LanScape VOIP Media Engine™ - Simplicity taken to the extreme™

Wouldn't it be great if you could easily integrate RFC compliant network telephony features into your current product offering? Wouldn't it be great if you could breath new life into that legacy client/server architecture you currently are struggling with? Wouldn't it be great if you could design and implement a single or multi line soft phone in under two weeks? Wouldn't it be great if you could do all of this without having to kill yourself trying to be a network telephony guru? Wouldn't it be great to achieve these goals in a fraction of the time it normally requires? The list goes on and on and on... Guess what, you can.

Introducing the LanScape VOIP Media Engine™, a fully RFC compliant turn key network telephony solution you can integrate and deploy immediately.

The LanScape VOIP Media Engine™ has been designed to support the most widely used network telephony features in use today. In addition, it is superiorly tailored for deployment in wireless network infrastructures such as Wi-Fi and 802.11 wireless ethernet. If you have never experienced voice over IP telephony using a wireless network, you are in for a pleasant surprise.

Your applications will be able to initiate phone calls, receive and answer phone calls, place calls on hold, perform call transfers and perform multi-party conference calls. All of this functionality is brought to you using a memory and resource footprint you would not have thought possible.

The telephony engine is designed to be used on all 32 bit versions of the Microsoft Windows family of operating systems. LanScape's VOIP Media Engine™ is particularly suited for Microsoft Visual C++ developers. In other words, the native LanScape telephony API is supplied with a C language interface. However, LanScape and our customers have used the telephony engine with just about every other language known to man, be it compiled languages or interpreted. If you are developing using another language other the Microsoft Visual C++, what you want to accomplish is absolutely possible and not difficult. We suggest you search the LanScape web site for further details associated with other language support.

Developing an application based on LanScape's VOIP Media Engine™ is straight forward. Developers need only be concerned with understanding the telephony API and associated callback (event notification) mechanisms. Depending on the task you need to complete, much of the API can be ignored. This is also true for the callback event notification mechanism and the other media based callbacks. It all depends on what you the developer must accomplish. For example, if you are developing a simple two line full featured voice over IP soft phone, you will have no need for the media streaming capabilities of the API. The telephony engine will do all the media handling for you.

Likewise, if you are not developing a voice enable telephony application (speech recognition capable), you can completely ignore the local speech interface and the phone line IVR interfaces.

Unlike other APIs or COM programming interfaces you might have already used in your development (which can have a huge number of scantly documented API procedures), the API associated with the telephony engine is modest in size. This is a deliberate attempt to give you the tools you need in addition to lessening the amount of new information you are forced to remember.

The figure below shows the basic programmer's view of the LanScape's VOIP Media Engine™. The diagram breaks the view into two areas: The "User's Application" space and the "Telephony Engine" space.

Depending on your intended telephony application, the LanScape's VOIP Media Engine™ will be there for you to handle all of the time consuming details. For example: Managing local audio hardware resources, managing session initiation using the SIP protocol, managing internal call states, performing media streaming using RTP protocol, to allowing you to access any and all media streams managed internally. You have complete access to information such as who is calling you, the data rate and format of active calls just to name a few.

For those of you developing automated attendant or speech recognition based telephony designs, the LanScape's VOIP Media Engine™ fully supports access to locally managed digital audio data streams. This access allows for complete speech enabled command and control telephony applications. In addition to this feature, you also have complete access to received telephony audio for any of the supported phone lines. Designing a speech recognition interactive voice response telephony application using this capability is extremely easy.

Also, for those developers creating their next generation of soft PBXs or PSTN gateways, your ability to stream telephony audio data to and from the telephony engine will make your task of integrating digital or analog line card support much easier.

These are just some of the many features offered to you when you integrate the LanScape's VOIP Media Engine™ into your next development effort:

Partial Feature List:

Initiate phone calls.

Receive phone calls.

Place one or more calls on hold.

Initiate call transfers to a new destination.

Perform complex multiparty conference calls.

For each concurrent phone call, mix formats and rates as you see fit.

Complete management of local audio hardware functions for playback and record of telephony audio.

Complete application access to all recorded digital audio media streams.

Supports speech recognition interface for locally recorded audio. Speech engine formats and rates supported are 11kHz PCM and 22kHz PC sampled data.

Aux digital output interface for applications wanting to mix down additional digital audio. Supports uLaw or PCM samples.

Complete access to received digital audio media streams for each phone line.

Complete support to allow applications to perform DSP functions or speech recognition of received phone line digital audio.

Jitter compensation can be activated and applied to received phone line audio data.

Noise discrimination and gate filtering control of all phone line transmitted audio.

Independent phone line volume control.

Phone calls can negotiate one of the following possible data rates and formats:

8kHz uLaw
8kHz aLaw
G729
G729A
iLBC 20Ms
iLBC 30Ms
Speex narrow band
Speex wide band
8kHz PCM
11kHz PCM
22kHz PCM

The MAX number of supported phone lines is limited only by host memory and operating system resources (licensing requirements apply).

Multiple Telephony engine instantiations supported (licensing requirements apply).

Stream telephony audio data between the telephony engine and telephony line card hardware with ease.

Take advantage of the underlying SIP/RTP capabilities of the Lanscape VOIP Media Engine™ to interact with a myriad of other telephony enable devices and equipment.

Incorporating the LanScape VOIP Media Engine™ into your next product development effort will greatly enhance what you can achieve not to mention the time that will be saved by designing with a turn key off the shelf solution.

As you pursue your network telephony goals, you have at hand a ready made solution you can take advantage of immediately... not next week, not next year, but right now.

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