The WebRTC.ventures Blog.

Many WebRTC applications struggle with outdated or inappropriate media server infrastructure, limiting their ability to scale effectively and support powerful AI features. Alfred Gonzalez, Senior WebRTC Engineer at WebRTC.ventures, walks us through the considerations, options, and steps to successfully migrate to another media server. He’ll then show how this rearchitecting opens the door to implement AI-powered voice assistants using STT, TTS, and LLM services to intelligently handle inbound customer calls. Alfred will also demonstrate how

When Sam Altman called GPT‑5 “a PhD in every discipline in your pocket,” it captured the awe surrounding modern large language models. As builders, we should be thrilled. This is an extraordinary leap in what’s technically possible. But here’s my unpopular opinion: just because we can use the most massive LLM for every task doesn’t mean we should.  In customer service AI, bigger isn’t always better. The real challenge is architectural—matching each task with the

Digital media delivery today relies on two powerful but separate approaches: scalable streaming and real-time conversation. Each one excels at a specific purpose, and understanding both reveals why MOQ represents a fundamental shift in how we may be building these applications in the future. Let's explore what makes the current landscape so divided, how MOQ aims to deliver a latency-tunable approach, and why this matters for anyone building media applications today. Current Media Delivery: HLS/DASH

Last year, I attended the RTC.ON conference organized by Software Mansion for the first time. I shared my take on the conference in this WebRTC.ventures blog post: A WebRTC Developer’s Take on RTC.ON 2024. I also spoke at the 2024 conference, a recording of that talk is available on YouTube: Boosting Inclusivity: Closed Captioning & Translations in WebRTC. I loved the conference and was excited to come again. This time I was only a participant,

Voicebot latency is the most critical performance metric for voice-enabled Conversational AI systems. While text-based interactions can tolerate response delays of several seconds, voice agents must respond as quickly as possible to maintain natural dialogue flow. Even slight delays create slow voicebots with perceptible awkwardness that degrades user experience and erodes trust in the system. The challenge stems from Conversational Voice AI's inherent architectural complexity. Voice-enabled agents often coordinate multiple AI technologies—speech recognition, language processing,

Earlier this week, I had the privilege to travel back to my alma mater, the Illinois Institute of Technology in Chicago, to speak at and attend the IEEE RTC Conference & Expo 2025, which covered WebRTC, Mobility, VoIP, and Next Generation 911. This annual event continues to be one of the leading gatherings for innovators, researchers, and practitioners shaping the future of real-time communication systems. It was a great opportunity to reconnect with colleagues across

WebRTC’s code may be open source and royalty-free, but deploying a reliable, production-grade real-time communications application involves both up-front development investment (engineering, integration, and setup) and ongoing operational spending (infrastructure, bandwidth, monitoring, and scaling). Teams who overlook either side risk major surprises as their product grows. So how much does building and running a WebRTC application cost? The answer varies widely depending on your approach and your usage. Building your own solution (DIY) versus using

Voice AI applications need real-time and reliable audio communication for natural conversations with AI customer service bots, virtual assistants, IVR platforms, and other voice-enabled systems. Choosing the appropriate transport protocol is crucial for teams, as using the wrong one can lead to choppy audio, noticeable delays, and dropped connections.  To support realistic conversational AI, the transport protocol must handle variable network conditions, secure media transport, and deliver high-quality audio with minimal latency. In this post,

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

Last week I attended the RTC.ON 2025 conference in Krakow, Poland, alongside my colleagues Alberto González Trastoy and Alfred González Trastoy from our WebRTC.ventures team. The conference is in its third year, and this was my second time attending. RTC.ON has quickly become one of the premier gatherings for video and voice developers. It’s always a fantastic opportunity to learn, share ideas, and connect with peers who are pushing the boundaries of real-time communication. This

There's one important catch to telehealth: not everyone has reliable internet access, and connectivity can fail at any point during remote medical appointments, even in regions with strong coverage. Telephony integration using traditional voice networks remains a vital backup solution when digital connections drop during virtual healthcare visits. This technology connects conventional phone systems with modern telehealth platforms, ensuring appointments can be completed whether patients are using smartphones with spotty data coverage or landlines in

Host Arin Sime was live from the RTC.ON 2025 conference in Krakow, Poland, with short discussions with three of the event speakers: Will Law, Chief Architect / Akamai. Conference presentation: A QUIC update on MOQ and WebTransport Cezary Siwek, Staff Engineer / Stream. Conference presentation: Challenges in Realtime livestreaming at 4k / 60FPS Christoph Guttandin, Developer / Media Codings. Conference presentation: Designing a media container library for the web Read our conference wrap up: WebRTC.ventures Visits RTC.ON

Enterprise video SDK integrations frequently expose critical security vulnerabilities. Client-side secrets, unvalidated webhooks, and bypassed authentication layers create attack vectors that persist in production systems. These implementations typically function correctly during development, but fail security audits and present significant risks in enterprise environments. We've built a real-time video platform architecture that addresses these fundamental security gaps. The implementation uses the Zoom Meeting SDK as the video foundation, chosen for its comprehensive enterprise feature set, embedded

If you're in the RTC community, you know that WebRTC.ventures excels at assessing, building, testing, and managing real-time communication applications. But what happens when your success leads to bigger ambitions? When you need full engineering teams, not just project-based development? When you're ready to scale beyond individual applications to building comprehensive development capabilities? This is where AgilityFeat, our parent company, comes in. From WebRTC Success to Strategic Scaling You’ve reached a pivotal moment. Your application

At WebRTC.ventures, quality testing is at the heart of everything we do. Robust API testing is the foundation of reliable performance, especially for WebRTC applications and AI integrations which depend on real-time performance and consistent connectivity. In our QA testing lab here in Panama City, Panama, we simulate real-world scenarios and network conditions to ensure APIs don't just work, but perform reliably under pressure. That's why Postman is one of our favorite tools in the

My name is Octavio Alvarado and I study Software Engineering at the Technological University of Panama, currently in my fifth year of my degree. I was part of the UTP's Professional Internship program, which allows students to join companies according to their skills and profiles. Through this program, I had the opportunity to work with the Quality Assurance and WebRTC testing team at WebRTC.ventures, an experience that has been very enriching for my professional development.

WebRTC VoIP systems enable voice calling directly through web browsers and mobile apps without requiring any software downloads or plugins. This makes them ideal for customer support platforms, telehealth consultations, sales calls, and any application where you want to add voice communication without asking people to install additional software. These systems are more flexible than traditional phone systems, but they also create operational challenges for DevOps teams. Instead of managing hardware-based PBX systems, you're now

Voice AI agents have unique deployment needs. Operational complexity multiplies quickly. You're not just deploying code; you're orchestrating real-time audio pipelines that need to maintain call quality under load, coordinate between AI services that each have their own scaling characteristics, and handle the networking complexities of audio delivery across diverse client environments. When choosing a deployment strategy, you'll find three main options, each making different trade-offs: Managed services handle all infrastructure but limit customization  Serverless

If you've worked with WebRTC applications long enough, you've probably heard these complaints:  "The video froze, but I could still hear them."  "The audio kept cutting in and out."  "The call just dropped for no reason." You check your application performance monitoring (APM) dashboard and everything looks normal. Server CPU is steady, error logs are quiet, infrastructure metrics are green. So what happened? Traditional monitoring tools were built for backend systems, not for catching the

If you’re looking to integrate Zoom into your web application, you have two main options: the Meeting SDK and the Video SDK. While both run in the browser, support popular frameworks like React, Vue, and Angular, and use JWT for authentication, they’re designed for very different purposes. Understanding the differences will help you decide which SDK fits your project best.  In this post, we will take a look at how they work and break down