In this article we demonstrate 7 technologically different ways to display a video stream from an IP camera with RTSP support on a web page in a browser.

As a rule, browsers do not support RTSP, so the video stream is converted for a browser using an intermediate server.

It’s been a while since HTTP-based video content delivery protocols such as HLS and DASH edged out Flash as a primary way to play online video content in browsers.

Nevertheless, the RTMP protocol initially intended for Flash is still one of the most popular ways to transfer Live video from the source of the video to the rebroadcasting server. RTMP is still used by such services as Facebook Live, Youtube Live, and others where live video broadcasting is a must. Therefore, in spite of predictions and prophecies of untimely death of Flash, the RTMP protocol is still in the catbird seat and holds the fort in the video broadcasting niche.

Flashphoner.createSession(...).createCall({callee:'+74957718000'}).call();

In this article we will explain how to create a button to make a call to any mobile or stationary phone number directly from a web page.

Evil empires are often a target of grudge and hate from end users. This is understandable and sometimes even justified, but nevertheless Uber still partially pays for our trips (even though it’s temporarily) and Google accelerated the WebRTC technology, which otherwise would stay a proprietary and expensive software tool for quite a narrow b2b purposes if not for the Evil Empire of Google.

Since WebRTC has been made public, video chats became much easier to develop. A number of API and services, servers and framework has emerged for that. In this article we thoroughly describe one more way to develop a video chat between a web browser and a native Android application.

In this article we will try to summarize everything we know about latency of online broadcasts from the web camera of a browser: what causes the latency, how to avoid it, and how to make an online broadcasting truly real time.

Then, we will show what happens to latency through the example of a WebRTC implementation, and how using WebRTC allows keeping latency comfortably low.

Both Facebook and YouTube offer broadcasting capabilities that allow live broadcasting of a video stream to wide audience of spectators. In this article we will tell how to capture a video stream from the web page using the WebRTC technology and send this video stream to Facebook and YouTube simultaneously, that is run live broadcasting in both services at once.

Before we begin broadcasting, let’s find out what RTMP endpoints Facebook and YouTube offer for streaming. Then, we need to send video streams to these RTMP addresses.

April 2016 brought a surprising press-release that Apple rolls support for WebRTC in Safari browsers for Mac OS and iOS. It’s been a year since the press-release, and Apple is still rolling WebRTC to Safari. Stay tuned.

Recently we wrote about how to share a screen in the Google Chrome browser. As a result, we created our own screensharing extension, published it on Chrome Store and tested screen broadcasting using Web Call Server in the one-to-many mode.

In this article, we will try the same using the Firefox browser. The approach is the same, and again we need to pack and publish the extension, but this time we will publish it on Mozilla Add-ons. As a result, we will be able to make screencasts of video streams from FF without installing any third-party software.

In this article we describe how we build our first WebRTC application for iOS, how we published it in App Store, what we faced in the process and how things turned out in the end.

Let’s implement a three-way video chat on the following platforms: desktop Google Chrome, an Android application on a tablet, and an iOS application for Apple iPhone.