TMC Labs Internet Telephony Innovation
Awards 2004: Part II
BY THE STAFF OF TMC LABS
With 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 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
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. This month, we start with Empirix and work
our way through to Ubiquity Software. We hope you find these products as
innovative as we did.
2004 TMC Labs Innovation Award Winners (Full List)
Touchstone Telephony Port
IPM-260 8 E1/T1 PCI VoIP Board
CITELlink IP & SIP Handset Gateway
Clarus Systems, Inc.
CONVERSip MP1000 Media Platform
CrystalVoice Communications, Inc.
CrystalVoice Software Version 4.0
Asterisk Open Source IP-PBX Platform
Emergent Network Solutions
ENTICE Product Suite
Hammer FX-IP, Hammer Call Analyzer
Hughes Software Systems
Assured Quality Routing (AQR)
Interactive Intelligence Inc.
Enterprise Interaction Center (EIC)
Harmony6000 Media Server
Level 3 Communications
(3)VoIP Local Inbound Service
MASERGYï¿½s Service Control Center (SCC)
Siemens Information and Communication Networks, Inc
HiPath 8000 Real-Time IP System
Toshiba America Information Systems, Inc., Digital
Toshiba SoftIPT SoftPhone
SIP Application Server
companies appearing in italics appeared in our July
2004 issue with a full description.
Hammer FX-IP & Call Analyzer
We had difficulty choosing between the Hammer FX-IP and the Hammer Call
Analyzer since they both had unique attributes, and as VoIP testing tools
they actually complement each other quite well. For example, you can use
Hammer FX-IP to simulate VoIP calls to a VoIP device and then use Hammer
Call Analyzer to analyze the call on the network. Thus, we decided both were
worthy of this award.
In a nutshell, the Hammer FX-IP is an IP functional test platform with
scripting capabilities for verifying VoIP applications. Hammer FX-IP can be
used to generate both the signaling and media required to emulate SIP, MGCP,
and H.323 endpoints as well as measure voice quality using PESQ or PAMS.
Extensive codec support as well as in-band and out-of-band DTMF is also
Empirix based the Hammer FX-IP tester on an internally developed,
programmable state machine, which goes well beyond all of the traditional
limitations of stack-based implementations. Stacks are programmed to behave
according to the particular RFC they represent. This makes negative testing
very difficult to accomplish since the stack cannot be forced to send
malformed or inappropriate messages during a call flow. The programmable
state machine architecture at the core of the Hammer FX-IP addresses this
limitation. This enables users to create negative test scenarios and custom
call sequences, which are critical in testing device interoperability with
Empirix claims that Hammer FX-IP is the first product to provide a
scripting language for IP-based call flows. It utilizes Hammer Visual Basic,
a scripting language first introduced on Empirixï¿½s product family of TDM
test tools that has been adapted for IP call generation.
Since the Hammer FX-IP incorporates the Hammer Testbuilder and Hammer
Visual Basic interfaces that are also used in Empirixï¿½s TDM test tools,
existing customers find that the learning curve for this product is
extremely short. Importantly, it is the ability to control both the IP and
TDM sides of a call from a single interface, which is invaluable for testing
next generation devices and applications.
Empirixï¿½s other innovative product ï¿½ the Hammer Call Analyzer ï¿½ is
similar to generic protocol analyzers, but much more specialized to analyze
the VoIP traffic on your network. Unlike generic protocol analyzers, the
Hammer Call Analyzer enables users to visualize signaling and voice quality
problems in VoIP networks by displaying multistage call flow via a ladder
Empirix claims that Call Analyzer was the first analyzer designed
specifically for VoIP applications. The initial release in February 2003
focused on signaling analysis and was the first tool to offer a ladder
diagram display and to automatically associate messages from a single call
within a protocol domain. It was also the first such tool to offer
protocol-aware filtering, search, and triggering, meaning that it
ï¿½understoodï¿½ and could take action on protocol level events. The 1.1 release
was the first to incorporate TDM and IP signaling analysis in a single
analyzer. Empirix claims that with the 1.2 release, the analyzer was the
first to incorporate detailed media analysis and playback with VoIP
In addition, the Hammer Call Analyzer provides media quality metrics and
displays waveforms and its unique Stream Quality Signature for any call.
These features allow engineers to visualize problems in the exchange of
messages and transmission of media between the various devices and to
quickly solve them.
Also, Hammer Call Analyzer was first to incorporate R-factor (the result
of Emodel) in an analyzer (1.2 release). In that release, the analyzer also
incorporated the concept of a jitter buffer for analyzing and listening to
The analyzer can automatically link call legs across signaling domains. The
result is that users can double-click on any message (or media packet)
within a call and bring up a ladder diagram with every message across
multiple protocols and including all the media stream packets for the call.
In the 1.4 release the analyzer became the first to display RFC 2833 encoded
digits on a ladder diagram.
In the their latest release of the product, theyï¿½ve added a unique ï¿½jitter
over timeï¿½ display and color coded the waveform display to help engineers
zero in on stream problems such as lost packets, out of sequence frames and
duplicated packets. A really cool feature is that the display enables users
to zoom in on parts of a waveform and automatically associate selected parts
of the waveform with the associated packet decodes.
Hughes Software Systems
As SIP becomes the predominant VoIP protocol of choice, there are many
vendors trying to ï¿½cash inï¿½ by selling SIP toolkits and chipsets. SIP is a
powerful protocol that can enable handset manufacturers to provide
value-added applications such as push-to-talk, instant messaging and
presence-based applications. For instance, 3G handset manufacturers are
looking to add SIP support to easily offer such applications. Hughes
Software Systems MicroSIP Toolkit is an innovative toolkit that helps reduce
development time and is specifically positioned for the handset market.
One of the key requirements for handset manufacturers is to minimize the
memory usage. HSS MicroSIP Toolkit has a hand-written parser, which ensures
minimum memory usage. In fact, HSS claims that their solution has the lowest
footprint in the market. The toolkit exposes call/transaction level APIs as
against the trigger/message level APIs provided by the stack to the
developers. The toolkit maintains call-states, forms and validates messages
based on the call states thus freeing the application writers to add
innovative features and services rapidly. It supports SIP messages such as
INVITE, OK, BYE, CANCEL, ACK, PROPOSE, INFO, REGISTER, OPTIONS, all response
codes and any other SIP Method that is compliant to SIP ABNF.
The solution supports the following key SIP headers: From, To, Call-Id,
CSEQ, Via, Contact, Route, Record-Route, Require, Unsupported, Warning,
Authorization, Timestamp, Content-Type, Content-Length, Content-Encoding. It
also supports SDP fields such as version (v=), session (s=), origin (o=),
connection (c=), time (t=), media (m=), attribute (a=). The solution
supports Multi-part MIME message body parsing and formation.
HSS MicroSIP toolkit supports a subset of SIP headers and SDP fields in
SDP messages. All other headers/SDP lines are accepted, but parsed simply as
a name and body pair. This ensures a low memory usage and at the same time
not losing on any of the key features provided by SIP protocol.
Assured Quality Routing (AQR)/PathEngine
We all know the cost savings
associated with VoIP, especially when least cost routing (LCR) is
implemented to select the cheapest route to get to a particular destination.
We also know that QoS is a critical aspect when it comes to VoIP. Well, what
if you were to ï¿½marryï¿½ LCR and QoS technologies onto a single VoIP
application or platform? iBasis has done just that with Assured Quality
Routing with PathEngine, which integrates global least cost routing (LCR)
with IP performance metrics to enable dynamic international call route
selection over the Internet based on real-time network quality data as well
as cost data.
AQR and PathEngine manage approximately 12,000 routes between iBasis
Internet Central Offices (ICO) and terminating partners in more than 95
countries. Route choices include multiple ISPs from each ICO, as well as
tandeming from one ICO to another to access a higher quality IP route to the
terminating destination. Performance is measured in terms of Quality
Reliability, which is the percent of time within the prior 60 minutes that
the latency and packet loss on a specific source/destination path has been
within iBasis quality standards; and Minimum Uptime, which refers to how
long a path has remained within the QR standards. AQR also provides
sophisticated advanced routing capabilities such as time-of-day and
day-of-week routing and percentage-based routing, which enable iBasis to
maintain highly cost-efficient routing as well as higher call completion
iBasis claims that to the best of their knowledge, their patent-pending
technology represents the first time any VoIP carrier has been able to
dynamically alter routing based on real-time IP performance data integrated
with LCR. This is certainly innovative and without a doubt worthy of this
Interactive Intelligence Inc.
Enterprise Interaction Center (EIC)
TMC Labs is quite familiar with
Interactive Intelligenceï¿½s EIC product. In fact, we were one of the first to
review the product back in 1999 at their Florida testing facility (http://www.tmcnet.com/17.1).
Well in four years, the EIC product has come a long way. We raved about the
modularity, openness, and extreme flexibility of EIC back in 1999, but now
with EICï¿½s SIP support, theyï¿½ve taken their IP PBX communications system to
a whole other level!
Originally released in 1997, EIC was one of the first open, Windows-based
PC-based PBXs in the industry ï¿½ although it was much more than just a PBX.
This innovation was made even more unique because, unlike its competitors
offering multi-vendor, multi-platform solutions, EIC was built from the
ground up as a single-vendor solution based on single-platform architecture.
EIC also qualifies as a truly innovative product because it was built as a
separate applications layer so that organizations could select the
networking infrastructure and peripheral devices from the vendors of their
choice ï¿½ even equipment for hybrid TDM and IP deployments ï¿½ thus avoiding
the kind of vendor lock-in required by many of its competitors.
In 2002, EIC became the first product of its kind to include SIP support
with built-in media processing on the same server architecture. EIC claims
that in 2003 they were the first IP PBX to incorporate Intelï¿½s Netstructure
Host Media Processing (HMP) software for an all-software IP telephony
option. Using Intelï¿½s HMP software eliminates the need for voice processing
boards, thus enabling organizations to cut costs by up to 40 percent
compared to board-based IP deployments.
EICï¿½s unified architecture requires fewer integration points compared to
competitive products. In fact, EIC claims that on average their solution
requires one-third fewer devices than traditional PBX products. This
translates into reduced costs, simplified maintenance and administration,
and faster customization. Also, EIC comes with a built-in graphical
application generator making it easy to create and customize applications
EIC certainly didnï¿½t hold back on including sophisticated applications and
features, which includes TDM and SIP-based IP switching, auto-attendant,
voice mail, desktop soft phone, unified messaging, find-me/follow-me,
conferencing, workgroup routing, basic screen-pop, supervisory monitoring,
Web chat and callback, fax services, reporting, remote support, multi-site
presence management, and hot desking.
Two of the greatest impediments
to mainstream SIP deployment today are security at the edge of the network
and NAT firewall traversal issues. In order to facilitate VoIP,
organizations had to contemplate throwing out their existing firewalls. This
is certainly not a practical solution considering not just the cost, but the
fact that the firewallï¿½s security policies have been set up, defined, and
indeed ï¿½hardenedï¿½ against attack over the course of years.
With Jasomi Networksï¿½ PeerPoint Centrex Edition (PPCE), which sits on a
service providerï¿½s network, you can not only keep your investment in your
existing firewall, but you can also solve the SIP over NAT traversal issues.
In fact, in some deployments, PPCE must also cope with multiple NATs in the
path between the service provider and IP phone. For example, the userï¿½s ISP
may use NAT, the userï¿½s firewall may use another NAT, and the user may then
have more internal NATs embedded in devices such as Internet line sharing
devices or wireless networking routers. PeerPoint Centrex Edition is able to
cope with any number of nested NATs of varying kinds, including both
symmetric and conical NATs.
Packets from the service
providerï¿½s equipment are directed to PPCE, where a reverse set of
translations is applied in order to prepare the packet for transit back
through the customerï¿½s NAT-enabled firewalls. PPCE ensures that when the
packet arrives back at the userï¿½s IP phone, its contents are once again
standards-compliant from the phoneï¿½s perspective.
In addition to far-end NAT traversal for VoIP service providers, PPCE also
provides intrusion prevention, protocol repair, and media path optimization
(MPO)MPO is an incredibly innovative feature, and Jasomi claims it can
reduce a clientï¿½s bandwidth costs by up to 90 percent and pay for the
product itself in less than a month. The end result of all this
functionality is that a service provider can provide voice (over IP) service
to customers behind NATs, without requiring any new equipment or software
deployments on the customerï¿½s premises. All of the NAT translation is done
by PPCE on the service provider network. Think of PPCE as a ï¿½hosted firewall
solutionï¿½ that works with customerï¿½s existing firewall or simply as ï¿½plug
and play VoIPï¿½ that just works!
Level 3 Communications
(3)VoIP Local Inbound service
(3)VoIP Local Inbound service is
quite unique in that it terminates PSTN-originated calls (dialed to a local
phone number) to IP endpoints anywhere in the world. Normally calls
originate on IP and terminate on the PSTN. This product does just the
reverse! The PSTN-to-IP service is available in markets across the United
States. Essentially it is a replacement for one-way PRI service from Local
Exchange Carriers, and an alternative to toll-free services from
Inter-Exchange Companies. Level 3 claims to be the first company to offer
local-number calls with a VoIP termination to IP addresses anywhere in the
world with local-number coverage blanketing over 90 percent of the U.S.
Call centers, customer care providers, conferencing companies, and enhanced
voice-services providers can all draw value from using this product. We
should point out that Level 3 is one of just a few ITSPs leading the way by
implementing SIP technology. In addition, the customer (not the carrier) has
the control of the call flow and how each call is treated. Flexibility and
control are in our customerï¿½s hands, giving them the power to develop their
own unique features and products to execute their business plan. The ability
to fully integrate with SS7-based network providers affords customers the
ability to smoothly transition from their legacy voice services to Level 3
MASERGYï¿½s Service Control Center (SCC)
With more companies looking to
cut costs by converging their voice, video, and data applications onto a
single data pipe, the importance of guaranteeing bandwidth, increasing
bandwidth on-the-fly, and guaranteed QoS is of utmost importance. Masergyï¿½s
Service Control Center (SCC) platform, running on an IP-based MPLS network,
does all of this and more. SCC can prioritize packets and give consistent
performance for business applications. Their SLA is very impressive and
includes 100 percent packet delivery and 100 percent packets in sequence,
jitter that will never exceed 10ms, and network recovery of less than 1
second. Masergy claims by using SCC that you can lower your overall cost for
voice, video, and data by one-third or more.
SCC is a Web-based management tool, which empowers you to provision services
instantaneously, change bandwidth, prioritize applications, view network
usage, monitor the performance of individual applications across the entire
network, or view application performance by defined groups. In addition, you
can receive alerts based on application metrics to prevent performance
degradation, drill down for individual-level details on device performance,
and generate standard or custom reports on the performance of applications
for distribution to internal clients. Service changes are integrated with
contact/billing terms for confirmed price implications before services are
What we liked most is that Masergyï¿½s Service Control Center saves companies
time and money by allowing them to provision services and increase bandwidth
in real-time with the simple click of a mouse. Importantly, we should
mention that most other solutions require one to eight weeks to add
additional bandwidth, such as a second T1 line. With SCC, customers can pay
for the bandwidth they need and not have to over-provision and over-pay for
bandwidth they donï¿½t need.
Internet Telephonyï¿½ Magazine has
espoused the benefits of ï¿½one wire to the desktopï¿½ for years. In order to be
ï¿½trulyï¿½ one-wire to the desktop, a method of providing power over Ethernet
with centralized backup power support must be devised. The alternative is
that each desktop has to have an A/C adaptor (aka ï¿½A/C brickï¿½) under the
desk along with a UPS to provide backup power to the IP phone in the event
of power failure. The cost economics of having a UPS underneath each desk is
unfeasible. That is why Power over Ethernet is an innovative technology that
is finally starting to gain traction in the industry.
One company, Red Hawk/CDT, has technology that allows IP telephones,
wireless LAN access points, security cameras and other enterprise terminals
to safely receive power over standard Category 5, 5e, or greater LAN cabling
without modification to existing infrastructure.
PowerSense is a Mid-Span ï¿½ a power patch panel-like device, residing between
an ordinary Ethernet Switch and the device to be powered. Power over
Ethernet Mid-Spans add power on the spare wire pairs of a Cat 5, 5e or 6 LAN
Data cable and do not disrupt data traffic. It supports the IEEE 802.3af
Power over Ethernet standard and it provides an impressive 15.4W per port.
Their high-end solution features a modular 24 port rack-mount chassis, is
scalable, and features hot-swappable modules that can be changed without
powering down in the event of failure. Importantly, each module has its own
DC to DC converter, which is completely voltage isolated from the other
ports. Each module is separately fused and protected from any unexpected
Also, Red Hawk/CDT claims to be the first to market with PowerSense
supporting Cisco Detect Protocol and providing power over Ethernet to Cisco
endpoints. Power over Ethernet isnï¿½t just for IP telephones either. One
innovative application of the 24V power being supplied over the Ethernet
cable is that it works with Savi Technologies products to power ï¿½sign-posts
and readersï¿½ that track RFID tags for asset management.
Siemens Information and
Communication Networks, Inc.
HiPath 8000 Real-Time IP System
There are now a plethora of
IP-PBXs on the market today to choose from, which is good news, since what
constitutes a good IP-PBX solution for one customer may not be right for
another. One of the key differentiating factors is scalability; some may
require just 50 ports while others may require thousands of ports. As such,
there are IP-PBXs that address each market segment. Certainly, the HiPath
8000 Real-Time IP System has scalability covered with its impressive 100,000
users per node and unlimited users per network, depending on configuration.
It was designed from the ground up for very large, dispersed enterprises.
Target markets include mega enterprises (Fortune 1,000, government, military
and universities, for example), and service providers.
The HiPath 8000 is based on a hosted, Web-services communications system
that can be deployed and managed from an IT data center enabling ï¿½enterprise
featuresï¿½ with carrier-class reliability and scalability for large
organizations. The HiPath 8000 is the one of the industryï¿½s first
carrier-grade Communication over IP systems that hosts communication
services on a Web-Services-based architecture. Using an IP-based, SIP
network overlay solution, enterprises can cost-effectively integrate
converged voice, multimedia, and multi-modal services and applications
across the organization. Also, subscriber self-care enables users to choose
the communication features they need through a company Web portal.
The HiPath 8000 is standards-based supporting SIP, SALT, XML, SOAP, and
SIMPLE, which enables you to easily build workflow applications as well as
integrate with a variety of third-party SIP clients and devices. In
addition, ComAssistant 8000 is a powerful Web-based application that
delivers desktop call control as well as filtering and routing of incoming
communication ï¿½ voice calls, e-mail, and voice mails ï¿½ based on presence and
rules. Additionally, applications from IP Unity (also a TMC Labs Innovation
Award winner) enable the scaling, customization, and bundling of
capabilities ranging from simple voice mail to unified messaging, auto
attendant, and multimedia Web conferencing. Siemens claims not five, but six
ï¿½ninesï¿½ of availability making it truly a carrier-class hosted IP-PBX
Toshiba America Information
Digital Solutions Division
Toshiba SoftIPT SoftPhone
Sure there are quite a few
general-purpose softphones on the market today designed to work with any
IP-enabled PBX, however, because they are ï¿½general purposeï¿½ they are not
specialized to emulate a specific phone. That is, these softphonesï¿½
interfaces do not look or act like the hard phones that they are replacing.
This results in an extra learning curve for users.
Well, the phone manufacturers have finally caught onto this fact and are
beginning to offer ï¿½softphoneï¿½ clients that emulate the look, feel, and
functionality of a traditional desktop phone, so it can become a ï¿½trueï¿½
desktop replacement. One perfect example is the Toshiba SoftIPT SoftPhone,
which extends the features and functionality of Toshiba desktop telephones
to laptops or PCs running Windows XP, giving users all of the features and
capabilities of their desktop telephones. The Toshiba SoftIPT SoftPhone can
be used anywhere users have access to wired or wireless connectivity,
providing them with increased productivity and mobility, as well as reduced
The SoftIPT is the first softphone designed for Toshiba Strata CTX business
telephone systems. With the SoftIPT, users can extend virtually all of the
features of their Toshiba desktop telephones to their Toshiba laptops or PCs
using a user friendly interface to manage both incoming and outgoing calls.
Unlike competitive IP telephones, the SoftIPT provides seamless feature
transparency with the userï¿½s desktop telephone. The new SoftIPT uses the
Megaco+ protocol, which allows the implementation of the telephone features
supported on Toshiba desktop phones on its IP-based phones. In fact, in
addition to the typical features you would expect it to support (CallerID,
speed dials, Call Forward, Transfer), it even allows you to record the call
and save it as a WAV file.
SIP Application Server
As most of us already know, SIP
is a much easier protocol to use and develop VoIP applications with than its
H.323 counterpart. Using a SIP application server allows ISVs, developers,
carriers, and technology providers to quickly and easily build and deploy
SIP-based applications, such as Push-To-Talk, Unified Messaging,
Conferencing, Interactive Gaming, and more for wireless and wireline
Ubiquityï¿½s SIP Application Server was one of the first SIP-based
applications servers on the market, powering a flexible service creation
environment for technology providers, developers, and carriers. The software
serves as an underlying platform to power global end-to-end
push-to-talk-over-cellular launches and trials. In fact, Ubiquity is in
launches and trials with more than 15 service providers around the world
using Siemens Mobile and they claim to be the first to be selected for
integration into Siemens IMS.
Ubiquity Software has designed its Application Server to have high standards
in redundancy and availability as well as the ability to process thousands
of calls in real time. For redundancy, it uses a clustered extensible
architecture leveraging embedded load balancing and SIP High Availability.
SIP High Availability is different from Hardware High Availability since
calls in SIP have state and concurrent session information that if lost will
interrupt the userï¿½s multimedia experience. Ubiquity Software insures that
this information is preserved in the event of failure and through its High
Availability Cluster technology, that no interruption to services is
experienced by the customer. This embedded SIP High Availability is a
critical feature since it leaves the developer to focus on application
creation rather than being concerned with deployment issues.
Ubiquity provides a comprehensive Software Development Kit (SDK) comprising
of Application Building Blocks that form the most commonly used basic
components of many applications such as Presence Control, Instant Messaging,
and Session Control. Ubiquity provides access via mechanisms such as SOAP
and RMI. Further, Ubiquity provides a comprehensive Software Development Kit
comprising of Application Building Blocks that form the most commonly used
basic components of many applications such as Presence Control, Instant
Messaging, and Session Control, thus reducing development time.
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