Web Application Developers are used to being able to write automated tests for their applications and have them run with every PR and before deploying to production to give a level of confidence that things are not broken. OpenTok and real-time applications in general present new challenges when it comes to writing and running automated tests. There are challenges when it comes to getting access to microphones and cameras, testing multiple participants and installing the plugin for Internet Explorer among others.
There has been lots of work around WebRTC testing automation and our friends at rtc.io and &yet have written some great articles on the subject. However these articles don’t cover some of the specifics of testing OpenTok applications for example testing Internet Explorer and installing the OpenTok plugin for Internet Explorer. If you haven’t already I would recommend taking some time to read the articles by the folks at rtc.io and &yet before coming back to this. Also if you’re not familiar with Travis and Selenium WebDriver you might want to check those out too.
We all have a fascination with the billion dollar startups. Venture Capitalists try and identify them early, media laud them (or bring them down to earth), and early adopters claim discovery. One new technology innovation has the potential to spark the creation of more billion dollar companies, and markets are starting to pay attention. So what is WebRTC, and why is there so much interest?
It begins with recognizing the emergence of two massive trends. The first is the increasing appetite for ‘on demand’. This is evident in everything from movies to car rides, hotels, relationships to groceries to well, everything. And communications is a core part of this, just look at Meerkat and Twitter’s latest acquisition, Periscope, bringing
We’re excited to announce the release of the OpenTok One-to-One Sample Application across web, iOS and Android. This open-source application enables you to speed up your development efforts to set up interoperable, production-quality audio/video communication between users.
As you get started with this OpenTok sample, you will learn the best practices used to develop and manage the audio, video, and camera elements on mobile devices or in the browser. We recommend this is as your first step in delivering Real Time Communications (WebRTC) solutions on the OpenTok platform.
In today’s hyper-connected world, individuals are increasingly looking to online services and solutions to give them more flexibility in their day-to-day life. Industries across the board are now operating online to meet the needs of today’s consumers and make their services more accessible – from e-commerce through to banking.
The healthcare and wellness industry is no exception. Patients can already visit the doctor virtually through their smartphone, or communicate with a specialist on the other side of the world without leaving the comfort of their own home.
But one area which has traditionally relied on physical presence has been exercise. You have to go the gym to get fit, right? Not any longer. This status quo is being disrupted by technologies such as WebRTC and embedded communications, where real-time video makes working out from home not only possible, but also personalized, effective and enjoyable.
When developing applications, the importance of mobile cannot be underestimated and, at TokBox, we recognize the need to communicate seamlessly between desktop and mobile devices. That’s why we were excited to attend Mobile World Congress last month and demonstrate the power of WebRTC in bringing contextual, embedded communications to a range of uses cases, across multiple devices.
Mobile World Congress this year surpassed a record 100,000 visitors from around the world. With a range of exhibitors and presentations from industry leaders including Facebook’s Zuckerburg and Cesar Alierta from Telefónica, the event showcased the latest technology and trends in the mobile world, from WebRTC to virtual reality and robotics.
Whether you’re developing a new website, building an app for mobile or working out web infrastructure, it’s important to keep up to date with all of the technologies contributing to the evolution of the web. The O’Reilly Fluent Conference aims to to help you do that. With a range of speakers across a number of different roles and industries Fluent covers the full scope of the Web Platform and its associated technologies.
As the WebRTC landscape continues to evolve it can be hard for developers to keep up. The Kranky Geek WebRTC event aims to fill in the gaps and jumpstart your knowledge about WebRTC and the ever-changing landscape of communications online.
The event has a jam-packed agenda with experts talking about a range of different topics from the very basics to real world applications, and building workshops. TokBox CTO, Badri Rajasekar, will be there to talk about the need to push the boundaries of WebRTC in order to cater to unprecedented broadcast use cases.
This post was co-authored by Gustavo Garcia Bernardo, Philipp Hancke and Charley Robinson.
When WebRTC stuff is really broken, it gets fixed very quickly.
Early in December 2015, shortly after the release of Chrome 47 to the general public, we started to notice a subtle and strange behavior in the Audio/Video of streams during our many daily meetings using WebRTC: the video occasionally wouldn’t stay caught up with the corresponding audio. As with many bugs noticed internally by developers, it took a while for any of us to believe that what we were seeing was a real issue. We call this the inverse of productive dogfooding: rather than assume we are just like our users, we can just as easily decide we are nothing like them.
Have you ever had to support a WebRTC application and needed to get packet dumps from the user? Wireshark is a great tool for this, but asking a user to install it and make a dump rarely works. It’s just too complicated. So I was pretty excited when I read the Chrome 49 release notes which described (not in much detail) a new feature called the ‘RTC event log’. This is described as follows:
We now provide a new debug option in chrome://webrtc-internals for tracing internal details (e.g., BWE, jitter buffer state) for audio and video sessions. This option creates a log containing the timing and headers of packets as well as the timing of various internal events. We hope this will help resolve issues related to media transport and jitter buffers; attaching this log when reporting such issues will help us tremendously.
Early December saw the roll-out of Chrome 47. When doing anything with WebRTC, this is always an interesting time. A release brings new features or may break things, like removing the getUserMedia functionality for insecure origins.
Our metrics clearly track such roll outs as seen below: