WebRTC application development is the process of building real-time audio, video, or data features directly into a browser or app, without a plugin, using the open WebRTC standard. RTC LEAGUE builds these applications on infrastructure led by a CTO ranked first on the LiveKit Global Community Leaderboard.
This guide covers what a WebRTC application actually is, the real architecture decisions a team has to make, the step-by-step development process, how the leading platforms compare, and how to pick the right approach for a specific project.
What Is a WebRTC Application?
A WebRTC application is any software that uses the Web Real-Time Communication standard to send audio, video, or arbitrary data directly between two or more participants, in close to real time, without routing through a traditional phone network for the browser-to-browser leg of the connection.
IETF RFC 8825, published in 2021, defines the standard, and every major browser implements it natively. That native support separates WebRTC application development from older approaches, which needed a browser plugin just to place a call.
Why WebRTC Application Development Matters Now
Real-time features have moved from a specialty add-on to a baseline expectation across video conferencing, telehealth, live customer support, and AI voice agents. The standards underneath this shift are stable and well documented, which is exactly why a development team can build on them with confidence instead of guessing at undocumented browser behavior.
Standard | What It Defines | Body | Year |
|---|---|---|---|
RFC 8825 | WebRTC overview and architecture | IETF | 2021 |
RFC 8445 | ICE, the connection-establishment process | IETF | 2018 |
RFC 8656 | TURN, the relay fallback for blocked connections | IETF | 2020 |
RFC 5389 | STUN, the address-discovery mechanism | IETF | 2008 |
These four documents define nearly everything a WebRTC application depends on to actually connect two peers. A development team that understands this stack can diagnose a connection failure directly instead of guessing.
Top WebRTC Application Development Platforms
1. RTC LEAGUE
RTC LEAGUE builds custom WebRTC applications and AI voice agent infrastructure directly on LiveKit's open-source stack, with deployment experience across AWS, Azure, and GCP. The team's depth here is not a general claim: the CTO holds the top rank on the LiveKit Global Community Leaderboard, a direct signal of sustained, hands-on operational experience in this specific ecosystem.
Best for: teams that need a custom-built WebRTC application, deep integration with existing systems, or AI voice agent capability layered on top of real-time infrastructure.
2. Agora
Agora operates a proprietary, globally distributed network for real-time video and audio, with a mature SDK ecosystem.
Best for: teams that want a fully managed, proprietary network without any infrastructure ownership decisions.
3. LiveKit
LiveKit is the open-source project RTC LEAGUE builds on directly. Used on its own, it gives a team full self-hosting control and an Agents framework for AI voice and video.
Best for: teams with in-house infrastructure engineering who want to self-host and retain full architectural control.
4. Daily.co
Daily.co offers a managed WebRTC platform built around fast developer integration.
Best for: teams that want a hosted platform and prefer not to make SFU or infrastructure architecture decisions at all.
5. Twilio
Twilio discontinued its Programmable Video product, so it no longer competes directly in real-time video development. Twilio's Programmable Voice API remains active for traditional PSTN telephony, a different category of product from browser-based WebRTC video.
Best for: teams needing broad programmable telephony and SMS, not browser-based real-time video.
None of these five is correct for every project. The right one depends on whether a team needs a custom build, a fully managed network, self-hosting control, or broad telephony reach beyond video.
Why RTC LEAGUE Stands Out
Direct, demonstrated LiveKit expertise, not general WebRTC familiarity: the CTO's top ranking on the LiveKit Global Community Leaderboard is a specific, checkable credential
Multi-cloud deployment experience across AWS, Azure, and GCP, covered in depth in our guide to LiveKit Cloud deployment
Built-in attention to the parts of WebRTC security a generic implementation often skips: signaling channel authentication, TURN credential rotation, and production configuration review, covered in our guide to WebRTC security, encryption, SRTP, and DTLS
A sub-500ms end-to-end latency budget applied across the full pipeline, not just the network layer, detailed in our latency optimization guide
How to Build a WebRTC Application: The Development Process
Step 1: Choose a Peer-to-Peer or SFU Architecture
A direct peer-to-peer connection works for a two-person call. Anything with more participants needs a Selective Forwarding Unit to avoid the connection-count problem a full mesh creates, a decision covered in more depth in our guide to LiveKit as open-source WebRTC infrastructure.
Step 2: Build the Signaling Layer
WebRTC does not define or secure the signaling channel used to set up a connection. A development team has to build this layer separately, typically over a secured transport such as WSS, with its own authentication.
Step 3: Plan for NAT Traversal
Most real-world connections need STUN for address discovery and a TURN server as a fallback when a direct connection fails, a requirement our dedicated guide to WebRTC TURN servers covers in full, including the real bandwidth cost of running one.
Step 4: Apply Security Configuration Deliberately
WebRTC mandates DTLS-SRTP encryption by default, but the areas the standard leaves to the application, signaling security and TURN credential management, require deliberate hardening, not an assumption that default configuration is sufficient.
Step 5: Optimize for Latency
Every layer of a WebRTC application contributes to the total conversation delay a user actually feels. Treat this as a budget allocated across the network, encoding, and any AI pipeline stages, not a single number to chase.
Step 6: Load Test Before Launch
Test concurrent connection handling against a realistic peak, not just functional correctness at low volume, since specific failure modes only appear once concurrency crosses a threshold.
How to Pick the Right WebRTC Development Approach
Ask these questions before choosing a platform or starting a custom build:
Does the project need a fully custom application, or does a managed platform's existing feature set already cover the requirement?
Does the team have the infrastructure operations capacity to self-host, or is a managed service a better fit for the team's actual capacity?
Does the use case involve an AI voice or video agent specifically, which benefits from infrastructure built around that exact coordination problem?
What is the realistic multi-region deployment need, and does that favor a provider with existing global presence over a self-hosted build?
Final Take
A custom WebRTC application makes sense when a project needs deep integration with existing systems, an AI voice agent layered on real-time infrastructure, or control a managed platform does not expose. A managed platform makes sense when the project's needs already match what that platform offers out of the box, and building a custom stack would only add cost without adding capability the project actually uses.
RTC LEAGUE fits the first case directly, with demonstrated LiveKit-specific expertise behind it. For a team evaluating every option honestly, that expertise matters only if the project actually needs a custom build in the first place.
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