Telephony Sound ToolKit
Tele-STK provides ultra-low latency for Windows and Linux-based soft phone products--good enough to compete with hard phones!
English
Supported Technologies
Windows 95/98/ME,
Windows XP/2000/NT
Software
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Pricing
Other
$15,000 to $105,000
tstk@dw.com
(480) 380-1122
Additional Product Information
The purpose of a telephony audio driver is to manage the PC sound card. It has two tasks: send captured microphone audio data to the application, and receive network audio data from the application. It must let the application control when to exchange data, and provide smooth playback even when audio data delivery is jittery.;Telephony Sound ToolKit (Tele-STK) has achieved the best latency performance and quality of service of any Windows telephony client. DiamondWare?s audio experts have systematically explored the Windows audio APIs, and have produced an integrated solution to the myriad problems of robust low-latency full-duplex audio. Tele-STK can reliably deliver audio buffers much faster than the Internet can transmit the packets, thus enabling IP telephone products to run at full speed, and LAN-based products to transmit audio with no perceptible delay.;Licenses to incorporate this radical new technology into commercial products are now available.;Tele-STK can achieve double-round trip times as low as 53ms. Double round trip means the sound goes into the microphone on machine A, across the network, plays at machine B?s speakers (whose microphone is aimed at its own speakers), goes into machine B's microphone, across the network again, and finally plays at machine A's speakers.;Tele-STK defines a new standard for Quality of Service. Its advanced architecture enables it to skip less often than other programs, and in the rare event of a skip, it will be shorter in duration and lower in intensity.;Ease of use. Audio buffer delivery is handled either by Tele-STK or application callbacks, or both, as required. Once it?s initialized, Tele-STK handles all audio issues.;Integrated DSP. Digital Signal Processing solutions are already in place to reconstruct missing packets, perform sample-rate and format conversion, and provide an automatic gain controller.;Automatic drift correction. Drift is caused by different sampling rates on each client PC (e.g. Machine A is 11023hz and B is 11034hz). Data exchange is specified by the application (typically 8000 Hz 16-bit mono).;Autodetect. Many sound cards report multiple devices, and it can be difficult to synchronize which playback devices, record devices, and mixer devices go together. This is compounded by the fact that some sound cards will only operate in full-duplex mode using one particular combination of record and playback devices at a particular sample rate and format. Tele-STK detects these cases, and automatically handles them.;Hardware mixer solution. Complete volume/mute control over all mixer lines necessary for telephony.;Time to market. Tele-STK is now shipping as a part of commercial applications. ;Comprehensive testing. Tele-STK has been tested on dozens of hardware platforms and configurations. It is reliable with ISA and PCI sound cards, desktop PCs, laptops. There is no substitute for testing on many permutations if you want to produce a bug-free product.;Supported Operating Systems: Windows 95, 98, Me, NT, 2000, Whistler (Linux to be released 4/01);Supported Sound APIs: waveOut, waveIn DirectSound, and DirectSoundCapture