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Innovation is the soul of Internet telephony. And
while we certainly recognize innovation as possessing
the outward appearance of a new product employing a new
technology, we've also recognized that it appears in
other shapes as well. Sometimes being innovative isn't
necessarily being "first" in the most literal sense, but
instead looking at things from a slightly different
perspective. Challenging the established "standards "
and introducing different approaches to achieve
distinctive results certainly helps to define innovation
in our corner of the telecom world, and perhaps even
well beyond.
TMC Labs recognizes the degree of wherewithal
necessary to formulate a quality product that captures
the essence of newer technologies. Building or
re-engineering a product in a new way to enhance
services, improve usability, and refine quality is also
an exceptional and innovative practice. Ultimately, the
industry as a whole stands to gain, as this further
solidifies the stake and acceptance of Internet
telephony as a viable means of communication, ensuring
the provision of a valuable, necessary service.
It was initially the intent to examine our collective
knowledge base (including vendor-submitted nominations)
as Technology Editors for TMC Labs, and nominate
products and (in some cases) product suites, which we
felt demonstrated technological uniqueness with the
promise of providing a viable service within the
industry of Internet telephony. After collaboration, we
researched each product further to not only confirm or
deny initial nominations, but in certain cases to also
corroborate products, which were not unanimously agreed
upon by all editors. Though we examined several
different methods of nomination, ultimately we felt that
sharing knowledge and good old-fashioned democracy were
the best ways to govern the award process.
Many, many quality products were discussed, and many
of the ones that do not appear in this issue were
innovative in their own right. In this case however,
only the highest finishers are eligible for an award.
And that's just what TMC Labs is responsible for doing.
Choosing this year's 12 most innovative
Internet-telephony products. These awards are the
subject of extensive research and a large investment of
time. We hope you find this exercise useful, and we hope
you enjoy the result.
TMC Labs'
Innovation Awards Winners
3100 VoIP
Firewall System
Aravox Technologies
4201 Lexington Ave. N.
Ste. 1105
Arden Hills, MN 55126-6198
Tel: 651-256-2700
Fax: 651-256-2750
Web: www.aravox.com
Over the past year, we have tested many gateways,
gatekeepers, IP phones, and other VoIP products. In most
of these tests, we always had an obstacle to overcome in
order to objectively test the equipment's sound quality.
Because our network is behind a firewall, we could not
accurately pass specific IP packets across the Internet
so that a proper call could be made. In most cases, we
could connect the call but could not hear the person
speaking from the other line. By opening certain ports
on our firewall, we could alleviate this problem.
Unfortunately, this option introduced security issues
because of the breaks in the firewall. Also, the
equipment we were testing was sometimes incompatible
with our firewall.
When we first addressed this problem, there was
really no viable solution out there. Soon after, Aravox
released the 3100 VoIP firewall system, a dynamic
firewall that completely protected the network and
offered a method to deliver reliable IP voice services.
To do this, the firewall sits on the edge of the network
and does not let packets pass unless directed by a SIP
proxy server or H.323 gatekeeper. When directed, the
firewall opens a pathway that only uses the ports needed
for an authorized VoIP call so that the call can be
established with the quality of the call being
preserved. When the call is terminated, the pathway is
immediately closed, thereby eliminating virtually all
security issues.
With this "open and close" technique, the firewall
prevents service disruptions while still protecting the
network from hacker attacks. On a per-call basis,
gatekeepers or proxy servers can manage all VoIP
applications without firewall interference, and billing
can be tracked more readily. In addition, the firewall
performs a Network Address Translation (NAT) capability
that imposes a constant route for voice packets,
ensuring a more consistent amount of quality voice
calls. In these ways, this open and close technique
proves not only an innovative solution, but also an
effective one.
[ Return to list of winners ]
CN1000
Broadband Loop Carrier
Catena Networks
307 Legget Dr.
Kanata, Ontario, Canada K2K 3C8
Tel: 866-2CATENA
Fax: 613-599-0445
Web: www.catena.com
Catena Networks' vision "to create the new access
architecture for the converged, packet-based public
network" intrigued us, so we took a closer look at this
innovative company.
Their CN1000 is a highly integrated broadband access
system for efficiently provisioning and deploying
residential DSL, while enabling the seamless migration
from today's TDM network to a converged, packet-based
softswitch environment on a line-by-line basis.
According to Catena, the challenge is that the DSL
data access network was previously being deployed as a
separate overlay to the voice network. This data overlay
network was satisfactory for the deployment of niche,
business-oriented services, but did not scale for
consumer mass-market deployment, especially since remote
DSLAMs, cross connects, POTS splitters, and adjunct
cabinets are required.
The CN1000 is based on a patented architecture
developed by Catena that terminates the loop at the
first access point and enables carriers to deploy DSL
and converge voice and data in the access network. This
unique line-termination technology integrates POTS and
DSL on every line without requiring a DSLAM or
POTS-splitter at the customer premises.
As a result, every subscriber is broadband-ready from
day one and service can be provisioned remotely without
truck rolls. This new access architecture enables a
graceful, line-by-line migration to voice over packet
(VoP) and softswitch architectures, as carriers evolve
their networks.
This innovative solution allows all subscribers to
continue to enjoy lifeline voice services using their
existing telephone set, while enabling the deployment of
next-generation services. By integrating POTS and DSL on
the same line in the CN1000, Catena is able to
accommodate the 0�4KHz voice spectrum and the 30�1100KHz
DSL spectrum. No integrated access device (IAD) is
needed at the home, because Catena's technology allows
packetization of voice traffic at the line termination
point in the CN1000.
Utilizing the CN1000, a Broadband Loop Carrier (BLC)
could have the following benefits:
- Eliminate DSLAM or POTS splitter and improve DSL
service availability;
- Fully integrate POTS and DSL, obviating the need
to roll trucks;
- Eliminate overlay access networks;
- Supports converged, packet-based network;
- Works in conjunction with packet Class 5 Switches
and softswitches;
- Supports voice packetization, on per-line basis,
at line-termination point; and
- There is no tradeoff of packet voice versus DSL
ports.
According to Catena, their CN1000 broadband loop
carrier scales to more than four times the port density
of existing next-generation digital loop carrier systems
(NGDLCs). Catena's highly integrated POTS+DSL silicon
technology enables 288 integrated POTS/DSL lines on the
first shelf; 672 integrated POTS/DSL lines per two-shelf
system; 1,056 integrated POTS/DSL lines per three-shelf
system; and 2,112 integrated POTS/DSL lines per
seven-foot rack.
The CN1000 supports ADSL and will support the
emerging G.SHDSL standard for high-speed symmetrical
services. Catena's programmable silicon technology
enables carriers to manage evolving ADSL standards by
delivering new features via software downloads. The
CN1000 is currently fully interoperable with leading
providers of DSL customer premises equipment (CPE) and
silicon and is fully compliant with the ANSI T1.413, ITU-T
G.992.1 (G.dmt), ITU-T G.992.2 (G.lite), and ITU-T
G.994.1 (G.handshake) DSL standards.
For remote terminal applications, the broadband loop
carrier differs considerably from current
next-generation digital loop carriers (DLCs).
Next-generation DLCs are based on TDM backplanes and
utilize DSL line cards that typically sacrifice POTS
port capacity. In contrast, Catena's CN1000 is
engineered from the ground up on a packet-based
architecture, which utilizes integrated POTS+DSL line
cards that do not sacrifice POTS density. For providing
an integrated and innovative POTS+DSL solution, which is
easier to deploy, we commend Catena Networks and bestow
our TMC Labs Innovation Award.
[ Return to list of winners ]
Session
Management Suite
dynamicsoft, Inc.
72 Eagle Rock Ave.
1st Fl.
East Hanover, NJ 07936
Tel: 973-952-5000
Fax: 973-952-5050
Web: www.dynamicsoft.com
dynamicsoft is an industry leader in both the
development of Session Initiation Protocol (SIP), and
the deployment of SIP-based products. Led by Chief
Scientist and (SIP) co-author Jonathan Rosenberg,
dynamicsoft was the first to release a comprehensive,
SIP-based solution. That solution is a SIP-based
infrastructure built utilizing open architecture to
foster system customization through implementation of
additional functionality integrating billing, network
management, security, and Web-based provisioning.
dynamicsoft calls it Session Management Suite (SMS). SMS
consists of: A SIP Proxy Server, SIP Location Server,
and SIP Firewall Control Proxy.
SMS employs many APIs to facilitate communication
with other services providing integration with existing
or supplementary network database management systems.
This for example allows domain-level security, the use
of media-based firewalls, and NAT to protect any service
provider's network. In addition, via the Simple Network
Management Protocol (SNMP) and SIP Management
Information Base (MIB), advanced network management
capabilities can be used to manage large SIP networks
centrally, using a compatible network management tool.
The Proxy Server is also capable of generating "billing-related"
information, which can then in turn, either be sent to
the dymanicsoft Remote Logging Server, or a certified
billing intermediation partner. This affords collection
of data, and provides mediation and data storage
capability.
SMS also manages the SIP network on three different
levels: Global, user specific, and user controlled. Each
exercises a separate degree of direction allotting for
deep system customization. Web-based provisioning for
example, allows end-users to control telephony features
such as: Time-based and address screening, call
forwarding, call forwarding on busy, call forward on no
answer, Time-based routing, direct group dialing, and
find me/follow.
The dynamicsoft technical staff employs many of the
world's authorities on SIP. This experience and
knowledge, coupled with APIs, solid infrastructure, and
the NetValue Partner program provide the robustness,
versatility, and the cutting edge that has made
dynamicsoft's SMS both an innovative and complete SIP
solution.
[ Return to list of winners ]
EdgeXpress
Suite
Kenetec
550 Spring St.
Naugatuck, CT 06770
Tel: 203-723-4242
Fax: 203-723-4187
Web: www.kenetec.com
EdgeXpress delivers a comprehensive solution for
enabling multimedia broadband services in a converged
network infrastructure. Targeting MTUs (multi-tenant
units), EdgeXpress, is unique in that it provides an
end-to-end services platform for enabling multimedia
services to businesses over a single converged network.
Kenetec is also the only company that we are aware of
providing a solution in this market space that supports
both ATM and IP. This is critically important in the
carrier market where ATM is still deployed and a fully
deployed IP infrastructure is still a few years away.
The EdgeXpress 5000 is a component in Kenetec's
three-tier QoS architecture (tagging, shaping, and
mapping). To efficiently deliver differentiated services
over a single ATM infrastructure, the EdgeXpress 5000
maps IP class of service (CoS) to ATM quality of service
(QoS) mechanisms. Pure IP infrastructures are supported
via native MPLS or DSCP (TOS), which allows the reliable
delivery of voice, video, and advanced IP applications
over integrated broadband infrastructures. We should
mention that 802.1p is also supported. As far as
scalability, the 5000 comes in three different sizes,
the largest of which has line cards that support up to
120 tenants.
The EdgeXpress 1000 is similar to an IAD (integrated
access device). It sits on the customer premises and
enables the delivery of voice, data, and advanced IP
services to small- and medium-sized businesses located
within a multi-tenant unit (MTU). The EdgeXpress 1000
can support up to 16 POTS voice lines or two DSX-1
interfaces. Internet access is available through a
10/100 BASE-TX LAN interface. The standards-based
EdgeXpress 1000 converges both voice and data traffic
onto a single copper or fiber riser.
Existing telephones can be plugged into the
EdgeXpress 1000 unit directly or interconnected from a
PBX. The EdgeXpress 1000 can then process it as an
ATM-based AAL2 voice packet for delivery to service
providers providing ATM backbones. The EdgeXpress 1000
complies with the MGCP and SIP gateway standards for
support of VoIP services. Thus, traditional voice
services and corresponding revenue can be made right up
front. Later on, enhanced services such as VoIP can be
added after the fact, when the service provider is ready
to exploit packetized voice services.
The EdgeXpress family is managed by Kenetec's Edge
Manager -- a comprehensive management system with
advanced service management features, as well as
financial functions to demonstrate the return on
investment. Together, the EdgeXpress 1000 and EdgeXpress
5000 allow service providers to create services-oriented
building area networks (SBANs). Service providers
targeting businesses within multi-tenant facilities
really need to be able to deliver services other than
just Internet access in order to generate new revenue
streams and prevent customers from going to competitors.
As a converged solution that allows for traditional
voice services today, as well as enabling migration to
future enhanced services, Kenetec's EdgeXpress product
suite really wowed TMC Labs. That, combined with support
for both IP and ATM protocols as well as its impressive
provisioning and administration tools, helped make this
a truly innovative solution.
[ Return to list of winners ]
Instant
Broadband EtherFast Cable/DSL & Voice Router
Linksys
17401 Armstrong Ave.
Irvine, CA 92614
Tel: 949-261-1288
Fax: 949-261-8868
Web: www.linksys.com
There are many broadband Internet sharing devices on
the market today. What is innovative about Linksys
Instant Broadband EtherFast Cable/DSL and Voice Router
is that not only does it act as an Internet/data sharing
device, but it features Voice over IP capabilities as
well. With the router installed, no other special
hardware is necessary to make telephone calls. For
instance, there is no need to boot up a PC to make a
VoIP call. Using just an ordinary analog telephone
connected to the RJ-11 port on the back of the router,
calls are routed across the Internet through Net2Phone's
network to a phone located anywhere in the world -- at
significantly reduced long-distance charges. Since the
router works with any analog devices, you can even use
your favorite cordless phone. Other significant features
include four switched 10/100 Ethernet ports, IPSec and
PPTP Pass through, remote administration and remote
management for more upgrades over the Internet, and DHCP
server functionality to assign IP addresses. The Linksys
router supports several standards including PAP, CHAP,
PPP, and PPPoE. Like most Internet sharing devices, the
Linksys product also sports an easy-to-navigate
Web-based administration GUI.
Although only four ports are available, the router
supports up to 253 PCs on the network by daisy chaining
other network switches and/or hubs. Of course this
router also acts as a firewall, with configurable
policies to protect your internal LAN. With so many
competing Internet sharing devices on the market today,
VoIP just may be the differentiating factor when making
a purchasing decision. As such, we commend Linksys for
combining data and voice onto a single platform, and
thus we proudly bestow our TMC Labs Innovation Award to
the Linksys Instant Broadband EtherFast Cable/DSL and
Voice Router.
[ Return to list of winners ]
iTeleFusion
Nissi
21515 Hawthorne Blvd.
2nd Fl.
Torrance, CA 90503
Tel: 310-792-2000
Fax: 310-792-3128
Web: www.nissi.com
TMC Labs recently discovered an interesting company,
NISSI, with an innovative product. Nissi has designed an
end-to-end convergence communications platform targeted
at service providers, ASPs, contact centers, and
enterprises. Their iTeleFusion product provides myriad
Internet and telecommunications applications, including
unified messaging (voice, fax, and e-mail), PIM,
voice/fax over IP, video/audio/text conferencing, and
instant messaging (IM) -- all over an IP network, an
intelligent network and the PSTN. This comprehensive
solution offers a complete Internet convergence platform
for communication, collaboration, and information
management.
iTeleFusion can be configured to meet the specific
needs of any service provider or enterprise. The user
client can be PC-based, Web-based, or WAP-based. Users
can easily communicate with one another using any
communication media they choose whether it be voice,
fax, video, VoIP, WAP, IM, etc., all from a single user
interface from anywhere in the world. Also, through
iTeleFusion's telephone user interface (TUI) or WAP
interface, users can access messages and information
from any telephone or WAP-enabled device.
NISSI likes to define their product as "total
communication convergence with complete communications
integration." After seeing the plethora of open
standards that the iTeleFusion platform supports, we
couldn't agree more. Standards such as H.323, SS7, WAP,
T.37/T.38 faxing, and 3G, as well as T1/E1 and ISDN are
supported. Their platform also fits nicely into existing
legacy hardware, including PBXs, Web servers, and e-mail
servers. Also, the system can be integrated with the
SS7, IP, and softswitch networks with NISSI's SS7 and
VoIP modules. NISSI has also developed an OSS tool for
easy management and provisioning.
NISSI also offers optional VoIP gateways,
gatekeepers, a FoIP gateway, and other optional modules.
One of the key differentiators of iTeleFusion is that it
is designed on NISSI's iTOS (Internet Telephony
Operating System). By using the iTOS platform across
NISSI's complete product line, integration with other
modules is much easier and more seamless. For example,
Nissi's ContactPrime, a product targeting the multimedia
contact center, is an optional product also developed
with iTOS. The distributed architecture and iTOS
platform allow for carrier-grade stability and
carrier-class scalability with customization support for
service providers and customers. Besides the modular
design, scalability, and plethora of standards support,
we also like the fact that NISSI provides an end-to-end
comprehensive solution, which is certainly innovative in
its own right.
[ Return to list of winners ]
Tenor
Carrier MultiPath Switch (CMS)
Quintum Technologies, Inc.
14 Christopher Way
Eatontown, NJ 07724
Tel: 1-877-SPEAK IP
Fax: 732-544-9119
Web: www.quintum.com
For the October issue of this magazine, we tested
Quintum Technology's Tenor
Gateway. In the review, we stated that the product "intelligently
switches calls over both IP networks and the PSTN. If IP
network congestion or a device failure impacts voice
quality, the Tenor Gateway can switch calls over to the
PSTN." No other VoIP product that we know of does this,
and what's more, the switch from one network to the
other occurs unbeknownst to the caller. Furthermore, it
acts both as a gateway or a gatekeeper; and with the
high QoS level when using their Transparent Auto-Switch
Quality (TASQ) technology, there was even more reason to
consider awarding Quintum.
However, with all the competition, we were still not
completely sold -- that is until we found out about
Quintum's new Tenor Carrier Multipath Switch (CMS),
which is the next version of the Tenor Gateway. Besides
continuously monitoring the data network for packet
loss, latency, and jitter and transparently switching
from VoIP to the PSTN for when the data network goes
below a particular quality threshold, the Tenor switch
offers enhancements to TASQ via a new patent-pending
technology called SelectNet. SelectNet provides call
protection as well as its other call management, QoS,
address translation, and security functions to IP
end-points that don't need to pass through the switch's
IP gateway.
The Tenor switch can also combine voice packets from
several calls into one packet by maximizing the data
utilization of the packet. Because of this feature,
bandwidth overhead can be minimized. Quintum calls this
their PacketSaver technology. As if all of this wasn't
enough, we also noticed the increased scalability of the
Tenor switch. While the configurations could use as
little as four trunks, the Tenor switch could scale up
to 32 trunks within a single chassis.
Our earlier tests indicated Quintum's fine
craftsmanship in their Tenor product, and the new
equipment reveals a well-built switch and even more
ingenuity. By the time we concluded our examinations of
the new switch, we were sure that Quintum deserved to be
honored with our Innovation Award.
[ Return to list of winners ]
SS8
SignalingSwitch
SS8 Networks, Inc.
2025 Gateway Pl.
Suite 200
San Jose, CA 95110
Tel: 408-501-2100
Fax: 408-501-2101
Web: www.ss8.com
The SS8 SignalingSwitch is the core of SS8 Networks'
product suite. SS8's switch is spun from an innovative,
forward-thinking vision and market-leading architecture,
aiding in the creation of a new standard for IP
Signaling Transfer Point (IP-STP). SS8 has created this
product by taking aim at uniting existing protocols and
delivering Intelligent Network (IN) Services across
gateways, proxy servers and softswitches, thereby
linking together smaller, disparate networks to help
form what can be recognized as the public internet
telephone network (PITN).
The SS8 SignalingSwitch combines the ability to read
both SIP and H.323 verse and services, creating a
platform engineered as a solution for lack of
interoperability between the two incumbent protocols.
The switch effectively renders readability for both "standards"
and provisions service through one to the other --
thereby relieving contrast between networks. Since the
SS8 SignalingSwitch is able to recognize and translate
both protocols synchronously, it is also capable of
connecting devices employing different protocols, from
different vendors, affording new levels of
interconnectivity. The SS8 SignalingSwitch also utilizes
high-speed, multi-processor architecture and
hot-swappable components aimed at delivering the highest
reliability and best system performance.
The SS8 SignalingSwitch is inclusive of what SS8
bills as the industry's first standards-based advanced
IP Telephony Routing Engine (ITRE). The ITRE provides
the necessary groundwork for advanced call routing
within an intelligent network (IN) such as: IP routing
by telephone number, automatic IP telephony route
discovery and updating, policy-based telephony routing,
automatic alternate routing of signaling messages (fault
tolerance), and more.
SS8 Networks has the vision of melding the best of
the PSTN and IP to idealize the now-emerging PITN. SS8
has targeted a new standard of carrier-class service
with the SS8 SignalingSwitch and its siblings in the SS8
Networks' product suite, which are engineered to meet
the high demands of real-time signaling necessary to
support telephony and video applications. It is both
their visionary forethought and timely follow-through
that make SS8 Networks' SignalingSwitch product a clear
nominee -- and winner -- of an Internet Telephony
Innovation Award.
[ Return to list of winners ]
Call
Agent
Telcordia Technologies
445 South St.
Morristown, NJ 07960-6438
Tel: 800-521-2673
E-mail: m-webleads@notes.cc.telcordia.com
Web: www.telcordia.com
During the last twelve months, there has been much
talk about softswitches that can provide VoIP
applications without the need for a Class 5 circuit
switch. These types of softswitches include traditional
telephony features such as call waiting and call
forwarding, more advanced functionality such as video
conferencing and integrating voice and data or voice
with the Web, and an open architecture that eliminates a
dependence on specific switch suppliers. These
next-generation switches can save companies a lot of
money and can even increase revenue through their
ever-expanding offerings. However, the older Class 5
switches are still engrained as the major part of
service provider's equipment, who therefore have a
vested interest in their legacy systems. Thus, these
providers are hesitant about adopting the new
softswitches. While there is much hoopla about either
replacing the Class 5 switches or bridging softswitches
with the existing equipment, most have still not been
deployed within a provider's network.
Because of its excellent feature set and its ability
to interoperate with IP or ATM gateways to perform call
control functions and deliver advanced services,
Telcordia's Call Agent was the first to begin to
overcome this obstacle and be deployed and run by major
providers CTC Communications and Sprint. With these two
companies deploying Call Agent to quickly and
efficiently add value to each company's disparate
network, VoIP services between internal offices have
already been established. Now Sprint and CTC can begin
supplying a plethora of new services to end users that
Class 5 switches cannot deliver.
All in all, the innovation inherent in softswitches,
in which Call Agent was one of the first and still one
of the best, is important, but that is now somewhat old
news to us. This year, we honor Telcordia not only for
Call Agent's offerings but more for their visible
leadership and experience in a young industry that needs
these qualities most in order to truly revolutionize the
communications world.
[ Return to list of winners ]
Media
Xchange Manager (MXM)
VCON, Inc.
10535 Boyer Blvd.
Suite 300
Austin, TX 78758
Tel: 512-583-7700
Fax: 512-583-7701
Web: www.vcon.com
Just last month, we reviewed VCON's Media
Xchange Manager (MXM) for this magazine. During the
review process, it became quite apparent to us just how
innovative this product is. That's because the MXM is
more than just a typical H.323 gatekeeper, mainly
because of its extensive administrative, management, and
monitoring features available for most IP and H.323
applications with its focus on video conferencing. It
supports a wide range of equipment, including multipoint
control units (MCU), gateways, and video endpoints, and
an administrative graphical interface can be installed
onto any PC on the network so that management and
monitoring functions can be achieved remotely.
What we found to be the main attraction of the MXM is
its centralized administration and the usability of the
GUI. The administration is centralized via the GUI,
which during our testing, we noticed was very intuitive
and laid out in a clean, hierarchal manner. Sometimes,
as in this case, innovation can take the form of
usability, especially since some of the most
feature-rich gatekeepers are awkward to use at best.
Fortunately, the centralized administrative GUI of the
MXM provides secure, mobile access to support multiple
consoles and servers. This remote GUI is where all of
the H.323 equipment on the network is identified and
where remote endpoint and dial plan with hunting group
configurations can be performed.
Other important functionality of the MXM that
impressed us was the centralized management and the
video-oriented telephony services. The centralized
management allows for conversation status monitoring,
event logging, registration and admission control, and
address translation capabilities. The telephony services
included ACD functionality, call forward, call transfer,
and call pickup that can all be performed while keeping
the video intact.
The beauty of all of this is that all VoIP (or at
least H.323 applications) can be monitored and managed
on the MXM from any PC. Any supervisor or administrator
in an organization can keep track of the applications
being used -- what extensions are logged in or not and
who is on a call or video conference. While this may not
be a new idea, there are a few in the Internet telephony
industry that have even attempted this and none that
have done this so well.
[ Return to list of winners ]
�Ensemble!
VIVE Synergies, Inc.
150 West Beaver Creek Rd.
Richmond Hill, ON, Canada L4B 1B4
Tel: 905-882-6107
Fax: 905-882-6238
Web: www.vive.com
VIVE Synergies' �Ensemble!
not only performs as an H.323 gateway, but also delivers
many other communications-enabling features critical to
servicing customers in a SOHO or SMB environment. The
�Ensemble! (about the size of a digital cable box) with
easily upgradeable firmware, also functions as a premise
PBX, is capable of producing CDR records and billing
information, and is equipped with "VoIPower." VoIPower
is VIVE's answer to affordable, Web-based call center
functionality.
Though the �Ensemble! can provide an entire
communications system for a SOHO or SMB, larger
companies with greater PBX demands (and incumbent PBX
systems) can just as easily implement the �Ensemble!
into their corporate environment in order to provide
e-customer service on any Web site. This can include
implementing a "click-to-talk" button on any Web page
providing customers with a conduit to connect with a
live agent via VoIP. The latest firmware release also
offers code for co-browsing, two-way applications
sharing, and text chat.
Remote offices, companies with branches abroad or
even domestically, may consider �Ensemble! for its
gateway functionality alone. A unit at each office would
not only provide a low-cost alternative to the PSTN for
communication from venue to venue, but callers could
also hop off to the local PSTN or employ the optional
gatekeeper software for gateway management and
least-cost routing. The �Ensemble! also works in
conjunction with the Net2Phone network.
VIVE Synergies has also recently upgraded their
product, making it both SIP and H.323 compliant. SIP and
H.323 clients are now available in the form of a plug-in
on the host Web site, which no longer requires the
customer to interface via NetMeeting. Additionally,
�Ensemble! functions independently of any PC employing
its own, built-in Web server. VIVE also says that
interoperability tests have been conducted and prove
that their product works in conjunction with Cisco VoIP
gateways and gatekeepers.
In conclusion, the �Ensemble! was selected for an
award because of its feature set, broad appeal, and
great value, which was packaged in a way we've not seen
before. Whether you've got a three-person operation and
you need a communication center; or you're part of a
Fortune 500 company in need of e-customer service, the
�Ensemble! may be the solution for your company. The
innovation in this product is not in the advent of a new
protocol or first-version software, rather it's in the
big business functionality -- with mass appeal -- at a
price a small business owner can afford. If you're
looking into a device for all, or even just some of the
aforementioned functionality, check into "VoIPower." You're
likely to be impressed.
[ Return to list of winners ]
WebInteract
Service
WebDialogs, Inc.
Concord Road Corporate Center
300 Concord Rd.
Billerica, MA 01821
Tel: 978-439-9600
Fax: 978-439-9962
Web: www.webdialogs.com
In a shoot-out
staged in January's issue of this publication, TMC Labs
compared ASP "Push to Talk" services from eStara, HearMe,
Lipstream, and WebDialogs. When the smoke cleared, it
was difficult to determine who stood above the rest as a
clear "winner." TMC Labs determined that while the
contest was close, WebDialogs clearly distinguished
itself by virtue of sheer feature-richness, offering
crucial bi-directional co-browsing and form-sharing
features where some or all its opponents did not -- and
this is in addition to full-duplex operation, text chat,
PC-to-phone call capability, and good VoIP quality when
used in conjunction with the then-beta Net2Phone client.
The value of their PSTN call back option also stood out,
not only in the immediate sense -- for less technical
consumers who don't own (or own but don't know how to
use) multimedia PCs -- but also for the long term, by
providing a comfortable, familiar means to transition
the public towards more VoIP in the home and in the
industry.
Integrated with an Internet Explorer 5 window in the
bottom of the screen, the agent interface was another
feature that distinguished WebDialogs' service (called
WebInteract) from their peers. From the screen's top
half, agents can connect to incoming VoIP calls or
initiate a PSTN callback. Once connected, the agent has
several options at his disposal, including the ability
to take "snapshots" of particular windows and transmit
them to the customer as images, to initiate file
transfers, or to engage in text-chat. As for co-browsing
and form-sharing, WebDialogs took a particularly
effective approach to these collaborative features --
which provide consumers with the often critical level of
comfort that can remain the difference between a sale
and the all-too-often abandoned shopping cart.
Co-browsing is bi-directional, meaning both agent and
customer can push Web pages to each other (a feature not
often or always found in co-browsing applications) and
can be extended into form sharing capabilities allowing
the agent to assist customers in filling out online
forms.
Innovation is often a question of firsts (i.e., a
company was the first to offer this feature, to take a
different path towards a known goal, to offer services
to a new
market, etc.). Where ecommerce and VoIP are concerned,
innovation could be defined by WebDialogs' refusal to
skimp on offering agents all of the resources needed to
provide a good and productive customer experience. This
requires a lot more than simply throwing as many tools
as possible into a single box, but rather ensuring they
can work together through the application of creativity,
drive, and sophistication.
[ Return to list of winners ]
[ Return to
the July 2001 table of contents ]
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