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Product Reviews
August 2004

TMC Labs Internet Telephony Innovation Awards 2004: Part II


With our fifth installment, the popularity of the TMC Labs Innovation Awards seems to grow by leaps and bounds each year. This year we�ve 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 �innovative products/services� to accommodate this increase in applicants? Well, TMC Labs doesn�t 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, we�ve often been asked, �Why doesn�t TMC Labs have a �TMC Labs 100� award given to the top 100 players in the VoIP market?� Well, our answer is we don�t 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 don�t need to rehash that information.


The purpose of the TMC Labs Innovation Awards is just as the name states � it�s about awarding �innovation.� We grant Innovation awards based solely upon how unique or innovative a particular product or service is. We don�t 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 �bandwidth� genre (i.e., improves bandwidth, maintains QoS, or performs �traffic shaping�). 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. 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)

Company Name

Product Name


VCX V7200


MetaLIGHT 1300


AnchorPoint 5.0


Touchstone Telephony Port


IPM-260 8 E1/T1 PCI VoIP Board

Citel Technologies

CITELlink IP & SIP Handset Gateway

Clarus Systems, Inc.

ClarusIPC Assurance


CONVERSip MP1000 Media Platform


CosmoCall Universe

CrystalVoice Communications, Inc.

CrystalVoice Software Version 4.0

DiamondWare, Ltd.



Asterisk Open Source IP-PBX Platform

Emergent Network Solutions

ENTICE Product Suite


Hammer FX-IP, Hammer Call Analyzer

Hughes Software Systems

MicroSIP Toolkit


Assured Quality Routing (AQR)

Interactive Intelligence Inc.

Enterprise Interaction Center (EIC)

IP Unity

Harmony6000 Media Server

Jasomi Networks

PeerPoint Centrex

Level 3 Communications

(3)VoIP Local Inbound Service


MASERGY�s Service Control Center (SCC)



Red Hawk/CDT


Siemens Information and Communication Networks, Inc

HiPath 8000 Real-Time IP System

Toshiba America Information Systems, Inc., Digital Solutions Division

Toshiba SoftIPT SoftPhone

Ubiquity Software

SIP Application Server


FlexLight Networks




Shunra Software


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 provided.


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 VoIP protocols.


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 Empirix�s 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 Empirix�s 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.


Empirix�s other innovative product � the Hammer Call Analyzer � 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 diagram.


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 �understood� 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 signaling analysis.
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 media.

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, they�ve added a unique �jitter over time� 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
MicroSIP Toolkit


As SIP becomes the predominant VoIP protocol of choice, there are many vendors trying to �cash in� 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 �marry� 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 rates.

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 award�s namesake!


Interactive Intelligence Inc.
Enterprise Interaction Center (EIC)


TMC Labs is quite familiar with Interactive Intelligence�s 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 EIC�s SIP support, they�ve 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 � 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 � even equipment for hybrid TDM and IP deployments � 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 Intel�s Netstructure Host Media Processing (HMP) software for an all-software IP telephony option. Using Intel�s 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.

EIC�s 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 in-house.

EIC certainly didn�t 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.


Jasomi Networks
PeerPoint Centrex


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 firewall�s security policies have been set up, defined, and indeed �hardened� against attack over the course of years.

With Jasomi Networks� PeerPoint Centrex Edition (PPCE), which sits on a service provider�s 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 user�s ISP may use NAT, the user�s 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 provider�s equipment are directed to PPCE, where a reverse set of translations is applied in order to prepare the packet for transit back through the customer�s NAT-enabled firewalls. PPCE ensures that when the packet arrives back at the user�s IP phone, its contents are once again standards-compliant from the phone�s 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 client�s 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 customer�s premises. All of the NAT translation is done by PPCE on the service provider network. Think of PPCE as a �hosted firewall solution� that works with customer�s existing firewall or simply as �plug and play VoIP� 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. population.

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 customer�s 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 VoIP.


MASERGY�s 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. Masergy�s 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 ordered.

What we liked most is that Masergy�s 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 don�t need.


Red Hawk/CDT


Internet Telephony� Magazine has espoused the benefits of �one wire to the desktop� for years. In order to be �truly� 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 �A/C brick�) 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 � 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 current surge.

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 isn�t 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 �sign-posts and readers� 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 �enterprise features� with carrier-class reliability and scalability for large organizations. The HiPath 8000 is the one of the industry�s 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 � voice calls, e-mail, and voice mails � 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 �nines� of availability making it truly a carrier-class hosted IP-PBX solution.


Toshiba America Information Systems, Inc.
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 �general purpose� they are not specialized to emulate a specific phone. That is, these softphones� 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 �softphone� clients that emulate the look, feel, and functionality of a traditional desktop phone, so it can become a �true� 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 communication costs.

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 user�s 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.


Ubiquity Software
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 next-generation networks.

Ubiquity�s 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 user�s 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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