WebRTC applications are often initially built to handle a minimal number of users. Later, as an app proves its usefulness in the marketplace and becomes successful, it is necessary to increase the number of WebRTC connections it can handle. This type of application scaling is an important and regular part of our work here at WebRTC.ventures. Often we can refactor the existing application. Other times, it becomes necessary to apply a new architecture entirely. In

“The mind is like a parachute: it only works if it opens.”Thomas Dewar Quality Assurance Software Testers are subject matter specialists. Their work focuses on the control, verification, and validation of products and software. They are responsible for identifying and reporting issues and faults prior to the deployment of technological tools and applications. They also make sure the final product is what the client is expecting to be.  The importance of QA software testing Here

On April 20, 2022, Arin welcomed Dr. Karl Stolley for a progress report on WebRTC browser implementation.  Karl is an Associate Professor of Information Technology & Management Illinois Institute of Technology where he teaches undergraduate and graduate courses on a range of web development topics, including one on WebRTC. He is also the author of Programming WebRTC: Build Real-Time Streaming Applications for the Web, currently in beta release with Pragmatic Programmers.  Karl began with a

Private WebRTC endpoints are often shielded from the public internet by a network that does the mapping from source to the destination. The mapping facilitates traffic going in or out of the private systems to reach the correct host. This process is called Network Address Translation, or NAT. While not necessarily restrictive, these interfaces need to be pin-holed to create a public-private pair for media streams to flow from the outside world into the WebRTC

Users today expect seamless, omnichannel communication across their devices. This is the promise of Unified Communications. UC allows a user to switch not only from computer to tablet to mobile within a single session, but also from one mode of communication (voice, video, messaging, etc.) to another. Live video, our speciality here at WebRTC.ventures, is an important component of the UC suite. But it is just one tool in a large toolbox. As we build

There’s a new type of WebRTC application architecture evolving right now referred to as “WebRTC Unbundling”. Although it may not be appropriate for all applications, it should at the least be considered for any new live video development project. It was an important topic in my recent WebRTC Live interview with Tsahi Levent-Levi of BlogGeek.me. In the past, we’ve spoken of three different types of WebRTC application architectures: WebRTC “to the standard”, open source media servers,

Subspace WebRTC-CDN not only delivers STUN and TURN based interconnectivity establishments, but also improves call quality performance for all WebRTC media traffic. This can help real-time communication systems achieve low latency streaming. Such low latency WebRTC streaming is useful for high interactive use cases ranging from online gaming with participant video conferencing to high-quality music streaming/podcasts.  Figure depicting WebRTC media traffic via Subspace WebRTC-CDN service relay in presence of a restrictive firewall. Since WebRTC endpoints

If you are loading a webpage, a delay of 3 to 5 seconds for static content may be acceptable. Further delays for dynamic content from 10 seconds up to 30 seconds may be seen. Sure, the user may get a bit impatient, but it is not a game changer. In live video, however, relay time can cause latencies that can make or break the user experience, and literally tank an application.  The Complicator: HTTP Live

On March 9, 2022, Arin welcomed Tsahi Levent-Levi to talk about how new technologies are going to affect the design and the development of WebRTC applications in the future.  After a brief update on the acquisition of testRTC by Spearline, Tsahi jumped right in with the question of whether WebRTC has reached its peak after 10 years.  Tsahi noted a few areas where WebRTC had a 4x increase in use: Work from homeRemote workHealthcareEducationNew use

What does it take to call yourself a WebRTC Developer? Or, if you’re looking to hire or contract a WebRTC Developer, what exactly does that mean? The term WebRTC Developer can actually be a pretty broad term. It’s best to know what you need before hiring anyone who uses that title. In my experience, there are five different types of WebRTC Developers: JavaScript CPaaS IntegratorMobile Video DeveloperOpen Source Media Server DeveloperDevOps ScalerWebRTC Protocol Engineer All of

Offscreen Canvas is an up and coming API that allows a developer to offload canvas rendering from the main JS thread in the browser. This allows a developer to provide high resolution or graphically intensive applications directly in the browser without affecting the user’s experience. Offscreen Canvas is still a little bit experimental, and is just now becoming standard in the major browsers like Chrome, Firefox and Edge, making this the perfect time to introduce

In my previous blog post, we saw how the SFU (single forwarding unit) saves on the network congestion and client side power consumption by using forwarding via a centralized server. While there are obvious advantages to SFU, it is worth noting how it performs with regard to energy efficiency and other factors in comparison to other established video conference topologies like Mesh, Mixed (MCU), and hybrid (MCU + streaming server). Mesh Mixer (MCU)Single Forwarding (SFU)Hybrid ( 

A Media Control Unit or MCU is the most established and time tested approach to setting up conferences via bridges. Conference bridges add centralized call and media features like mixing and quality control among other advanced controls such as a DID number, announcement, admin control, secure PIN-based access. Due to centralized media management, they are also ideal for connecting mixed streams with media pipelines for recording, broadcasting or plugging into machine learning models for transcription

For our 64th episode of WebRTC Live, Arin welcomed back Lorenzo Miniero, Chairman of real-time communications experts Meetecho and Founder of the Janus WebRTC media server to discuss hybrid events, a growing and fascinating use case for WebRTC. While the topic of the day was Meetecho's experience and lessons learned in this journey there and back again to a more intensive hybrid meeting experience, the episode started with a nod to the upcoming 8th birthday

If you’re ready to build a WebRTC based live video application, then the most important architectural decision you need to make is what media server to use. A purely Peer-to-Peer (P2P) WebRTC video call does not require a media server. All video and audio media is transferred between the peers, and all your application has to do is establish the P2P connection using a process known as signaling.  However, the vast majority of WebRTC applications

After a successful first outing last year, we again invited a few members of our expert team for a roundtable to share a piece of WebRTC wisdom with our viewers. Who better to learn from than those who work with WebRTC each and every day? Alfred Gonzalez discussed refactoring a WebRTC app to scale. Increasing the number of user connections that an application can handle is an important part of our work. Alfred reviewed the

WebRTC was designed to be a peer to peer communication system. However, it gives rise to a complicated mesh system when the number of participants increases. A Selective Forwarding Unit (SFU) is an alternate topology for connecting through a centralized server to route outgoing media streams from one to many users.  Peer to Peer WebRTC Stream Multi-point Peer to Peer WebRTC Stream The multi-party peer to peer party configuration leads to enormous strain on a user

Have you experimented with building WebRTC apps and wished that you could get in on one of the hottest development paths out there in 2022? Now is your chance to get free expert training on WebRTC and to join our team of WebRTC development experts! A new industrial revolution With the pandemic, the business world has been forced to make a shift to remote work and more creative business models. This is not going to

Arnaud Phommasone is a talented mobile developer that is a valuable member of our WebRTC.ventures team. He recently wrote a tutorial to create a Flutter app using an unofficial Flutter package for interfacing with Twilio Video, that will allow users to host a call and be joined by multiple other users. Flutter is Google’s free, open-source UI toolkit for building applications for mobile, web, and desktop from one single codebase. The tutorial focuses on creating a

In large, multi-party calls there are often just one or two main speakers. The rest of the participants are simply listening with their mic muted. The calls could be updates for shareholders, sales teams, vendors, regional offices, or factories – the use cases are endless. However, also are the number of empty audio packets going back and forth. This can cause a waste of audio bandwidth for silent participants. Handling low bandwidth scenarios to ensure the