
Blog
Category: Enhanced WebRTC
Learn about the innovative ways developers used WebRTC at the 2016 AT&T Developer Summit Hackathon
The annual AT&T Developer Summit just keeps getting bigger and better every year, and this year was no exception. Over 1,000 developers in Las Vegas and Mexico working through the night to make their ideas a reality. The Rain nightclub in Las Vegas was teaming with inspiration. See a sampling of the apps that used WebRTC and ultimately find which team won the Best Use of the AT&T Enhanced WebRTC API prize.
Read more....How to send/receive calls in your iOS app using AT&T Enhanced WebRTC
The Enhanced WebRTC API enables you to integrate real-time audio/video calling into your iOS app. Here’s what you need to know if you’re integrating this feature into your app at the hackathon.
How to send/receive calls in your Android app using AT&T Enhanced WebRTC
The Enhanced WebRTC API enables you to integrate real-time audio/video calling into your AndroidTM app. Here’s what you need to know if you’re integrating this feature into your app at the hackathon.
Final Summit Hackathon challenges, themes, and prizes revealed
This year’s AT&T Summit Hackathon features more than 40 prizes that combine for over $300,000 in cash and prizes. We understand with so many themes, topics, and challenges it can be very difficult to come up with a plan onsite. Learn more about them before you arrive at the event.
Recap: AT&T Mobile App Hackathon in Berkeley
We had a great time in Berkeley, CA hacking at The Batchery. Keep reading to find out about the cools apps we built!
7 Simple ways to enhance the WebRTC user experience
If you’ve been following along with this WebRTC UX series, you now have some fundamentals under your belt. In this fifth and final post, we’ll look at ways to enhance the WebRTC experience, ensure user privacy and security, improve accessibility, and gather user feedback.
WebRTC series: 5 Crucial ways connectivity and bandwidth affect WebRTC UX
In the previous post in this WebRTC series, we looked at some useful UX patterns for voice and video calls to ensure things go smoothly for your users. Now, let’s talk about connectivity, bandwidth, and call quality.
New Chrome HTTPs requirements for voice and video calls: Enhanced WebRTC has you covered
You may have heard that Google is increasing its security requirement for WebRTC and will be requiring all WebRTC sessions to be run over HTTPS connections. With an estimated launch date of December 2015, your end users will no longer be able to share their voice and video.
WebRTC series: 4 Indispensable UX patterns to create transparent voice and video calls
Voice and video calls can range from enjoyable to frustrating and there are a lot of variables that contribute to making it a positive or negative experience. In this WebRTC series installment, we’ll look at the experience of the call itself and some reliable UX patterns.
3 Essential UX considerations when initiating or joining a call via WebRTC
In the previous post in this series, we looked at some UX considerations when adding WebRTC-enabled features to an application. Next, we’ll go over the initial major steps the user will take with WebRTC: initiating or joining a call.