General Terms
SC – Simultaneous calls. One browser client in the ‘talking’ state. Browser is receiving and/or sending an audio and/or video streams.
For messaging features the license allows SC*5 active text chatters.
For WebRTC live video and broadcasting features, the license allows SC*10 concurrent connections.
STARTER, ADVANCED, ENTERPRISE-1, ENTERPRISE-2, ULTIMATE – licensing plans.
‘Pre-ordering’ – some features are disabled and not currently available for selection and order. These are pre-ordered features. Please, contact us to agree upon an order of such software features.
Audio and Signaling
Browser – Browser G.711, Opus calls – WebRTC audio calls from Browser to Browser with the use of specified codecs.
Browser – SIP/GSM G.711, Opus calls – WebRTC audio calls from Browser to SIP and from SIP to Browser with the use of specified codecs.
Browser – SIP/GSM G.729 calls – WebRTC audio calls from Browser to SIP and from SIP to Browser with the use of specified codecs.
Plain Instant Messaging – Websocket Browser to Browser, Browser to SIP, SIP to Browser messaging using text/plain MESSAGE requests.
SIP SIMPLE Instant Messaging – Websocket advanced messaging using SIP SIMPLE specifications: CPIM, IMDN, multiple recipients messaging.
Hold and Transfer calls – WebRTC – SIP Hold and Transfer features – hold and transfer of calls.
DTMF RFC 2833 and SIP INFO – Websocket/WebRTC – SIP DTMF via SIP INFO or RFC2833.
Audio Call Recording – WebRTC – recording of audio calls.
Add Flash Fallback – possibility to use audio and signaling features mentioned above in WebRTC compatible browsers such as Chrome, Firefox, Opera, Android Chrome, Android Firefox, as well as browsers without WebRTC, but having Adobe Flash Player, such as IE10, Safari and other browsers with the latest version of Flash Player. This option is applicable to “Browser – Browser G.711, Opus calls”, “Browser – SIP/GSM G.711, Opus calls”, “Browser – SIP/GSM G.729 calls”, “Plain Instant Messaging”, “SIP SIMPLE Instant Messaging”, “Hold and Transfer calls”, “DTMF RFC 2833 and SIP INFO”, “Audio Call Recording” features.
Video Calls
Browser – Browser VP8 calls – WebRTC video calls from Browser to Browser using WebRTC VP8 codec.
Browser – SIP VP8 calls – WebRTC video calls from Browser to SIP and from SIP to Browser using WebRTC VP8 codec.
Browser – SIP VP8-H.263 calls – WebRTC – SIP Video calls using H.263 and H.263+ video codecs.
Browser – SIP VP8-H.264 calls – WebRTC – SIP Video calls using H.264 video codec.
Browser – SIP H.264 calls – Flash Player Browser – SIP H.264.
Browser – SIP H.263 calls – Flash Player Browser – SIP H.263.
Browser – Browser H.264 calls – Flash Player Browser – Browser H.264.
WebRTC Live Video and Broadcasting
WebRTC broadcasting between browsers – web browser is a broadcating source.
WebRTC broadcasting between RTSP IP Cam and browsers – IP camera is a broadcasting source.
WebRTC broadcasting using arbitrary third party RTSP server – Third party RTSP server is a broadcasting source.
Sharing of a WebRTC stream as an RTSP stream – Broadcasting of a WebRTC stream to RTSP players, RTSP re-streamers and other RTSP devices: VLC, QuickTime, Wowza, RTSP top boxes, etc. Stream source should be WebRTC browser like Chrome or Firefox. Stream source can be a screen-captured stream for real-time screen sharing.
Sharing of a WebRTC stream as a Websocket stream – Broadcasting of a WebRTC stream to browsers supporting HTML5 Websockets and HTML5 Canvas: iOS Safari, Internet Explorer, Chrome, Firefox, etc. Stream source should be WebRTC browser like Chrome or Firefox. Stream source can be a screen-captured stream for real-time screen sharing.
Sharing of an RTSP stream as a Websocket stream – Re-Streaming of an RTSP source and delivery the RTSP source as HTML5 Websocket stream to compatible browsers: iOS Safari, Internet Explorer, Chrome, Firefox, etc. Stream source should be an RTSP source supporting H.264 and G.711 codecs.
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