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Product Reviews
July 2004

TMC Labs Internet Telephony
Innovation Awards 2004: Part I


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 fact, we have two very controversial selections involving Digium/Asterisk and Pingtel that resulted in some heated exchanges between all of our judges. We�ll discuss that later in this article.

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 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.


VCX V7200

When 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. We�ve never thought of 3Com�s IP telephony product line as targeting larger enterprises. In fact, one of the knocks of 3Com�s NBX-100 and 3Com�s VoIP product line in general was that it didn�t 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 �the first SIP-based, carrier-proven softswitch product scaled down to serve the needs of the large enterprise customer.�
We should mention that several mechanisms exist to connect multiple IP-PBXs together, but it�s more of a kludge, and they don�t 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, 3Com�s 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.


MetaLIGHT 1300

You like bandwidth, right? We know we do � we can�t get enough of it! That�s why when we heard about Actelis�s 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, Actelis�s 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 �plug and play� 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 � all over copper, which is not only more ubiquitous than fiber, but it�s 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 industry�s 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 Actelis� patented MetaLOOP technology.


AnchorPoint 5.0

If 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 � 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.

AnchorPoint�s 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.

AnchorPoint�s view is that VoIP is bigger than simply a �killer app.� 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 10�20 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 won�t creep back in.


Touchstone Telephony Port

The 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 �Outdoor Network Interface Unit� 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 service.

While the technically-savvy minded folks would scoff and say, �I can install an internal ATA in two minutes,� 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

This 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 �innovative� � whether it�s Digium or Pingtel � two similar products.

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: http://www.tmcnet.com/comsol/1101/1101cc.htm.
In Asterisk�s short existence, they have quickly harnessed a loyal developer community that has contributed to the source code. Digium has taken Asterisk�s 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, isn�t being the �first� the very definition of being �innovative�? 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. Asterisk�s 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 doesn�t have full SIP support yet, and they pointed out that Digium utilizes their own proprietary hardware � which is where they make their money. They also noted that Asterisk, which Digium is based upon, has no core developers � it�s all volunteers, whereas Pingtel not only intends to leverage the �volunteer developer community� 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 Pingtel�s SIPxchange is not tied to any proprietary hardware at all � 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 �innovative� 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 � especially considering someone can download Pingtel�s 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 �Yes, but who was the first to successfully build and sell an open-source-based IP-PBX? Who was the innovator?� Our Pingtel proponents couldn�t dispute this claim. So TMC Labs was stuck in a quandary � both Digium and Pingtel were both �innovative� 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 �innovation� 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 �best of breed� 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.


Its 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 � often a critical consideration in carrier deployments.


Citel Technologies
CITELlink IP & SIP Handset Gateway

As 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? Wouldn�t 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, wouldn�t 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 Gateway.

Citel Technologies� 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 world�s 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 � 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.
ClarusIPC Assurance

There are various tools out there to test VoIP, however, Clarus Systems� Clarus IPC Assurance tests VoIP from a different and unique angle. Clarus Systems� ClarusIPC Assurance simplifies and automates VoIP management tasks through its unique �end-user level� 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 � whether �on net� or �off net.�

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, �Because 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.�


Comdial Corporation
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, �The MP1000 meets the demand for a cost-effective, plug-and-play communications solution for four to 40 seat offices.�

The MP1000 is innovative in that it delivers a complete media platform solution in a single modular design. The MP1000�s 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 �plug and play� package, we commend Comdial on a product that is sure to be a winner in the SOHO marketplace.


CosmoCall Universe

If you want to see �universal convergence� 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 �unified multimedia queue� 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 market� 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 �Acoustic QoS.� By running CrystalVoice software at both ends of an Internet connection, the user�s experience is similar to what would be expected in a managed QoS network.

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 �click to talk� 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 caller�s PC, and the software verifies that the caller�s 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 troubleshooting.


DiamondWare, Ltd.

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.

DiamondWare�s 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 � 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

The 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 Linux servers.

The ENTICE Softswitch is expandable to over 100,000 ports of capacity � 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 �lock you in� to using strictly their own proprietary hardware. What�s 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 of efficiency.
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.

[ Return To The July 2004 Table Of Contents ]

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