TMC Labs Internet Telephony
Innovation Awards 2004: Part I
BY THE STAFF OF TMC LABS
our fifth installment, the popularity of the TMC Labs Innovation Awards
seems to grow by leaps and bounds each year. This year weï¿½ve seen
double-digit growth in the number of applicants over last year, which is
certainly indicative of the tremendous growth VoIP is experiencing. That
begs the question, should we increase the number of ï¿½innovative
products/servicesï¿½ to accommodate this increase in applicants? Well, TMC
Labs doesnï¿½t look at it that way; we just try and pick the top innovative
products within the Internet telephony/VoIP industry without having any
particular number in mind.
In fact, weï¿½ve often been asked, ï¿½Why doesnï¿½t TMC Labs have a ï¿½TMC Labs 100ï¿½
award given to the top 100 players in the VoIP market?ï¿½ Well, our answer is
we donï¿½t need to create a list of 100 VoIP companies based upon the fact
that the company has good revenue, a high market cap, or is a leading VoIP
player. You can easily find out the leading VoIP players on the Web, in this
magazine, by performing some stock quote lookups as well as several other
methods, so we donï¿½t need to rehash that information.
The purpose of the TMC Labs Innovation Awards is just as the name states ï¿½
itï¿½s about awarding ï¿½innovation.ï¿½ We grant Innovation awards based solely
upon how unique or innovative a particular product or service is. We donï¿½t
care if company XYZ sold 1,000,000 widgets or 0 widgets, as long as the
concept of the widget is innovative.
Nevertheless, our task to find the truly innovative products and services
was quite difficult this year. We had our hands full with a plethora of
applications, making the judging a very difficult decision. In fact, we have
two very controversial selections involving Digium/Asterisk and Pingtel that
resulted in some heated exchanges between all of our judges. Weï¿½ll discuss
that later in this article.
In addition, we had several applicants all within the same ï¿½bandwidthï¿½ genre
(i.e., improves bandwidth, maintains QoS, or performs ï¿½traffic shapingï¿½).
The importance of having the maximum amount of bandwidth available and to
optimize that bandwidth with QoS is very important to enable you to add more
services and applications such as VoIP and video. Well, it certainly was
difficult to judge which products within this same genre were more
innovative since they all did something unique but in a different way.
For 2004, we proudly bestow 27 winners (detailed write-ups) and three
honorable mentions, which will be published in two parts in order to
accommodate our in-depth write-ups for the winners. The complete winners
list will be published in both issues, however we will write the detailed
write-ups in alphabetical order beginning with Actelis this month and ending
with Emergent Network Solutions. Next month, we start with Empirix and work
our way through the alphabet to Ubiquity Software. We hope you find these
products as innovative as we did.
we think of 3Com within the VoIP space, we think of their excellent NBX-100
IP-PBX, which is targeted solely at the small-to-mid sized enterprise and
not large organizations. Weï¿½ve never thought of 3Comï¿½s IP telephony product
line as targeting larger enterprises. In fact, one of the knocks of 3Comï¿½s
NBX-100 and 3Comï¿½s VoIP product line in general was that it didnï¿½t scale
well. Well, 3Com has changed all that with the 3Com VCX V7200, which targets
large enterprises with multiple branch offices due to its ability to scale
to more than 50,000 users. 3Com describes their VCX V7200 as ï¿½the first
SIP-based, carrier-proven softswitch product scaled down to serve the needs
of the large enterprise customer.ï¿½
We should mention that several mechanisms exist to connect multiple IP-PBXs
together, but itï¿½s more of a kludge, and they donï¿½t provide a single
provisioning and management interface that makes these multiple entities
exist as one logical voice system with a common dial plan.
3Com has developed a SIP-based Voice Boundary Routing architecture that
allows for multiple remote locations, along with centralized provisioning,
global dial plans, and company-wide voice messaging. They also claim it is
lightweight, fault-resilient, and real-time.
Voice Boundary Routing deploys call routing in a unique way in that it
provides both local routing and a centralized routing intelligence. Local
call routing decisions are determined at the local level in accordance with
accepted best practices, i.e., least cost. However, 3Comï¿½s architecture
allows for a more powerful central call controller that retains images of
all of its associated local controllers.
A very obvious advantage of the 3Com method of call routing is the ability
to efficiently enforce least cost routing and toll bypass rules. A less
obvious advantage is the fault tolerance such an architecture would instill,
including the ability to use local call routing rules in the event the
centralized call controller fails or in the case of an IP WAN failure.
In the situation where an outbound call is initiated and the local call
routing rules do not have applicable information, it is still possible that
the call could be routed over IP by one of the other branches. In this case,
the call processor where the call is initiated forwards the call request to
the central call controller.
The central VCX V7200 can then search through its cloned routing tables for
all the branches. If the appropriate routing rules are handled by another
call processor at a different branch, it returns the IP address of that call
processor to the originating device. This call is then routed to the local
branch where the local rule set is invoked and the local call processor
sends the call out to the PSTN. This preserves the least cost routing and
toll bypass advantages of IP telephony.
like bandwidth, right? We know we do ï¿½ we canï¿½t get enough of it! Thatï¿½s why
when we heard about Actelisï¿½s MetaLIGHT 1300 copper-based solution we were
pretty excited. Forget about that paltry single-line DSL speed over a single
pair of copper or even the 1.544Mbps bandwidth of a T1, which also runs on
copper. Amazingly, Actelisï¿½s MetaLIGHT 1300 screams at 40Mbps simply using
several bonded pairs of copper as opposed to fiber. On top of that, it
actually runs standard Ethernet over the copper for a transparent, nearly
ï¿½plug and playï¿½ installation for customers.
Actelis claims to be the first and still the only company to release a
Point-to-Multipoint Ethernet over Copper platform (there are a few
point-to-point Ethernet over Copper solutions). The MetaLIGHT 1300 platform
can deliver up to 40Mbps of Ethernet services to up to 64 sites from one
centralized platform. The platform can deliver Ethernet services to
locations more than 18,000 feet away ï¿½ all over copper, which is not only
more ubiquitous than fiber, but itï¿½s also less expensive.
Although there are other bonded broadband over copper technologies, Actelis
is the first company to provide high-quality (low error-rate),
high-bandwidth broadband services over multiple bonded copper pairs to
distances beyond 18,000 feet. Since the MetaLIGHT 1300 delivers true
high-quality Ethernet services (typical Bit Error Rate of 10-10 of better),
it is ideal for CLECs, RBOCs, service providers, and enterprises who are
deploying packet-based applications such as voice and video over IP, etc.
The MetaLIGHT 1300 platform is the industryï¿½s first Ethernet over Copper
platform designed to comply with the emerging IEEE 802.3ah Ethernet in the
First Mile (EFM) standard. It also utilizes standards-based G.SHDSL line
code technology which is enhanced with Actelisï¿½ patented MetaLOOP
you are a CEO, CFO, CTO/CIO, or telecom manager, then you have probably
experienced the headache of trying to figure out the monthly telecom costs.
Reading a typical telecom bill (voice/data circuits) is like reading
hieroglyphics with all the discounts, fees, taxes, and seemingly random per
minute charges. Keeping track of whether you are being billed correctly and
honestly (because phone companies such as WorldCom are never involved with
scandal or dishonesty, right?) is nearly impossible. Well, AnchorPoint 5.0
aims to change all that.
AnchorPoint is a Telecom Financial Management (TFM) solution that is used in
all levels of the organization ï¿½ from telecom managers who audit bills, make
purchasing decisions, manage VoIP, PBX, and wireless usage and negotiate
vendor contracts; to business unit executives who are accountable for the
costs consumed by their departments; to financial managers and CFOs who need
data and trend analysis for budgeting, forecasting and cost-saving
initiatives; to CIOs who have to contain IT budgets and align their costs
with business objectives.
The AnchorPoint TFM application suite is comprised of six fully integrated
Telecommunications Financial Management (TFM) applications: Inventory
Management, Invoice Management, Usage Management, Chargeback Reporting,
Change Management, and Enterprise Application Integration.
The TFM business process lets you institute procedures and guidelines that
ensure that: goods and services are purchased only if required; vendors
deliver what was ordered; vendor invoices reflect the terms, conditions, and
prices agreed upon; and the individual or business unit ordering the goods
or services is accountable (charged) based on consumption.
AnchorPointï¿½s TFM applications help companies get a true picture of their
voice and data spend, as well as create streamlined business processes for
maintaining an accurate inventory of voice and data assets, processing
vendor invoices, providing reporting on VoIP, PBX and wireless usage,
delivering timely telecom chargeback information, and providing historical
information for budgeting and forecasting.
AnchorPointï¿½s view is that VoIP is bigger than simply a ï¿½killer app.ï¿½ The
process and approach that AnchorPoint sees companies adopting with great
success is the deployment of the TFM technology to reduce infrastructure
costs prior to implementing VoIP shaving 10ï¿½20 percent off of their telecom
costs. The TFM applications enable companies to eliminate over-billing and
waste, and get rid of unnecessary resources. It also establishes a baseline,
both financial and physical, for telecom infrastructure. Once VoIP is
implemented, companies can continue to maintain this baseline of cost by
making sure waste wonï¿½t creep back in.
Touchstone Telephony Port
TMC Labs staff is a huge fan of ATAs (analog telephony adaptors), including
the Cisco ATA-186, the Grandstream Handytone-486, the Motorola VT1000, and
the Sipura SPA-3000. They are typically one- or two-port VoIP gateways, or
more specifically, they allow you to plug your analog phone into the unit
allowing your voice to be packetized across an IP network (the Internet,
LAN, or WAN) which terminates at an ITSP (Internet Telephony Service
Provider), such as Vonage, Packet8, Net2Phone, etc.
Well, now the cable companies have been itching to get into the voice market
for years and with the explosion of cable broadband users, cable operators
are certainly looking to expand their offerings beyond just cable TV and
broadband Internet. ARRIS addresses the cable operators need by offering an
ï¿½Outdoor Network Interface Unitï¿½ for PacketCable-based VoIP and high-speed
data access as well cable TV access. This is similar to the embedded
multimedia terminal adapters (E-MTAs) that many cable operators are
deploying indoors today for VoIP.
ARRIS claims to be the first commercially available outdoor E-MTA called the
Touchstone Telephony Port (TTP). Interestingly enough, their initial model
was available in a four-line unit while the latest reflects the more common
need for just two lines. Another innovation is the fact that the TTP was
intentionally designed to be placed outside of the home or business in a
manner similar to what RBOCs use for providing telephone service. This makes
installation and maintenance of voice service much easier because the
operator does not need to enter the premise to install or upgrade the
While the technically-savvy minded folks would scoff and say, ï¿½I can install
an internal ATA in two minutes,ï¿½ one must consider the considerable market
of grandmas, grandpas, and other technically challenged individuals who
think a firewall is a fire-proof wall and that NAT is something you swat at!
So certainly the TTP addresses a vastly underserved market. In addition to
being located outside the residence, the Touchstone Telephony Port also
provides a network controlled CATV relay which is not typically available on
an indoor E-MTA.
Asterisk/Digium and Pingtel
is our most controversial Innovation award selection of all time. TMC Labs
had some pretty heated battles amongst all of the judges to determine who is
indeed more ï¿½innovativeï¿½ ï¿½ whether itï¿½s Digium or Pingtel ï¿½ two similar
First a backgrounder. Digium is the creator of and primary developer of
Asterisk (www.asterisk.org), an open source Linux-based IP-PBX that you can
download for free. Tom Keating wrote about Asterisk in 2001 before Asterisk
started making waves in the VoIP community. You can check it out here:
In Asteriskï¿½s short existence, they have quickly harnessed a loyal developer
community that has contributed to the source code. Digium has taken
Asteriskï¿½s source code and added their own hardware and code to create a
working IP-PBX system.
The judges in favor of granting Asterisk/Digium the TMC Labs Innovation
Award argue that Digium was the first to create a working IP-PBX based on
open-source and leveraging the open-source community. After all, isnï¿½t being
the ï¿½firstï¿½ the very definition of being ï¿½innovativeï¿½? In addition Asterisk/Digium
supports a plethora of protocols including IAX, MGCP, H.323, and SCCP
(Skinny) VoIP protocols as well as T1, E1, PRI, ISDN, BRI, FXO, FXS and RBS
signaling. Furthermore, Asterisk is the first PBX to use Bluetooth for
establishing user presence. Asterisk contains the first Open Source PRI
stack to be certified for use in the U.S., Europe, and Australia. Asteriskï¿½s
Zaptel driver framework is the first open source host media processing (HMP)
suite providing DTMF detection, and echo cancellation.
Our other judges countered with some good arguments of their own in favor of
Pingtel. They noted that Digium is not 100-percent standards-based, it
doesnï¿½t have full SIP support yet, and they pointed out that Digium utilizes
their own proprietary hardware ï¿½ which is where they make their money. They
also noted that Asterisk, which Digium is based upon, has no core developers
ï¿½ itï¿½s all volunteers, whereas Pingtel not only intends to leverage the
ï¿½volunteer developer communityï¿½ but they also have core developers under
salary. The judges favoring Digium countered that Digium does have paid
employees that contribute to the Asterisk source code.
The Pingtel SIPxchange proponents also pointed out that Pingtel is 100
percent standards-based, has native SIP support, and that SIPxchange is 100
percent software (no hardware required). In fact, they argued that Pingtelï¿½s
SIPxchange is not tied to any proprietary hardware at all ï¿½ it can work with
third-party SIP phones, third-party SIP gateways, or even third-party
servers, such as media gateways. So, for example, you can use SIPxchange
with a Cisco gateway and a snom IP phone. Digium on the other hand does
require the use of their proprietary hardware.
What we also found ï¿½innovativeï¿½ about SIPxchange was that Pingtel is
attempting to work out how to survive being a commercial company utilizing
open source development. This is not an easy task ï¿½ especially considering
someone can download Pingtelï¿½s SIPxchange software for free, install it on a
Linux box, and get it to work with a third-party SIP gateway without paying
Pingtel a dime!
In fact, we asked Pingtel what is to prevent users from just downloading the
software and not pay Pingtel anything. They said that they feel that the
majority of users will pay the license fees for several reasons, including
access to the documentation, technical support, as well as pay for the
enhanced codecs that are better than the open source version.
Our Digium proponents kept coming back to ï¿½Yes, but who was the first to
successfully build and sell an open-source-based IP-PBX? Who was the
innovator?ï¿½ Our Pingtel proponents couldnï¿½t dispute this claim. So TMC Labs
was stuck in a quandary ï¿½ both Digium and Pingtel were both ï¿½innovativeï¿½ in
their own right, with ardent supporters in both camps.
TMC Labs does not like to grant an Innovation Award to two very similar
products. It goes against the very nature of ï¿½innovationï¿½ if two or more
companies do almost the same exact thing. After examining the cases for both
sides, and with he votes tied amongst the judges, we decided that both
Digium and Pingtel were both worthy of an Innovation Award and because they
were innovative for different reasons.
Digium was innovative for being first to market with an open-source IP-PBX
and Pingtel was innovative for taking the open-source model, extending it,
and offering a 100 percent standards-based solution whose business model is
completely predicated on software license fees. Further, Pingtel really has
opened the door to allowing customers to choose ï¿½best of breedï¿½ hardware to
work with their software.
IPM-260 8 E1/T1 PCI VoIP Board
Within the VoIP market, especially within the carrier, service provider, and
ITSP marketplaces, scalability/port density is absolutely key. Well,
AudioCodes is the first to market with the IPM-260 8 E1/T1 PCI VoIP board
which features eight T1/E1s in a single PCI slot. Currently, their
competitors are at four T1s maximum per slot. OEM providers that build
systems for IP-enabled contact centers, media servers, conferencing servers,
PSTN, and IP IVR solutions certainly have a need for high port densities and
the IPM-260 certainly addresses their need.
specifications include up to 240 conferencing resource capabilities (IP, TDM,
PSTN), up to 64 active participants per single conference, up to 80
conferences per board. Other conferencing features include Moderator, Active
participants, listener only, whisper coach, mute, DTMF mute, and many more.
It also support record/playback, voice/energy detectors, DTMF detectors, and
voice and/or speech barge-in detectors. It also supports a plethora of
industry standards including MGCP, MEGACO, SIP, H.323, and T.38. It supports
various voice coders such as G.711, G.723, G.729, and more.
This board, which is a complete media server on a blade, can be controlled
via the PCI or via the network. The new IPM-260 offers enhanced performance
and double the existing channel density of similar products in the market,
enabling developers to maintain the same configuration, infrastructure and
size for systems with increased capacities while at the same time reducing
power consumption ï¿½ often a critical consideration in carrier deployments.
CITELlink IP & SIP Handset Gateway
we previously mentioned, we love analog telephony adaptors (ATAs) that
enable you to utilize your analog phone to packetize and transmit voice over
IP. But, what about the surfeit of existing digital telephones? Wouldnï¿½t it
be great if corporations could enable their remote workers to use one of
their existing digital phones from say Avaya, NEC, Nortel, etc. and it would
work from their home broadband network? That is to say, instead of an ATA,
wouldnï¿½t it be great if a DTA (Digital Telephony Adaptor) existed that
allowed you to use digital phones across an IP network? Well, Citel
Technologies has such a product called the CITELlink IP and SIP Handset
Citel Technologiesï¿½ CITELlink IP and SIP Handset Gateway enables traditional
digital handset users to benefit from enhanced VoIP and SIP telephony
applications. Citel claims that CITELlink is the worldï¿½s first handset
gateway to provide IP PBX and SIP interoperability for legacy PBX telephones
across multiple vendor lines. What is truly innovative is that the CITELlink
gateway communicates in legacy PBX protocol to traditional telephone
handsets and Internet Protocol (IP) to the IP PBX and SIP to Hosted or IP
Centrex platforms, and then it performs the translation between the two
simultaneously across multiple vendor products.
Essentially, the CITELlink gateway bridges the worlds of legacy and
next-generation IP and SIP applications opening the previously closed PBX
market ï¿½ by linking the digital telephone handsets of one vendor with an IP
PBX and SIP platform of another vendor. With CITELlink, Citel has just put
the final nail in the coffin of closed proprietary PBX solutions!
Clarus Systems, Inc.
There are various tools out there to test VoIP, however, Clarus Systemsï¿½
Clarus IPC Assurance tests VoIP from a different and unique angle. Clarus
Systemsï¿½ ClarusIPC Assurance simplifies and automates VoIP management tasks
through its unique ï¿½end-user levelï¿½ approach. ClarusIPC Assurance is an
enterprise software-only solution that utilizes voice quality measurement
technology found only in expensive hardware designed for carrier networks.
Once an IP telephony system is in production use, the product eliminates the
manual processes used today to verify system performance and proactively
discover end-user VoIP issues before they occur. An innovative feature of
the software is that it automatically discovers every IP communications
device on your network.
They claim to be the first product to take advantage of CTI-level interfaces
and use them in a new way for gathering management and monitoring
information through active testing down to the IP telephone handset. Another
unique feature is that it can measure voice quality in a distributed
enterprise environment including calls that go off-net. In fact, ClarusIPC
Assurance was designed and architected to test and manage the features that
impact end-users from handset to handset and everything in between ï¿½ whether
ï¿½on netï¿½ or ï¿½off net.ï¿½
This product provides both a comprehensive full-service view and a granular
end-user level view. ClarusIPC Assurance displays end-user measurements into
meaningful and actionable information for use during system installation,
during system upgrades, fixes, or patches, and during daily production use.
Clarus was quoted as saying, ï¿½Because of its focus on the end-user level
metrics and its architecture necessary to enable those capabilities, as
opposed to others with appliance or client-server topologies, it will be
very difficult for others to develop comparable products without
re-designing their entire existing platforms to capture the full-duplex
nature of conversational technologies such as IP telephony.ï¿½
CONVERSip MP1000 Media Platform
Comdial has always had a reputation for creating innovative PBX solutions.
They were one of the pioneers for supporting CTI via the TSAPI and TAPI
standards. As a fairly sizable company, Comdial has typically targeted
medium to large businesses and has never shown any real interest in the SOHO
market or enterprises with less than 40 employees. So when Comdial decides
to target smaller enterprises, we take notice!
The MP1000 is a SIP media platform that is easy to deploy (important in SOHO
environments) and enhances productivity while lowering overall
communications costs. It features LAN telephony, an eight-port gateway, auto
attendant, voice mail, unified messaging, browser-based admin, and software
upgradeability. Comdial states, ï¿½The MP1000 meets the demand for a
cost-effective, plug-and-play communications solution for four to 40 seat
The MP1000 is innovative in that it delivers a complete media platform
solution in a single modular design. The MP1000ï¿½s SIP foundation, combined
with the EP200 Multimedia Endpoint (Windows XP softphone), allows users to
communicate in unique ways within a single application, using voice, video,
instant messaging, and presence management.
Additionally, the MP1000 comes with auto-attendant, voice mail, unified
messaging, eight-port IP-PSTN gateway, and Web-based administration
built-in. With all these capabilities bundled in the product, businesses
benefit from a highly integrated solution without having to add additional
cards or devices. With all of these combined features all in a virtual ï¿½plug
and playï¿½ package, we commend Comdial on a product that is sure to be a
winner in the SOHO marketplace.
you want to see ï¿½universal convergenceï¿½ of voice, data, fax, and other
communications methods, then you need look no further than the aptly named
CosmoCall Universe, which is a complete, unified contact center suite, that
includes ACD, IVR, CTI, predictive dialing, and multimedia recording.
Further, CosmoCall Universe supports multi-channel contacts including
telephone, Web chat, Web voice, Web video, Web collaboration, e-mail, and
voice mail all in one scalable, multi-tenant platform.
Since CosmoCall Universe is a hosted-model targeting service providers and
carriers, it is network-based by design, requiring agents and supervisors to
have nothing more than a PC, browser, a headset, and a connection to an IP
network. (PSTN support for agents without an IP connection is also
available.) Using its highly redundant, IP-based infrastructure, call center
agents, located anywhere, can interact with voice or message callers,
regardless of whether those callers originated on the PSTN or an IP network.
CosmoCall Universe enables calls and messages originating on telephone and
IP networks to be queued and routed to CSRs anywhere in the world. These
CSRs have complete feature functionality, including the ability to receive
regular telephone calls and supervisor capabilities.
It supports a plethora of features including SIP and H.323 support, CTI, Web
co-browsing, predictive dialing, pre-routing, IVR, Web chat, CRM, helpdesk,
recording, workforce management, and more. As far as we know, CosmoCom was
one of first companies to offer a fully unified multimedia contact center
solution that delivers a ï¿½unified multimedia queueï¿½ that includes VoIP,
chat, and Web co-browsing, and is based upon a hosted model.
CrystalVoice Communications, Inc.
CrystalVoice Software Version 4.0
There are many VoIP softphone clients on the marketï¿½ but not all are created
equal. Some offer better latency, jitter, and other voice quality metrics.
CrystalVoice software provides a high-quality voice communications
capability over the Internet, regardless of the Internet access method
(dialup, broadband, wireless, satellite, etc.). CrystalVoice software is
able to deliver a high-quality communication experience with as little as 8
kbps of sustained network bandwidth. It is further able to make the
necessary acoustic adjustments during the call that would otherwise make the
call unintelligible. Auto-adaptive voice compression (ranging from 8 kbps to
22 kbps), dynamic jitter buffer technology, packet loss concealment
techniques, and sophisticated transparent network access methods are all
part of the collection of techniques they have trademarked as ï¿½Acoustic QoS.ï¿½
By running CrystalVoice software at both ends of an Internet connection, the
userï¿½s experience is similar to what would be expected in a managed QoS
Importantly, the product has been designed and implemented to ensure that
calls can be connected, regardless of the firewall implementation, which so
often plague VoIP implementations. Another unique feature of CrystalVoice
software is that it has been integrated to work with the popular Cisco
CallManager. In addition to making outbound calls through CallManager via
the CrystalVoice softphone application, it also supports inbound
Web-initiated ï¿½click to talkï¿½ functionality. The Web visitor simply views
the Web site and selects the Click-to-Talk button to initiate the call. The
CrystalVoice software is then downloaded to the callerï¿½s PC, and the
software verifies that the callerï¿½s audio device is functioning properly.
The call is then placed, entirely over the Internet, and connects the caller
to their desired party.
In addition to redundancy and Unicode support, the product has sophisticated
audio device detection and configuration capabilities. The administrative
capabilities of the product include an advanced Systems Manager that
provides a Web-based interface for system management, administration, and
Having a VoIP softphone client on your PC to receive phone calls is surely
innovative, however having a softphone client installed on a PDA is even
more so. Imagine being on-the-go and instead of pulling out your cell phone,
you reach for your Palm or PocketPC to make and receive VoIP calls, which
travel across a wireless connection (WiFi, 3G, etc.) to an ITSP for
terminating the call. Already, there are PocketPC/cell-phone and
Palm/cell-phone hybrids, thus you may soon you get to choose whether to
terminate the call across the cell phone network or use a softphone client
on the PDA.
DiamondWareï¿½s Wi-Fone is a PDA software application that leverages 802.11
and an ITSP to make/receive calls to/from any phone number in the world.
DiamondWare claims to be the first on the market to partner with a major
ITSP (Net2Phone) to make a PDA into a PSTN edge device. DiamondWare has also
done extensive research to make their softphone application have one of the
lowest latencies and one of the best voice qualities around. Wi-Fone also
has no difficulty traversing firewalls with NAT.
Using the application is easy ï¿½ you simply download it for free, and then
enter in a credit card for the ITSP (currently only Net2Phone right now) to
use for billing and then you can make/receive phone calls from your PDA.
Currently, the product is only supported on the PocketPC environment;
however DiamondWare told us they are working on a Palm version.
Emergent Network Solutions
ENTICE Suite is comprised of several modules, including ENTICE Softswitch,
ENTICE Session Controller, ENTICE Elicit, ENTICE Tandem/International
Gateway, ENTICE VoIP Gateway, ENTICE Enhanced Services Platform, and a
couple other modules that can be deployed on general purpose Solaris or
The ENTICE Softswitch is expandable to over 100,000 ports of capacity ï¿½
quite impressive. ENTICE also supports H.248 and will be supporting MGCP in
the next release. The ENTICE Softswitch and Session Controller solutions can
be combined to provide a great solution for carriers who want to run a VoIP
network running a combination of wholesale traffic (passing traffic between
IP gateways) and retail traffic (passing traffic through firewalls to and
from residential and enterprise networks). Some softswitch solutions ï¿½lock
you inï¿½ to using strictly their own proprietary hardware. Whatï¿½s nice about
ENTICE is that their solution is vendor-agnostic. We should also mention
that the ENTICE Session Controller can also be used to perform protocol
conversion between H.323 and SIP.
All ENTICE solutions allow customers to program and customize the product
using the application layer APIs. Combined with source code availability,
this allows customers to deploy new services and solutions, which are
integrated into their front-end and back-end systems to achieve high levels
ENTICE claims to be the first to combine a softswitch and a session
controller into a single solution, which has now become the norm, not the
exception. They also claim to be the first softswitch to support the Excel
switch as a standard gateway product (including provisioning).
With a combined softswitch/session controller ENTICE enables carriers an
operationally efficient way of carrying wholesale traffic and retail
traffic. Wholesale carriers need the features of the session controller to
hide the underlying topology of IP-to-PSTN traffic and they need the NAT
traversal features to handle retail (enterprise or residential) traffic. The
ENTICE session controller uses their Virtual Media Server to manipulate the
media sessions being passed through the system. The VMS is being expanded to
include conferencing and other server based media services such as voice
prompts, digit collection and voicemail. With these capabilities, ENTICE
allows carriers to efficiently offer new features to their retail
subscribers such as prepaid, voice mail, and unified messaging.
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