WebRTC: Web Real-Time Communication

January 07 , 2022
What Is WebRTC?
WebRTC (Web Real-Time Communication)  is a free, open framework for the web that enables the real-time communication (RTC) its name promises to deliver. As a combination of standards, protocols, and JavaScript APIs, WebRTC leverages peer-to-peer connections between browsers to support a near-simultaneous exchange of data — without requiring third-party software or plug-ins.

In other words, WebRTC allows users to initiate click-to-start video chats from their browsers and exchange information quickly enough to replicate in-person interactions. This supports interactive live streaming between individuals, as well as browser-to-browser communication through a set of standard protocols.



With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others. This page is maintained by the Google WebRTC team.


The main audio codec of WebRTC is Opus. Opus is an audio coding format developed by the Xiph.Org Foundation, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication. Tonmind IP Speaker supports 48K Opus codec, which is not featured by other brand IP Speaker in market including 2N and Axis. Opus can reduce bandwidth to most extent while ensuring extremely high sound quality.



How Does WebRTC Work?
WebRTC employs three HTML5 APIs that allow users’ browsers to capture, encode, and transmit live streams between one another, enabling two-way communication. For this reason, WebRTC is referred to as peer-to-peer technology, whereby each browser communicates directly with one another.

The beauty of WebRTC lies therein: It eliminates the need for any intermediary web servers during these exchanges, not to mention additional equipment or software. URL-based meeting rooms are an excellent example of the convenience and real-time communication delivered by WebRTC.

While some streaming workflows require a live streaming camera, encoder, and media server, the simplest WebRTC deployments can accomplish everything with a connected webcam and browser. And unlike Flash-based video, WebRTC can be played back on any HTML5 player that supports WebRTC APIs.

However, because WebRTC was designed for native information exchange without an intermediary server, it can’t handle large audiences. Anyone looking to stream WebRTC at scale will require the help of a streaming server or service. From repackaging the content into a more scalable format to delivering live streams across a custom-built WebRTC content delivery network (CDN), Wowza has options for configuring your WebRTC workflow to accommodate audiences of up to a million viewers.

WebRTC Snapshot
Audio Codecs: Opus, iSAC, iLBC
Video Codecs: H.264, VP8, VP9
Playback Compatibility: Chrome, Firefox, and Safari support WebRTC without any plugin
Benefits: Super fast and browser-based
Drawbacks: Designed for video conferencing and not scale, thus requiring a streaming platform like Wowza when streaming to large audiences
Latency: Sub-500-millisecond delivery


With application of Opus, Tonmind Network Speaker transmits excellent sound quality. Opus Primarily starts with a combination of the SILK voice codec for Skype's early Internet calling and Xiph.org's CELT music codec. It's designed to transmit voice over the web and audio streams for VOIP, video conferencing, in-game chat and other applications, and is considered superior in quality to existing proprietary audio codecs. After many comparative tests, Opus beat the once superior HE AAC at low bitrate, and it is now match for AAC with about 30% higher bitrate, while high bitrate is closer to raw audio encoding.


Apart from Tonmind Network speaker, Tonmind PA System also supports OPUS, which enables as less sound quality sound during network transmission. Tonmind PA System is audio software with built-in SIP server. It can play various audio sources from SIP call, live radio, local media player, universal windows media player (for example, Spotify, iTunes, VLC, ect.). It also supports SIP Call. User can control zones, contents, rings, volume and scheduling, which can be widely used in School PA System, Commercial PA System, Hospital PA System, Hotel PA System, etc.


WebRTC Benefits

When considering the many advantages that WebRTC delivers to both users and developers, it makes sense why there’s so much hype surrounding it. Everything from low-latency delivery to interoperability makes it an attractive choice.
Inherently Low Latency.  WebRTC knocks it out of the park when it comes to delivery speed. At sub-500-millisecond glass-to-glass latency, WebRTC offers the quickest method for transporting video across the internet.
Platform and Device Independence. All major browsers and devices support WebRTC, making it simple to integrate into a wide range of apps without dedicated infrastructure.
Open Source and Standardized.  The open-source framework is standardized by the IETF and W3C, thereby eliminating any interoperability challenges that come with proprietary streaming technologies.
Adapts to Network Conditions. WebRTC ensures reliable publishing over poor network conditions with adaptive network encoding.

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