RTC:engine – WebRTC TelecCommunication API

Enabling Real-Time Communication Services with RTC:engine

RTC:egine is a WebRTC-based Communication API. It is lean and simple that it takes away the complexity of real-time communication systems, enabling rapid application development based on the RTC:engine interface. This allows you as operators and over-the-top service providers to offer instant real-time communication services in a web browser. Your end customers can enjoy the benefits of having access to cloud communications, such as live video, voice, messaging and conferencing services on websites and mobile apps without installing any plugins or software.

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RTC:engine also manages the signalling between the connected clients and is able to expand the communication to external VoIP devices using protocols like SIP or XMPP or even to PSTN networks.

With Sipwise RTC:engine, providing real time communication services has never been so easy.

RTC:engine Features

Voice and Video

A simple set of API calls is sufficient to enable secure, peer-to-peer or anchored voice and video streams between web browsers without requiring the developer to know about the underlying signalling and media details. Both one-to-one and multi-party sessions can be established by means of a few lines of Javascript code.
Messaging

Exchange messages within chats and chat rooms in a protocol-agnostic way allow to connect users over different networks using XMPP with GSM-SMS fallback.
Presence

Exposing the presence status of users via API callbacks enables customers to implement buddy lists into their web application, making it easy for users to determine the availability of their communication partners.
Transcoding

Media streams can be delivered via narrow-band, ISDN and high definition quality codecs, depending on the access networks. For example, WiFi enabled users can communicate in full HD, whereas participants on 3G with restricted band will fall back to narrow-band communication.