Introduction:
Do you need to develop a network telephony application relying on SIP and RTP protocols in addition to managing call states?
DO YOU WANT TO DRAMATICALLY REDUCE YOUR TELEPHONY DEVELOPMENT EFFORT?
Are you developing a network telephony application for Wi-Fi, 802.11 wireless, dial-up or bandwith constrained environments?
Are you fed up with your current overly complex VOIP application development platform?
Do you want to focus on your telephony solution and forget the underlying technology, bugs, details?
If you answer yes to any of these questions, then the LanScape VOIP Media Engine™ is for you.
The LanScape VOIP Media Engine™ is a complete drop in call engine solution for Microsoft Windows® applications requiring SIP/RTP network telephony capabilities.
The telephony capabilities of the LanScape VOIP Media Engine™ will blow you away when you see how quickly your telephony applications can be developed. Most developers report spending more time developing the application's user interface than they spend on integrating the telephony features of the Lanscape VOIP Media Engine™ .
You are now free to forget all of the miniscule details associated with SIP and RTP protocols and the management of call states. Also, the system memory footprint and operating system resource demands of the telephony engine are extremely low versus the incredible capabilities offered to your applications. Your development effort will be able to harness a wealth of truly powerful features. Base your next voice over IP telephony design on something more than a "protocol stack".
Brief Overview:
Designed for the Microsoft Windows® family of operating systems. Supports all versions since Windows 2000.
Upgrade to the next higher line version in the first year and only pay the difference in price.
Initiate calls, receive calls, place calls on hold, transfer calls, busy out phone lines, and perform multiparty conference calling. Wickedly cool!
Supports telephony industry
standard formats and rates in addition to higher voice quality formats and rates. Supported codecs include: uLaw, aLaw, G729/G729A, iLBC (20Ms and 30Ms frame size), Speex narrow and wide band, 8k PCM, 11k PCM and 22k PCM.
Improved interoperability with other SIP software and devices.
Belcore compliant DTMF support.
Call recording.
Complete call state handling and access to all telephony media data streams.
Provides speech recognition interface for command and control applications. Supports 11kHz PCM and 22kHz PCM voice recognition data streams.
Complete phone line IVR interfaces. Perform DSP operations or speech recognition on all phone line received telephony voice data. Also, stream voice or audio data to any phone line as your application requires.
Will completely manage host multimedia hardware if required by the application.
Perfect for stand alone Windows applications in addition to client or server side WEB telephony solutions.
Stream data to/from all phone lines, and to/from local multimedia hardware.
Incorporates smart RTP transceivers that perform jitter compensation and noise discrimination/gating of telephony audio data.
The perfect telephony engine solution for Wi-Fi and 802.11 wireless ethernet networks. The incredible low voice latency will
ensure the highest possible voice characteristics.
Supports inbound and outbound Digest authentication using MD5 hashing algorithm for REGISTER, INVITE, BYE, SUBSCRIBE and NOTIY SIP messages.
Build superior class telephony applications that have small memory requirements and small operating system demands.
...and much more.
For additional details, please click on the product image to the left.
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