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Product type: Desktop software

WebRTC is a web based Real Time Communication platform that will allow different browsers to connect with each other for chat, audio & video exchange, sharing files, etc. without the plugins for e.g. Gtalk, Skype etc, however you do need a compatible browser to access it.

It’s a new collection of standards for RTC that enable setting up video & audio communications over the Web. It is an open source project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs. The WebRTC components have been optimized to best serve this purpose. WebRTC implements the W3C's proposal for video conferencing on the web. In summary WebRTC can do what Skype, Google talk do but more efficiently without plugins.

The core of WebRTC is the getUserMedia JavaScript API, which gives the browser access to hardware features like the camera and microphone.


Audio: The WebRTC project offers acoustic echo cancellation (AEC), automatic gain control (AGC), noise reduction, noise suppression and hardware access and control across multiple platforms.

Video: It includes components to conceal packet loss, clean up noisy images as well as capture and playback capabilities across multiple platforms.

Network: Mitigates the effects of packet loss and unreliable networks. Also included are components for establishing a Peer to Peer connection.

Things needed for getting WebRTC up & running:

1.      Some source of streaming the video, audio or data

2.      Communicate the video

3.      Exchange some messages to initiate & close sessions

4.      Exchange media & configuration information like resolution, format information between peers

5.      Way for people to get in touch with each other


Needs a codec such as Opus- MTI status so it can spread widely

Google wants VP8 to become MTI for WEBRTC

Mozilla supports Opus for Web RTC in Firefox


Media stream – to get streaming data

Peer connection – to communicate data

Data channel – to exchange streaming data -DataChannels offer a way to send data from one WebRTC-enabled browser to another. DataChannels can send pretty much any data the browser can access, be it images, videos, webpages or local files


User discovery - Communication signalling with clients

NatTraversal and streaming data communication

Technical Specifications
Hardware standards:
  • Other
  • Currently supported voice codecs are G.711, G.722, iLBC, and iSAC, and VP8 is the supported video codec. The list of supported codecs may change in the future.