This feature embodies the third volume of TMC Labs Innovation Awards
awards for INTERNET TELEPHONY� magazine and, we feel, well represents a
milestone of maturity for the industry. With a variety of entrenched
products and companies in our field, we�ve really drilled down into the
product space this year. In the past we�ve always come up with a
comprehensive list of products that in many ways will shape voice�s
technological year to come (approximately 30 products have already been
chosen for an Innovation Award by TMC Labs); this year we feel is no
exception.
In keeping with its namesake, the Innovation Awards are presented to
products TMC Labs feels are exceptional and innovative; the products
selected must additionally be on the market contributing to the evolution of
packet voice. We found ourselves digging much deeper for this year�s
awards, exploring many new, more specialized solutions employing
up-to-the-minute technologies in this space: XML, 802.11, and Bluetooth to
name only a few. SIP seemed to be the protocol of choice in most of our
product entrants this year; we also noticed a dramatic increase in
presence-based solutions, some of which gated both IP and wireless
architecture.
Our most successful Innovation Award campaign to date, we received almost
200 applications for this year�s event. While it obviously meant more work
for us, the task didn�t seem quite so daunting as we reveled in delight at
the sheer volume of VoIP product submissions. Both sophomore and tenured
companies submitted applications this year announcing new, recently released
products. More intriguing still were the number of applicants providing
solutions to major PSTN carriers. We hope you enjoy Internet Telephony�s
third annual TMC Labs� Innovation Awards. With an overwhelming response
this year, it has truly become a labor of love.
Choosing Alcatel�s OmniPCX 4400 as one of our Innovation Award winners was a no-brainer for several reasons. By running on UNIX, Alcatel certainly will avail itself of a ton of developer support from the vast UNIX developer community. Besides running on the UNIX operating system, the OmniPCX supports all the main relevant IP standards, including embedded DHCP, RTP, RTCP, UDP, TCP/IP, H.225, H.245, RAS, TFTP, and more. It also supports QoS standards such as 802.1 p/Q, ToS, and DiffServ.
Another innovative aspect of the OmniPCX is that unlike most of its
competitors, it doesn�t require proprietary phones. The OmniPCX 4400
supports third-party H.323 phones, such as the PolyCom SoundPoint IP and
even third-party SIP phones, such as Pingtel�s SIP phones.
Alcatel�s ABC protocol is a superset of QSIG that optimizes reliability
and performance in an �all Alcatel� network. For heterogeneous
configurations (not a 100 percent Alcatel network), the QSIG multi-vendor
protocol standard can be used to connect to legacy PBXs. In single site
configurations it scales up to 5,000 users, and by networking up to 100 Call
Servers it can scale up to an impressive 50,000 users.
As if the OmniPCX didn�t have enough features rolled all in one, it
also features a unique in-building wireless functionality that is integrated
with the OmniPCX. The radio base station connects directly to the OmniPCX
using digital cards instead of going through a gateway. In addition, the
OmniPCX also works with Symbol�s wireless VoIP phones (another 2002 TMC
Labs Innovation Award winner).
The OmniPCX has an extensive list of optional call center modules,
including ACD/IVR functionality, outbound dialer for campaigns, agent
login/logout, discrete call listening for supervisor, multimedia
skills-based routing, and extensive reporting and statistics. Its multimedia
capabilities include Web call back, Web call through (VoIP), text chat,
e-mail, and collaborative browsing. TMC Labs loved the Alcatel OmniPCX 4400
right from the start and we wish we had one in the lab right now just so we
could tinker with it.
So your company just bought one of those newfangled IP-PBXs. Certainly
one of the purchasing factors was that IP-PBXs allows for a far simpler
installation making moves, adds, and changes much simpler. Unfortunately, if
your organization is a call center or any other company that implemented an
IP-PBX and requires call recording � you were out of luck. There were no
call recording products that could record VoIP conversations traveling
across the IP network.
We should point out that the recording of traditional PBXs required the
connection to proprietary telsets or telephone trunk lines along with
proprietary CTI links. Simple software configuration of the business rules
for recording are just some of the benefits of VoIP recording. The
complexities and costs associated with CTI begin to disappear as voice and
data converge onto a single network allowing managers to record, monitor,
and analyze customer interactions, including the agents� screens. In
theory the ability to record both voice and data over IP is great, but until
recently the technology to record VoIP calls did not exist. Now, with
Eyretel�s MediaStore IP product, a comprehensive, enterprise-wide VoIP and
screen recording solution is available.
MediaStore IP integrates with the Cisco IP Contact Center (IPCC)
supporting Cisco�s SCCP (�skinny� protocol) and other IP telephony
solutions, enabling contact centers to use the same IP infrastructure to
process both voice and data. Using MediaStore IP, contact centers will be
able to record all the various customer interaction touch points. In
addition to recording the voice, MediaStore IP will also record the
agent�s screen to capture such important activities as Web chat, agent
assisted Web browsing, desktop computer activity and e-mail. MediaStore IP
provides the ability to monitor and analyze all of the recorded information
via a single system. One unique feature of MediaStoreIP is its ability to
record conversations in stereo, allowing for on-screen playback that
visually displays each side of the conversation. Also, calls can be queried
and retrieved and then replayed through a browser interface, allowing
managers or agents with appropriate security rights to view calls from
anywhere. Overall, TMC Labs was very impressed with Eyretel�s MediaStore
IP product, which establishes new ground for IP telephony.
Last year Genuity launched a TV ad campaign that featured a model of a
black rocket at some point in the commercial. One of them was a boardroom
with a black rocket sitting on the conference table and a twenty-something
geek getting the old board members to somehow perform the �wave� by
raising their hands. If you were like us, you were probably dumbfounded what
the heck the black rocket did. The commercial seemed to elicit some strange
magical powers upon the �black rocket� � as though it were the
solution to all your problems. Although not as annoying as the SuperBowl�s
nebulous mLife commercials, it certainly did its job in branding the Black
Rocket name. One year later when TMC Labs received an online application
from Genuity applying for the TMC Labs Innovation Awards we instantly
flashed back to their commercials. Thus, their brainwashing, err marketing
messages sure did the trick!
We checked out Black Rocket on Genuity�s Web site and determined that
Black Rocket is actually made up of several disparate modules often
completely unrelated to one another. For example, Black Rocket Voice is
VoIP-related and Black Rocket Storage is backup, restoration, and archiving
of data, with the ability to restore entire systems. Yet another module is
Black Rocket Hosting, which is Genuity�s solution for secure, enterprise
hosting infrastructures. With Black Rocket having such a varied product
line, our confusion about what exactly made up Black Rocket only seemed to
worsen. However, after careful examination and research we figured out just
what Black Rocket is all about. We could best sum up Black Rocket as a
solution that provides voice/date convergence on a managed backbone network
that offers enhanced services such as disaster recovery, VoIP with quality
of service guarantees, managed Web hosting, managed security services.
We focused our attention on Genuity�s Black Rocket Voice module, which
is a Voice over Internet Protocol (VoIP) communication solution for
enterprises that integrates voice and data traffic onto a single,
multi-protocol IP network infrastructure utilizing Genuity�s Tier 1 IP
backbone. Black Rocket Voice is designed to supplement or replace existing
enterprise voice solutions for both inbound and outbound calling including
intra-company (�on-net�) voice and data communications, and also for
inter-company (�off-net�) communications anywhere in the world. Genuity
can work in conjunction with legacy PBXs to deliver enterprise-wide IP
telephony service while protecting an enterprise�s investment in existing
PBX and telephony handset equipment.
Genuity claims that their Black Rocket Voice solution is the first
convergence solution to deliver end-to-end prioritization of voice traffic
from the customer premise across a backbone network. To back up their claim
of voice quality, Genuity is the only company that we are aware of that
offers voice quality metrics for service level guarantees. In fact, Genuity
provides a proactive service level guarantee for the Black Rocket Voice
services and a reactive service level guarantee for service availability.
Genuity guarantees a Service Availability of 99.97 percent, Network Latency
of 55ms, Packet Loss of up to 0.25 percent, Jitter of 10ms, and Voice
Quality � as measured by the Perceptual Analysis Measurement System (PAMS)
scale of 4.0 (carrier quality) for the G.711 codec and 3.7 (business
quality) for the G.729 codec.
Cisco and Verizon (a minority stakeholder) are two of the first customers
for Genuity�s Black Rocket Voice. It�s part of the Black Rocket
eBusiness Network Platform for enterprise Internet access, hosting and IP
transport over Genuity�s Tier 1 managed OC-192 backbone, delivering 50
times the capacity of a frame relay network to the customer site. The Black
Rocket Voice service extends the Black Rocket platform to now include
next-generation business communication services and enables business
continuity for enterprises with a potential savings of up to 40 percent in
total data and telecommunications costs.
�By deploying Black Rocket Voice to 41 domestic sites, Cisco expects to
streamline internal communications, using the 35,000 IP phones we�ve
currently deployed,� said John Bruno, vice president, information
technology, at Cisco. �Genuity�s Black Rocket Voice service is regarded
by Cisco as an important milestone in our long-range plans to migrate many
of our voice and data services to outside service providers and the key to
ensuring the continuity of our business communications. Genuity�s network
management expertise allows us to focus on our core business.�
For providing a converged voice/data solution for the enterprise while
offering QoS guarantees and measurable service-level agreements, TMC Labs
commends Genuity and we grant our TMC Labs Innovation Award without
reservations.
Any administrator relying on a VoIP solution to deliver voice
communication between an employee desktop and the Internet no doubt
understands the Internet telephony idiom: �NAT issues.� While Network
Address Translation (NAT) is a proven method used to secure data networks by
converting the address of each LAN node into one external WAN IP address
(and vice versa) preventing a direct path to internal hardware, it�s not
quite as effective a system for sending and assembling voice packets
correctly. Since most enterprises use NAT-enabled firewalls, many VoIP
devices lose quality, partial functionality or are rendered entirely
inoperable from behind such enterprise protection.
Enter Jasomi Networks� PeerPoint Enterprise Edition. The PeerPoint
Enterprise Edition is a SIP-to-SIP gateway that translates VoIP between an
internal, private IP address to the external global address space, enabling
calls to pass into and out of the organization across the NAT boundary.
Jasomi accomplishes this by employing what they call a Back-To-Back User
Agent (B2BUA) combined with an integrated media proxy; one side of the B2BUA
handles internal call flows on the private address space, while the other
side handles external flows on the global address space. PeerPoint
essentially divides each VoIP call into two halves, the voice communication
up to the firewall, and the internal communication behind the firewall.
This results in what looks like call termination at the box, in much the
same way a IP-to-PSTN call is terminated, however while maintaining the
external, incoming voice transmission, PeerPoint simultaneously begins
another internal SIP session, communicating the voice information to its
destination while performing all protocol translations and directing media
streams between the both halves of the call. In this way PeerPoint
Enterprise Edition uniquely and innovatively handles �NAT issues�
providing the appearance of a single, uninterrupted VoIP call.
Contrary to what you may be thinking, Jasomi Networks says that
deployment of PeerPoint Enterprise Edition does not impose an additional
load on an incumbent firewall affecting its traffic, security, fraud
protection, or topology hiding. Using B2BUA allows an enterprise to use VoIP
without gating to a circuit-switched network, thereby minimizing costs
related to equipment, services and telco charges. Since it�s an out-of-box
solution, Jasomi also says �PeerPoint Enterprise Edition can be installed
without controversy, and configured in a matter of minutes.� Engineered
for housing within an enterprise subnet, or DMZ, the unit also allows
companies to utilize Communication Application Service Providers (CASPs)
that otherwise would not have access due to equipment limitations.
Jasomi Networks has defined SIP-to-SIP solutions on several different
levels. In addition to the Enterprise Edition, they�ve created a PeerPoint
for carrier and service provider networks to manage and control SIP streams
that, among many additional features, have the ability to perform
Communications Assistance for Law Enforcement Act (CALEA) legal intercept.
Since all signaling streams pass though PeerPoint hardware, the device has
the capability to examine, intercept, duplicate, and forward copies of call
information to a central collection point.
After TMC Labs tested the APA III-4FXS product for the Jan �02 issue of
Internet Telephony magazine, we realized first hand what a truly unique
product Mediatrix had developed. Though voice gateways alone aren�t quite
as innovate as they were three years ago, add a four-port FXS interface,
remote management, PSTN fallback; pack it all into a small box and employ
full-duplex and silence suppression along with industry-standard compression
codecs, and this device is in a space all its own.
The APA III-4FXS is a standalone device compliant with all the major
voice protocols: H.323, SIP, and MGCP. This telephony adapter connects up to
four analog terminals to a LAN providing a gateway for packetized audio
communication. It additionally supports the features provided with Analog
Display Service Interface (ADSI) phones such as the Nortel Vista 350. Along
with voice, G3 and Super G3 fax transmissions are also supported at speeds
up to 33.6 Kbps. Real-Time Fax Over IP (FoIP) with T.38 protocol stack can
be used and automatic fax mode detection is available on all ports.
Similarly, all ports support standard codecs: G.711, G.723, and G. 729. The
unit�s software is also upgradeable via TFTP and supports DHCP.
The APA III-4FXS also has a fifth RJ-11 jack used for connection with a
PSTN line. During normal operation the line is �switched out� or
excluded from the circuit. When the power is disconnected from the unit, or
the power fails, the relay setting is restored to a connected state and the
PSTN line can be used as an emergency bypass line.
When power is restored to the unit, the APA III-4FXS does provide PSTN
tones, just as if a user were on a POTS line. On any of the four RJ-11 ports
dial tone is provided when a connected handset is lifted from its cradle.
Similarly, call progress tones such as ring back or a busy tone are also
provided.
As a SOHO gateway with four FXS ports employing H.323, SIP and a PSTN
fail-safe the APA III-4FXS appears to be a unique solution. Add upgradeable
software, DHCP support, and G3 fax support and the unit is definitely
innovative. Couple this feature set with the Mediatrix IP Communication
server compatibility and you�ll be hard pressed to find a standalone SOHO
gateway that can be deployed in various network configurations as a
PBX-to-IP solution, an enterprise CLEC solution, used as an IP-PBX, or to
connect branch offices.
The VoIP Manager from NetIQ is a software solution that actively monitors
VoIP call quality across IP telephony networks. VoIP Manager provides
comprehensive real-time monitoring, management, and reporting on system
health, call quality, in addition to the performance and availability of
critical IP telephony servers, such as the Cisco Call Manager.
NetIQ�s VoIP Manager is doing far more than simply assessing and
monitoring throughput and response time. While these criteria are effective
for assessing and maintaining data networks, voice requires a different
method. NetIQ approaches network assessment and management through
provisioning of its nine modules dedicated to uncovering a VoIP network�s
golden mean. NetIQ targets the areas that have the most effect on a
system�s readiness to handle VoIP traffic: codecs deployed, one-way delay,
jitter, and types and degree of packet loss; and further uses these factors
to calculate the MOS. Additionally, instead of exclusively using SNMP
polling NetIQ incorporates Intelligent Agents that run directly on the
network and interface with native Windows management capabilities.
Since VoIP Manager is entirely software-based, it does not require any
special-purpose probe hardware, or simulation equipment used to collect
network statistics. Instead, NetIQ uses what they call Performance
Endpoints: small executable files designed to run in the background on any
computer connected to the network. The Endpoints simulate traffic based
using a manager�s choice of codecs to calculate a MOS.
NetIQ has also developed Connectors, which allow customers to use their
existing management products in conjunction with VoIP Manager to correlate
events on their network. The Connectors have been verified by solution
manufacturers such as Cisco, to assess their VoIP products. Additionally,
NetIQ offers connectors for Micromuse NetCool/Omnibus, Dell OpenManage, and
HP OpenView Network Node Manager. For example, VoIP manager allows users to
proactively monitor and manage Dell PowerEdge hardware performance when used
(as an IP telephony servers) such as UPS battery levels, network interface
card errors, and computer temperature.
VoIP Manager is a powerful, easy to deploy, out-of-box solution that
enables administrators to assess, provision, and continually monitor VoIP on
their networks. Incorporating automated VoIP management, problem resolution,
automatic alert management, in addition to powerful reporting and charting
tools make it an easy selection for a 2002 TMC Labs� Innovation Award.
Symbol Technologies� NetVision Phone is the first to incorporate
the rapidly growing 802.11b, and the widely adopted H.323 VoIP protocol,
into one wireless handset offering new levels of enterprise functionality
and mobility. The handset essentially provides the feature set of an
Ethernet telephone and the mobility of a wireless handset based on the range
of in-building Wi-Fi access point(s), or network.
Lying face up on an office desk, the NetVision phone could easily be
mistaken for a Nokia or a Samsung cell phone; the circuitry underneath its
familiar looking shell however, is quite a far stretch from a GSM chipset.
The NetVision is both unique and innovative in its housing of 802.11b, ITU
standard H.323, and POP3 technologies, in a small and fully portable
handset. Symbol also provides optional support for Cisco�s Skinny (SCCP)
protocol and enhanced H.323 solutions from Mitel and Nortel. Partnering with
Ericsson, Nortel, Mitel, and Cisco, Symbol has designed the NetVision as an
extension of offerings from these major IP PBX manufacturers, able to
function with all the capabilities you�d expect from a native VoIP
handset: peer-to-peer, extension, name, and speed dialing, call hold, Caller
ID, park, transfer, conference, forward, call waiting, and do not disturb
features.
The NetVision phone handles all the processing, compressing voice via
G.711 and G.729a, packetizes and sends via an 11Mbps Wi-Fi network
connection using the CSMA/CA wireless access protocol to either an H.323
gateway, or IP-PBX with built-in gateway functionality. The NetVision also
houses embedded wireless communication features such as Symbol Wireless
Voice Prioritization, which gives voice packets precedence on the wireless
LAN, preemptive roaming to maximize bandwidth and sustain voice quality, and
up to 10 active calls per wireless access point.
The phone itself stands about five inches in height, 1� inches in width,
with a �-inch diameter. At about 5.5 ounces, it features a small
three-line, 12-character display, and carries a rechargeable Lithium Ion
battery. With three hours talk time, NetVision has its user�s choice of
ringing options while offering vibrating call notification, text paging,
intercom functionality and user configuration options as well.
Symbol also produces a similar product called the NetVision Data Phone,
which addition to VoIP functionality, has integrated bar code scanning and
Web-client data capabilities. Symbol plans to sell the NetVision Phone
directly to its core customers and indirectly to its community of wireless
VoIP telephony and channel partners through OEM and reseller agreements.
Working in conjunction with their Application Switch, the Communications
Suite from Sylantro targets service providers interested in providing
SIP-based, IP-hosted communications infrastructure. The Communication Suite
provides a comprehensive set of hosted PBX and IP-Centrex applications that
allow providers to offer many different degrees of functionality while
allowing end users to forego high installation, hardware, and management
costs. The suite consists of the following components: c-Business, ComCierge,
ComTraveler, ComMerchant, ComRIO, and ComOffice.
c-Business handles the hosted PBX and IP Centrex functionality while
additionally allowing a service provider to provide customers with a Web
portal along with new directory and browser-based features: individual call
logs, click-to-call, and single number support. e-Business also provides
Find Me/Follow Me and other advanced features of the product.
A service provider and notable adopter of Sylantro�s solution,
GoBeam�s DashBoard offering is based on the Communication Suite of
products. TMC Labs took GoBeam�s product for a �First Look�
application test in the April issue of CIS magazine http://www.tmcnet.com/cis/0402/0402lab2.htm
Through our testing, we had the opportunity to interact firsthand with many
components in the Suite, such as ComCierge. ComCierge provides call control,
personalized call treatments, and CallerID, allowing end users to decide how
their incoming calls should be managed.
ComMerchant focuses on distributed call center capabilities for SMBs. The
ComMerchant module features a network-hosted ACD, one of the groundbreaking
features of this product. The browser-based GUI makes it simple for users to
check in or out of queues, display caller information, and balance call
distribution.
Mobility is also factored into the Suite supporting travelers and remote
workers through the implementation of ComRIO (Remote Instant Office) and
ComTravler. Both armed with the provider�s offering and branding, ComRIO
targets the remote worker by extending single-number and single-mailbox
functionality, while ComTravler extends portal capabilities with WAP-enabled
mobile phones allowing users to change reach me or click to call settings
from their mobile phones � another feature we found to be quite unique.
The Communication Suite also offers support for legacy business phone
sets. Sylantro promises to breathe new life into existing hardware by
supporting LCD displays and assignment of features and speed dial to legacy
phone feature keys. Sylantro is also using SOAP, part of Microsoft�s .NET
initiative to provision click-to-call from any Web page and well as
interworking with Microsoft Outlook, which helps comprise their ComOffice
offering.
While there are literally dozens of softswitch manufacturers, making it
seem as though they are a �dime a dozen,� there are softswitches on the
market that are clearly head and shoulders above the rest. Telica�s Plexus
9000 is a very flexible and highly scalable softswitch-based switching
platform that certainly carries the mark of a premier softswitch
manufacturer. Their solution is a softswitch-based packet switch for TDM,
VoATM, and VoP (Voice over Packet) switching, supporting traditional and
next-generation services.
Service providers deploying the Plexus 9000 can cap their investment in
traditional circuit switches and eventually migrate to a converged switching
infrastructure. They can use the Plexus 9000 today as a high-capacity tandem
or Class 5 switch for traditional services such as 3/6/7/10 Routing,
International Dialing, CLASS, E800, E911, LNP, and CALEA. Then when market
conditions are right they can offer next-generation services such as
IP-based voice services including IP-Centrex.
It features a fault-tolerant NEBS Level-3 certified architecture that
Telica claims has an estimated reliability rate of 99.99994 percent. Also,
the Plexus 9000 can easily handle a dynamic mix of circuit- and
packet-switched traffic for Class 4 and Class 5 applications. The Plexus
9000 supports a comprehensive suite of interfaces for both subscriber and
network access and hundreds of services for Class 4 and Class 5 switching
applications The platform includes all the elements of a softswitch-based
solution: Media Gateway, Media Gateway Controller (Softswitch) and Signaling
Gateway.
Supported signaling protocols include SS7, ISDN, CAS, Session Initiation
Protocol (SIP) for VoIP, Broadband Loop Emulation Services (BLES) for VoATM,
and GR-303 for DS1 and Voice Frequency (VF) interfaces. It supports a
variety of interfaces and protocols, including DS-1, E-1, DS-3, STS-1, OC-3,
STM-1, OC-12, STM-4, OC-12c and 10/100/1000 Mbps Ethernet. Telica�s
solution includes VoIP subscriber side access through industry standard SIP
signaling enabling the deployment of new services such as IP-Centrex. It
also includes support for SS7, MEGACO, M3UA/SCTP, and BICC. Additionally,
ISDN, CAS, and GR-303 signaling interfaces allow the Plexus 9000 to leverage
the large installed base of subscribers connecting to hundreds of existing
TDM access devices such as PBXs or Digital Loop Carriers to deliver business
and residential voice services.
One key advantage of the Plexus 9000 is its I/O density, which Telica
claims is an industry-leading 137,088 TDM ports or 24,192 VoIP calls per
shelf (13RU) supported and 411,264 per rack. It also features low power
consumption (0.3 watts/DS1), which is very important in the carrier space.
Its components are hot-swappable and you can configure �hot spares� for
increased redundancy. It includes the required CLASS services such as
calling number delivery, calling name delivery, call waiting, call
forwarding, speed dialing, anonymous call reject, caller identification,
distinctive ringing, call add-on and others. In fact, Telica recently added
software to support call forwarding, call waiting, call reject if caller ID
is blocked, speed dialing, distinctive ring, operator barge-in and the
ability for law enforcement agencies to wiretap phone calls. These are
necessary features to offer residential phone services that are
traditionally delivered via Class 5 switches. Telica will also be supporting
AIN 0.2 that enables Plexus 9000 to tap databases in existing phone networks
to add even more Class 5 features. The Plexus 9000 is the perfect solution
today to perform �toll bypass� while at the same time this platform is
prepared for the future to offer enhanced services. By offering a solution
for today�s and tomorrow�s needs, along with impressive density,
scalability, and reliability, TMC Labs is proud to bestow our TMC Labs
Innovation Award to Telica�s Plexus 9000.
Voyant�s MobileMeeting solution allows carriers to deploy a
turnkey, instant conferencing solution connecting disparate endpoints across
separate networks. Voyant has partnered with OZ and Odigo to create a
wireless point of presence conferencing solution. Uniting OZ�s ICS 2.0,
Odigo�s Open IM Server (OIMS) and Presence Management Server with
Voyant�s ReadiVoice platform has resulted in a unique, reservation-less
conferencing infrastructure for wireless carriers and service providers
called MobileMeeting.
Utilizing the Innovox conferencing media server with ReadiVoice
conferencing solution along with the aforementioned offerings from both OZ
and Odigo, Voyant�s MobileMeeting allows users to initiate a full-duplex
group voice session from any mobile phone, PDA, or PC on-the-fly. This
capability represents a shift in traditional voice conferencing, bringing
instant conferencing to subscribers anywhere and via any device. This
solution also utilizes a carrier-grade, centralized contact and presence
management system providing presence information and thereby identifying who
is available to initiate a point-to-point or group conference.
Open APIs enable service providers to integrate MobileMeeting with an
existing solution, perhaps to fortify an already branded offering and allow
further service customization by integrating voice conferencing with
presence management. Utilizing XML and speech recognition from Nuance,
MobileMeeting is offering some unique conferencing capabities to
subscribers: SMS or e-mail conference notification, personalized
�welcome� conference greetings, as well as a Web portal and WAP access.
MobileMeeting speech recognition offers �hands free� conference
initiation from any mobile phone utilizing conversational speech to invite
and join participants into a conference. Via a WAP interface, mobile users
can simultaneously dial out to multiple participants by selecting
individuals from a contact list. The more detailed address book and group
applications can be edited from the Web portal. The Web portal additionally
allows members to dial out and control calls. Additionally, when dialing
into or creating a call, the conference host has the option to alert
participants via e-mail, or SMS notification.
The uniquely innovative MobileMeeting platform offers anytime, anywhere
voice collaboration to the end user featuring speech recognition and
multiple points of presence using standard devices. Deployed by a carrier or
wireless provider, this solution may also offer an enhanced revenue
opportunity through its distinctive service offering.
Webley�s Media Switching Platform (MSP) supplies CommuniKate with a
foundation for providing both traditional PSTN services and IP-powered
services to carriers. With Web-based tools, XML, speech technology, standard
SIP protocol, and scalable architecture, these offerings foster the carrier
and service provider deployment of Webley�s voice-enabled unified
communications suite. The switch and CommuniKate together provide a platform
and services that not only converge IP and PSTN communication, but also
their respective services.
Communi-�Kate� is the voice-enabled, phonetically-British, hands-free
virtual assistant tasked with helping its users navigate through hosted
PBX-functionality, voice mail, e-mail and fax, calendar, PIM information, as
well as setting up conference calls, and adding contact information.
CommuniKate�s unified messaging service offers integrated communication
between voice mail, e-mail, and fax retrieval. Listening to e-mail messages
over the phone, or accessing your voice mail via a browser are two reasons
why carriers such as WorldCom have deployed Webley�s solution. Other
messaging features allow users to send, respond to, or forward messages to
anyone else on the CommuniKate powered network with either the touch of a
key or voice command. Additionally users can simultaneously broadcast any
message type one or a hundred recipients while using the address book and
pre-established distribution lists.
Conferencing and Calling Services provide the means to connect directly
with individuals in different ways. Conferencing-On-The-Fly, for example,
allows a user to conference in a phenomenal 32 parties, without having to
set up the call ahead of time. While traditional conferencing options are
available, Webley also offers a Calling Party Pays service, which charges
each party equally for the call. Though conferencing is effective when
people know where to reach you, Calling Services can track you if you�ve
strayed from your office: One Number Reach Me, Find Me Forwarding, Call
Blast (transfers four calls to one destination), and Transfer Rules.
Webley�s CommuniKate is also integrated tightly with PIM management
tools such as an Address Book capable of storing up to 2,000 contacts and a
Calendar that will remind you about its documented events via phone, e-mail,
pager, or simultaneously notify using all three methods. A simple voice
prompt such as, �Call my contact �Jack Walsh,�� instructs
CommuniKate to initiate the call using the phone information stored in your
address book.
Webley assigns the user a toll-free number, which when dialed has the
capability of �call blasting� the user at five possible endpoints. These
endpoints can be any traditional phone, such as a cell phone, office phone,
etc., but the real beauty of Webley is that it supports SIP endpoints.
Ultimately, this allows both PC-based phone and presence features to exist
on users� desktops while enabling call origination and reception without
the use of a traditional handset. Thus, a user could simply enter their SIP
address as one of their possible endpoints and Webley would attempt to reach
the user on any SIP client, such as MSN Messenger or PingTel�s SIP phones.
Since the TMC Labs Innovation Awards are granted to products that are
truly �unique� or innovative, we had yet to award the same company with
a second TMC Labs Innovation Award. After all, it�s difficult enough for a
company to develop just one innovative product � much less two. Thus, when
we gave Empirix an Innovation award for their H.323 Call Generator in the
July 2000 issue, we did not expect to grant Empirix a second Innovation
Award any time soon. Well, Empirix proved us wrong when they launched Hammer
NXT, an integrated carrier class TDM/IP test platform running of a single
21-slot CompactPCI chassis that can generate up to an impressive 50,000
simultaneous calls using a mix of TDM over DS-3 interfaces using SS7, CAS
and ISDN signaling, and SIP/IP with a real RTP media. Multiple NXT platforms
can be combined to achieve traffic volumes of over 250,000 simultaneous
calls.
Running on an embedded dual 1-GHz single board along with a blade-based,
modular architecture, performance and expandability are non-issues. The
system also supports the Hammer Voice Quality Test Suite (VQTS) option,
which provides TDM-to-TDM and IP-to-TDM voice quality testing. It measures
QoS with real voice, including packet loss, latency, and jitter. It also
performs all the standard voice quality scoring algorithms such as PSQM and
PAMS. We�re told subsequent Empirix NXT releases will support MGCP and
MEGACO signaling in addition to STM1 and OC3 interfaces.
We asked Empirix whether or not the Empirix NXT was simply the �old�
Empirix software ported to a more scalable hardware platform. We were
informed that the Hammer NXT utilizes base designs from industry leaders
such as NMS Communications and Advantech that designed and developed custom
versions of their components to address the challenging requirements of
Empirix�s carrier-class converged test platform. In fact, the Advantech
11U height CPCI chassis was designed especially for the Empirix NXT.
In addition, Empirix mentioned that they are using enhanced versions of
the TestBuilder User Interface and Hammer Test Server that are uniquely
designed to address high-density performance testing and deliver converged
test capabilities. We should also mention that the NXT also integrates with
the PacketSphere network-degradation emulator, for simulating real-world
conditions. The Hammer NXT enables engineers to isolate and test network
edge devices with just a single system rather than a plethora of separate
test equipment. The Hammer NXT�s modular, blade-based architecture allows
systems to be configured with flexible mixes of TDM, IP and signaling
capabilities.
With the Hammer NXT, equipment manufacturers and service providers can
verify multiple next gen applications such as toll bypass, and Class 5
enhanced services. Empirix has set a new bar with their high-density,
scalable, carrier-class NXT test platform. Empirix is our first two-time
Innovation Award winner and we couldn�t have selected a more worthy
product.
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