Let’s have a closer look at WebRTC and how it can impact on implementation of real-time communication platform.
WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs.
WebRTC also represents the latest evolution in the world of real-time communication. WebRTC reuses many of the protocols and standards that we have analyzed in the previous articles (Media transfer in Unified Communications – SDP Protocol, RTP, RTCP and Jitter Buffer) to create real-time communications between different devices. Other standards, such as TURN / ICE / STUN, have been also used by WebRTC. We will discuss these standards later in this article.
The complexity of using these standards is hidden behind a simple set of Javascript APIs, which are immediately available to developers via browsers.
WebRTC brings to life the best of standardized Real Time Communication Technologies, and, in most scenarios, allows real peer-to-peer communications between endpoints.
Continue reading “WebRTC as a reliable standard for Real-Time Communication Technologies”

Let’s examine which available solutions allow for the transmission of faxes over IP and the best practises that reduce related problems.
Desired video codecs mainly fall between VP8 (and its evolution VP9) and H.264 and H.265. Other formats, such as H261 – H263 – H263p, can be mainly found in legacy conferencing systems and should be avoided.
Wideband audio provides high-definition voice quality for telephony calls,