When WebRTC was first introduced by Google over a decade ago, it came with the promise of simplicity. “Just drop in a little JavaScript and you’ve got video chat in the browser with no downloads necessary!” While that vision helped kickstart a wave of innovation in real-time communication, the reality has always been more complex. Especially when building real-time communication apps at scale.
At WebRTC.ventures, we regularly hear from teams who have working demos that run fine on their machines, but break down in staging or production, or as they try to scale beyond beta users. Why are there so many WebRTC development challenges after all these years?
Let’s take a look at a few reasons why real-time communication app development with WebRTC remains deceptively complex.
#1: Establishing WebRTC Peer Connections Isn’t Always Easy
One of the most common causes of “it works on my machine” issues in WebRTC applications comes down to improper configuration of STUN and TURN servers. These servers are essential for establishing peer-to-peer connections across networks, especially when dealing with corporate firewalls or NAT traversal.
While many developers know of the necessity for a TURN server to prevent failed connections, setting up robust and scalable TURN infrastructure for a production environment is a different challenge entirely. High availability, low latency, and geographic distribution all play a role. Getting this wrong can kill your user experience.
#2: Scaling WebRTC Beyond 1-1 Calls
The original WebRTC use case was a simple one-to-one video conversation with no intermediary media server. This is still the most straightforward way to use WebRTC, but also the least common in production applications.
While a single peer-to-peer call is simple to implement, scaling to support thousands of concurrent 1-1 connections introduces significant infrastructure challenges around signaling servers, STUN/TURN relay management, and load balancing. Beyond volume, most production applications also need multi-party calls, which fundamentally changes the architecture from peer-to-peer to one with media servers in the middle.
CPaaS platforms make it easier to scale WebRTC, whether it’s hundreds of concurrent group calls or even real-time broadcasting to thousands of viewers. However, CPaaS platforms must still be architected carefully to manage costs. This requires a deep understanding of both WebRTC internals and cloud DevOps practices in order to make the best decisions around media server topology and bandwidth optimization.
#3: WebRTC Broadcasting, Live Streaming, and Low-Latency Challenges
WebRTC shines in low-latency scenarios, but building a one-to-many or few-to-many broadcast system with it isn’t trivial. CPaaS vendors have made strides in supporting these patterns, but you still have to choose the right tools, configure them correctly, and build an intuitive, reliable user experience around them.
Whether you’re livestreaming town halls, powering virtual classrooms, or hosting large-scale webinars, a scalable WebRTC broadcast solution demands frontend, backend, DevOps, and real-time UX expertise.
#4: Security and Privacy Are Critical in Real-Time Apps
WebRTC includes strong security foundations, like mandatory encryption on the video, audio, and data channels. But for industries like telehealth, fintech, and insurance where privacy and compliance are non-negotiable, more WebRTC security is required.
End-to-end encryption (E2EE) may be needed in some cases. In others, you’ll need detailed audit logging, secure media recording, or HIPAA and SOC2 compliance. Each layer of protection adds complexity to your application design and infrastructure.
#5: WebRTC Is Often Just One Part of the Puzzle
Most real-world WebRTC applications involve far more than just peer-to-peer video. You may need to integrate with SIP telephony systems for field service support, or add interpreters for accessibility.
Advanced applications often include machine learning for real-time transcription and summarization, or LLM-based assistants that take notes or even participate in the conversation. These AI-powered communication features dramatically enhance your app, but also multiply the architectural complexity.
#6: Evolving WebRTC Architecture: Media over QUIC, HTTP/3, and Beyond
Real-time communication apps are increasingly built with enhanced WebRTC architecture, including new protocols like Media over QUIC (MoQ), HTTP/3, and advanced browser APIs such as WebTransport, WebCodecs, and WebAssembly.
These technologies offer improved performance and scalability for high-quality streaming but add new interoperability and maintenance challenges. Navigating these choices requires deep expertise to ensure optimal results for global and enterprise-grade deployments.
Turning Complexity into Competitive Advantage
The complexity of modern WebRTC development isn’t a drawback. It’s an opportunity. When managed well, it becomes a strategic differentiator, allowing teams to deliver ultra-reliable, scalable, and secure real-time communication experiences that stand out in a competitive market.
Whether you’re launching an MVP, preparing to scale, or integrating advanced AI features or industry-specific compliance, the right expertise is critical. WebRTC.ventures brings more than a decade of hands-on experience to every project. Our team can help you design, build, and operate high-quality WebRTC applications every step of the way.
Contact our experts today to learn how you can turn WebRTC complexity into your competitive advantage.
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