Adding Recording to Your WebRTC Application
Building recording into your WebRTC video or audio application should never be an afterthought. When clients come to us at WebRTC.ventures looking to develop a WebRTC video or audio application, one of the first questions we ask is, “do you need recording?” If the answer is yes, the next question is always, “what for?” And then, “how much recording?” It is essential to think about these things up front. The answers will be an important
Notifications Can Make or Break Your WebRTC User Experience
There are a number of roadblocks that can prevent users in a WebRTC video, audio, or chat application from connecting. Making this connection is the make-or-break moment in terms of providing a quality user experience. Adequate and timely notifications provide essential feedback to mitigate these issues. User experience (UX) design is the process design teams use to create products that provide meaningful and relevant experiences to users. Delivering quality experiences within your application is all
Integrating Telephony into Your WebRTC Application
How can you add dial-in or dial-out capability to your WebRTC video application? What are the use cases where you would want to connect a traditional phone call on a public switched telephone network (PSTN) into your WebRTC video and/or audio conference? Let’s explore integrating telephony into WebRTC. You can watch this and other tips from our WebRTC.ventures engineering team as part of our WebRTC Tips YouTube video series. Or, read on below. Use Cases
WebRTC Assessments and Consulting
Our team has built countless WebRTC applications over the years. We’ve integrated live video and audio for a variety of industries, use cases, and using different WebRTC communication platforms and libraries. We work closely with the teams at Vonage, Agora, Twilio, Janus, Jitsi (8x8) and others. Many take advantage of this deep expertise by contracting with us for our WebRTC development services or for our WebRTC testing services. However, another very popular request we get
Watch WebRTC Live #51: NV – The Next Version of the WebRTC Standard with Bernard Aboba
Now that WebRTC 1.0 is here, what comes next? On February 17, 2021, WebRTC.ventures produced our 51st episode of WebRTC Live. Arin was joined by Microsoft Principal Architect and W3C WebRTC Co-Chair, Bernard Aboba. They discussed what is coming down the pipe in WebRTC-NV. This content built upon the conversation Arin had last month with Tsahi Levent-Levi on WebRTC Live Episode 50: WebRTC Trends for 2021. And also on this in-depth Q&A that Bernard did
Vonage Communication APIs
Last week, we talked about WebRTC application architecture and the trade-offs in choosing Native, Open Source, or CPaaS solutions. Many who go with a Commercial Platform as a Service (CPaaS) option will choose Vonage. Vonage is one of the commercial platforms that can be used to integrate WebRTC video, as well as other functionality, into your application. As custom developers specializing in live video integration, we at WebRTC.ventures often use CPaaS solutions and Vonage is
WebRTC 1.0 is here!
After first being introduced by Google way back in 2011, WebRTC 1.0 has finally arrived! It was officially made a W3C Recommendation on January 26, 2021. A lot has happened in the ten years since the WebRTC standard for HTML5 was first introduced. WebRTC has advanced from its original vision WebRTC was originally envisioned as a way for javascript to easily access the camera and microphone of browser users. With their permission, of course. Then,
WebRTC Cloud Gaming: Unboxing Stadia
While the gaming industry has been around for decades, its popularity has grown tremendously in recent years. Gaming is expected to grow at a yearly rate of 12.9% from 2020 to 2027. eSports have become even a full time job for some. And as old technology challenges are overcome, companies like Google and Microsoft are pushing toward building games in the cloud. Cloud gaming is the future of seamless entertainment Although it requires a high-speed
Watch WebRTC Live #50: WebRTC Trends for 2021 with Tsahi Levent-Levi
On January 27, 2021, WebRTC.ventures produced our 50th episode of WebRTC Live. Arin's guest was Tsahi Levent-Levi to discuss what's next in WebRTC development in 2021. Tsahi is a leading authority in our community, keeping us up to speed on the world of communications, WebRTC, and CPaaS. He blogs at bloggeek.me. Both of the presenters have blogged about this topic. You can read those posts here and here. The topic was timely as just the day prior, The Internet
Native, Open Source, or CPaaS?
A very common and very important discussion we have with our clients before we build their custom WebRTC application is about architecture. Should it be a native WeRTC application, built on an open source media server, or use a commercial CPaaS platform? Let’s talk about each one, their benefits and drawbacks, and what factors play into this important decision. This content is also available in our WebRTC Tips by WebRTC.ventures YouTube series. WebRTC is not
Things I Wish I Knew Before Building My First WebRTC App
Naive me in 2015: Thinking WebRTC is just another browser API. Experienced me after more than five years and dozens of WebRTC applications: WebRTC based applications are different from other web applications with a unique set of challenges. But also, solutions, if you have a little experience behind you. In this post, I will review some of the things I wish I had known back then. But first, some WebRTC basics. WebRTC is: an open
Layers of Testing for WebRTC Applications
The only thing harder than developing a WebRTC application is testing it. When our team at WebRTC.ventures started building live video applications five years ago, most of the work we did for clients and ourselves were essentially prototypes. The success metric was an alpha or beta phase with potential customers, or simply “demo ware” that could be shown to potential investors in order to secure the funding to build the full-scale application. My, how times have changed!
What is WebRTC?
Let’s talk about the very basics of what WebRTC is and when and why we might use it. WebRTC: Is an HTLM5 standardUses JavascriptAllows you to control the camera and microphone in the browser with no downloadsIs encrypted and secure In other words, WebRTC allows you to set up a live peer-to-peer connection between two different browsers - or more - to exchange private video, audio, and data between them. WebRTC is a form of
WebRTC Signaling Servers – STUN vs TURN
STUN and TURN servers are two types of WebRTC signaling servers that can be used to create your peer-to-peer (P2P) connection when you are building a real-time communication application. Signaling is not part of the WebRTC standard and thus needs to be handled separately if you are building the application yourself rather than using a commercial platform (more on this later!) If you are a visual learner, we have embedded a WebRTC Tips by WebRTC.ventures
Watch WebRTC Live #49: Debugging WebRTC Applications
On December 9, 2020, WebRTC.ventures produced our 49th episode of WebRTC Live. Arin's guest was our own Senior WebRTC Engineer and Developer Evangelist, Germán Goldenstein, to discuss tools and methodologies for debugging WebRTC applications. Germán has been working in the software development industry since 2002 and with WebRTC.ventures since 2015. He is one of the core developers of Interactive Broadcast Solution, a TokBox (now Vonage) product. “I had a great connection, but the video quality
Trends for WebRTC Development in 2021
2020 has been a devastating year globally. The COVID 19 pandemic has affected nearly every aspect of our lives. There’s no doubt it will continue to affect our lives throughout 2021. Sadly, we will see many more lives lost. Long lasting scars - and healing Traumatic events such as this have a way of leaving scars on society for a long time, sometimes as long as the lifetime of the generations most affected. For example, my
IIT Real Time Communications 2020: WebRTC Track Talks
For the second year running, our CEO and Founder, Arin Sime, curated the WebRTC track of the Real Time Communications Conference at the Illinois Institute of Technology. We were also proud to be a sponsor. The IIT RTC annual conference is a globally recognized collaborative event where industry and academia connect. It brings together software developers, network engineers, entrepreneurs, business executives, students, and academic educators and researchers to promote an open exchange of ideas to lead future
Pre-Call Tests in Live Video Apps
What’s the most important thing that you can do to make your live video application successful? Anything you can do to improve call quality! This will certainly make your users more satisfied with your app and therefore make you more successful. However, optimization can only get us so far. This is because we can’t always control call quality, no matter how much tweaking we do on our media servers. And the biggest factor in call quality is
A Guide to the Amazon Chime SDK
The pandemic has forced the big players in cloud computing to offer communication services, of which arguably the biggest is Amazon with their Amazon Web Services platform. Now AWS is offering SDK’s and API’s for their pre-existing Amazon Chime video conferencing application. This will allow customers to easily build their own real-time communication apps with Amazon's infrastructure as the backbone. In this post, we will go through setting up a simple videoconferencing app with the
Building a Video Conferencing App Using Janus WebRTC Media Server
For basic one to one video communications on the web, we could just use WebRTC and connect peers directly. However, when we want advanced functionalities (ie, recording or transcoding), or if we have more participants, we will need the help of a media server. A WebRTC media server essentially is a server optimized to more efficiently receive and send media. There are many great open source WebRTC media servers out there. (We mentioned some of them
