Web Call Server ENTERPRISE License

This is a calculator for enterprise license. It is a preliminary pricing that can be adjusted depending on your final requirements.

Please contact us sales@flashphoner.com with any questions

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	HOW THE CALCULATOR WORKS
	Please select a suitable licensing plan and a set of features. Aren’t sure which features you need? Please, read LICENSING TERMS below or contact us via email sales@flashphoner.com or skypeID flashphoner.com.

LICENSING TERMS

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.

LICENSING

1. The license is lifetime.

2. Periods of free included technical support and updating are 3, 6, 9, 12, 18 months for Starter, Advanced, Enterprise-1, Enterprise-2, Ultimate plans, respectively.

3. Guaranteed response time is the next business day by email or via ticket system.

4. Cost of the next period of technical support and updating is 100%, 80%, 60%, 50%, 40% for Starter, Advanced, Enterprise-1, Enterprise-2, Ultimate plans, respectively.

5. The license is designed for one hardware or VPS server.

6. The license can be allocated between 2, 4, 8, 16 servers for Enterprise-2 128 SC, Enterprise-2 256 SC, Ultimate 512 SC, Ultimate 1024 SC plans, respectively. For example, Enterprise-2 128SC can be allocated between two hardware or VPS servers, 64 SC max per server.

PAYMENT METHODS

We are working with SWREG payment gateway.

Visa, Mastercard, PayPal, Wire Transfer and other payment methods are accepted.

If you have any issues related the payment methods, please contact our sales department sales@flashphoner.com.

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		Starter		Advanced		Enterprise-1		Enterprise-2		Ultimate
Features	Simultaneous Calls (SC)	2	4	8	16	32	64	128	256	512	1024
Features
Browser-SIP Audio and Signaling Features
WebRTC Audio & Signaling	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
Add Flash Fallback
Browser-SIP Video Features
WebRTC Video	Browser - Browser VP8 calls
	Browser - SIP VP8 calls
	Browser - SIP VP8-H.263 calls
	Browser - SIP VP8-H.264 calls
Flash Video	Browser - SIP H.263 calls
	Browser - SIP H.264 calls
	Browser - Browser H.264 calls
Streaming Features
Video Chat	WebRTC as WebRTC
	RTMFP as RTMFP
	RTMFP as WebRTC
	WebRTC as RTMFP
iOS Safari Playback	WebRTC as HTML5 Websocket
	RTSP as HTML5 Websocket
	RTMFP as HTML5 Websocket
Stream Recording	WebRTC Stream Recording
Stream Recording	Flash Stream Recording
RTSP Playback	RTSP as WebRTC
RTSP Playback	RTSP as RTMFP
RTSP Server	WebRTC as RTSP
RTSP Server	RTMFP as RTSP
Supplementary Features	Broadcasting of a SIP call to RTMP server
Supplementary Features	WebRTC Screen Sharing
Support and Maintenance
Included Support and Maintenance Period		3 month		6 months		9 months		1 year		1.5 year
Cost of Next Support and Maintenance Period		100%		80%		60%		50%		40%