc, etc). This offers more control but requires a deeper understanding of WebRTC’s inner workings.

Backend Development:

With a CPaaS: The backend’s primary role is to leverage providers’ server-side APIs to authenticate users and generate secure access tokens that the frontend uses to connect to the CPaaS service. The heavy lifting of signaling and media management is offloaded to the provider.

With a Self-Managed Platform: In addition to managing authentication, your backend implements the signaling mechanism, manages room and user state, and integrates directly with your deployed WebRTC platform using the APIs exposed by its components.

Your WebRTC Dream Team 🧑‍💻

Assembling the right team is crucial. You’ll need specialists who understand the unique challenges of real-time communication:

  • UI/UX Designer: Focuses on creating intuitive experiences for real-time interactions, handling various call states (connecting, muted, poor connection), and minimizing perceived latency.
  • Frontend Developer: Needs strong familiarity with the WebRTC browser APIs and any high-level SDKs being used.
  • Backend Developer: Requires experience building signaling servers, integrating with media servers, and potentially using server-side WebRTC implementations for bots or AI agents.
  • DevOps Engineer: Must be skilled in deploying, maintaining and monitoring high-availability, low-latency infrastructure suitable for real-time traffic and VoIP.
  • QA Engineer: Needs expertise in testing complex real-time workflows, simulating different network conditions (packet loss, jitter), and ensuring cross-device compatibility.

The Five Types of WebRTC Developers: Which do you need?

Wrapping Up

WebRTC is a powerful framework that has democratized real-time communication on the web. It enables secure, direct peer-to-peer connections by using a clever combination of signaling, ICE servers for NAT traversal, and media servers for scalability. Media and data are exchanged efficiently using battle-tested protocols like SRTP and SCTP. While building a WebRTC application from scratch is a significant undertaking that requires a specialized team and a robust architecture, the results can be incredibly rewarding.

Need Help Building Your WebRTC Application?

Building a robust, scalable WebRTC application requires a team with specialized skills. Whether you need to augment your existing team or want a dedicated group of experts to build your application from the ground up, we can help.

  • Want us to build it for you? Our team at WebRTC.ventures lives and breathes real-time communication. We can design, build, and deploy your entire WebRTC application, ensuring a world-class experience for your users.
  • Ready to move beyond WebRTC projects? Our parent company, AgilityFeat, can help you build a full engineering team with top nearshore talent in Latin America. 

Ready to dive in and build the next big communication app? Contact WebRTC.ventures today!

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