WebRTC is an open-source application programming interface (API) first introduced by Google in 2011. The goal of Google for WebRTC is to deliver a standard-based, real-time media engine that will be free and resident in all available browsers.
Platform and Device Independence
Any WebRTC-enabled browser with any operating system and a web services application can direct the browser to create a real-time voice or video connection to another WebRTC device or to a WebRTC media server. Developers can write HTML5 code that can work on desktop and mobile devices.
Secure Voice and Video
WebRTC has always-on voice and video encryption. The Secure RTP protocol (SRTP) is used for encryption and authentication of both voice and video. This is especially beneficial over WiFi networks. This prevents eavesdropping and recording of the voice and video.
Advanced Voice and Video Quality
WebRTC uses the Opus audio codec that produces high fidelity voice. The Opus codec is based on Skype’s SILK codec technology. The VP8 codec is used for video. These selections ensure interoperability and avoid the need for codec downloads that may contain malicious code.
Reliable Session Establishment
WebRTC supports reliable session establishment. This is true for Network Address Translators (NAT), something that hinders and may block other communications and collaboration protocols. The reliable operation avoids server-relayed media and thereby reduces latency and increases quality. It also reduces the server load.
Multiple Media Streams
WebRTC is an adaptive network solution that compensates and adjusts to changing network conditions. It adjusts the communications quality, responds to bandwidth availability, detecting and avoiding congestion. This is accomplished using the multiplexed RTP Control Protocol (RTCP) and Secure Audio Video Profile with Feedback (SAVPF).
Adaptive to Network Conditions
WebRTC supports the negotiation of multiple media types and endpoints. This produces an efficient use of bandwidth delivering the best possible voice and video communications. The APIs and signaling can negotiate the size and format for each endpoint individually.
Interoperability with VoIP and Video
The biggest value of WebRTC is its promise of interoperability with existing voice and video systems. This includes devices using SIP, Jingle, XMPP, and the PSTN. What may hinder the global interoperability will be the upgrades necessary in exiting devices.
Rapid Application Development
Detailed knowledge of WebRTC will not be necessary because of the standardized APIs. Finally, the voice and video codecs are license-free.