IntraVoice CO Employs The Most Advanced
IP/Voice Codec.
Digitcom's engineers have worked with Voxware, Inc., the
leading developer of codec technology, to be the first to
integrate a MetaVoice RT (Real Time) codec with a UNIX
data network. DLD's UNIX-based IP/Voice network provides
more solid channel communications, lower operating system
overhead, more flexible network administration and higher
reliability than other operating system gateways.
Digitcom Engineering Is Advancing IP/Voice Technology.
The RT29 codec was developed to transmit high quality
speech over
low-bandwidth channels with an extended speech model to
deliver the highest quality speech at 2978bps. RT29 v2.0 also
supports technology which allows DLD engineers to modify
the characteristics of the voice fidelity. In addition, RT29
supports other add-on functionality including time scale
modification, Voice Activity Detection, and Automatic Gain
Control. The RT29 also enables IP-based multimedia
development and interactive applications that require the
highest quality audio at low bandwidth transmission.
Digital Long Distance Delivers Value-Added Capabilities.
Both as a dial-up point-of-presence (POP), and in our
IntraVoice
enterprise-level gateway server, the patented technology
DLD employs that bridges the public switched telephone
network to the Internet can deliver an array of services.
IntraVoice technology makes custom computer/telephone
integration (CTI) applications simple to develop and easy to
administer. Applications provided for in the DLD Network
include: * LAN/WAN voice communications * IP faxing *
global voice messaging * data transmission * video
conferencing * computer to computer document-sharing *
integrated e-mail/paging/text-to-voice messaging *
Web-based call center applications.
Patented Technology Delivers Seamless Service.
The DLD Network Node gateway handles all aspects of call
setup: management, IP address mapping, and delivery of the
data stream to the IP router. Call management software
determines the availability of channels, and assigns calls to
the next available channel. The node nearest the destination
uses the customer's call information to dial out and complete
the voice circuit. Administrative software times and tracks
the call data, which is automatically sent to the DLD Network
Operations Center for databased tariff and billing functions
IntraVoice CO Is The Gateway To 21st Century
Telecommunications.
Internet voice transmission is poised to revolutionize the
telecommunications industry. The convergence of voice, fax,
and multimedia communications with data transmission over
computer networks is bringing dramatic changes in products
and services offered to telecom customers. And, it offers
remarkable savings -- especially on expensive international
routes. The key to these developments is digital
compression and transmission over Internet
Protocol networks. Technology that is found in the
IntraVoice System.
- About Internet Protocol voice transmission.
- IntraVoice is the most robust IP/voice platform available
today.
- Basic IntraVoice installation specifications.
Transmitting Voice Communications over
Intranets and the Internet.
The Internet Protocol (IP) is a universal standard for
handling digital
information on the Internet, on corporate "intranets" and
wide area networks (WAN). IP telephony translates voice,
fax, or graphics analog input into a digital packets of data
and sends them to a distant computer using the Internet to
transport the data. The digital packets are re-assembled at
the other end and translated back to a graphic or audio signal
which can then be delivered over standard telephone lines.
In IP/Voice transmission there is always a
compression/decompression (codec) algorithm employed to
reduce the data load on the network. A codec is used at both
ends of the data link to compress and decompress audio.
There are several characteristics to consider when evaluating
a codec - bit rate, number of channels, sampling frequency,
output quality, complexity, and the availability of "tuning"
features.
Elements of Digital Compression/Decompression.
Bit Rate - is a measure of the number of bits of compressed
data in a fixed
interval of time (usually bits per second). Codecs can
produce and consume a fixed (constant) or variable number
of bits in any established interval of time. A variable-rate
codec categorizes speech input into four classes: silence,
unvoiced, mixed voice, and voiced. The bit rate and the
perceived quality of the voice reproduction are closely
related. It is the engineer's artistry to optimize the tradeoff
between low bit rate and high quality sound output. Number
of Channels - Codecs support mono (1 channel) or stereo (2
channels) transmission. Sampling Frequency - is the interval
at which an analog sound signal is sampled. Higher sampling
frequencies allow for higher quality sound reproduction. The
native sampling frequencies of codecs limit their fidelity. The
IntraVoice technology can use input with a higher sampling
rate because it employs the Voxware* Compression Toolkit
(VCT), which supports down-sampling. Usually, speech is
sampled at 8 kHz; CD audio quality is at 44.1 kHz. Output
Quality - the Mean Opinion Score (MOS) is used to measure
output quality. A MOS value of 4.0 is considered 3toll
quality2. That is, MOS 4.0 is equivalent to the voice
reproduction of a standard Public Switched Telephone
Network (PSTN) connection. Complexity - can be stated as
the percent of CPU tasking dedicated to a (de)compression
algorithm, which must be considered in conjunction with bit
rate and output quality of the decompressed sound data.
Lower complexity is better. Additional codec features -
depending on the codec, support for the following features
can be available to engineers: pitch shift, automatic gain
control, warping, or VoiceFont on compression or
decompression.
Point-of-Presence Basic installation
IntraVoice Central Office Point-of-Presence (Basic
installation -
Expandable) Dual Pentium 200 CPU (or) Pentium Pro 200 (or)
Pentium 200 MMX minimum 64 meg. RAM Adaptec 3940 4
gig SCSI HD 4x CD ROM Drive SCSI Super VGA monitor
Super VGA Video Card with 2mg DRAM Microsoft
compatible serial mouse 101 keyboard 350 to 500watt power
supply Solaris 2.6,Sco-Unix OS and Unixware 2.0/2.1 Remote
diagnostics/support. Up to 1024 time slots per entity
IntraVoice multitask gateway server Optional features:
Resound CO voice processing Linguist multilingual voice
prompts module Digitfax module NETcall real-time voice over
IP Passive back plane rack Client Requirements: Dedicated
24hr access Internet Link (frame relay, T1, T3, ATM, etc)
using either twisted pair or BNC connection. T1, PRI/ISDN
or E1 connection to interface with callers Connected to the
same network as to the IVR you must have a Windows
95/98/ NT 4.0 with TCPIP installed for administration and
maintenance of the IVR. NetCall Module: Solaris 2.6 only
RT29HQ voice compression Silence suppression Delay
Management System (optimizes audio) Full-duplex
Echo-canceling algorithms Forward error correction
Interpolation algorithms (smoothes missing packet data with
preceding and following voice samples) Office to Office
and/or Least cost routing capabilities FAXport Module: SCO
UNIX or UnixWare Operating Systems only Requires
additional FAX resources Office to Office and/or Least cost
routing capabilities VOXport Module: Optimizations can be
made if using Solaris 2.6 on both source and destination
systems Office to Office and/or Least cost routing
capabilities Technicians monitor network operations and
network node performance 24/7, and correct almost all
problems remotely. Least cost PSTN routing updatedand
available 24 hours a day.
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