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Product Reviews
July 2002

TMC Labs� Third Annual Innovation Awards


This feature embodies the third volume of TMC Labs Innovation Awards awards for INTERNET TELEPHONY� magazine and, we feel, well represents a milestone of maturity for the industry. With a variety of entrenched products and companies in our field, we�ve really drilled down into the product space this year. In the past we�ve always come up with a comprehensive list of products that in many ways will shape voice�s technological year to come (approximately 30 products have already been chosen for an Innovation Award by TMC Labs); this year we feel is no exception.

In keeping with its namesake, the Innovation Awards are presented to products TMC Labs feels are exceptional and innovative; the products selected must additionally be on the market contributing to the evolution of packet voice. We found ourselves digging much deeper for this year�s awards, exploring many new, more specialized solutions employing up-to-the-minute technologies in this space: XML, 802.11, and Bluetooth to name only a few. SIP seemed to be the protocol of choice in most of our product entrants this year; we also noticed a dramatic increase in presence-based solutions, some of which gated both IP and wireless architecture.

Our most successful Innovation Award campaign to date, we received almost 200 applications for this year�s event. While it obviously meant more work for us, the task didn�t seem quite so daunting as we reveled in delight at the sheer volume of VoIP product submissions. Both sophomore and tenured companies submitted applications this year announcing new, recently released products. More intriguing still were the number of applicants providing solutions to major PSTN carriers. We hope you enjoy Internet Telephony�s third annual TMC Labs� Innovation Awards. With an overwhelming response this year, it has truly become a labor of love.

TMC Labs Innovation Awards Winners
Alcatel Eyretel
Genuity Jasomi Networks, Inc
Mediatrix Telecom, Inc. NetIQ Corporation
Symbol Technologies Sylantro
Telica Voyant Technologies
Empirix Webley Systems

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OmniPCX 4400

Choosing Alcatel�s OmniPCX 4400 as one of our Innovation Award winners was a no-brainer for several reasons. By running on UNIX, Alcatel certainly will avail itself of a ton of developer support from the vast UNIX developer community. Besides running on the UNIX operating system, the OmniPCX supports all the main relevant IP standards, including embedded DHCP, RTP, RTCP, UDP, TCP/IP, H.225, H.245, RAS, TFTP, and more. It also supports QoS standards such as 802.1 p/Q, ToS, and DiffServ.

Another innovative aspect of the OmniPCX is that unlike most of its competitors, it doesn�t require proprietary phones. The OmniPCX 4400 supports third-party H.323 phones, such as the PolyCom SoundPoint IP and even third-party SIP phones, such as Pingtel�s SIP phones.

Alcatel�s ABC protocol is a superset of QSIG that optimizes reliability and performance in an �all Alcatel� network. For heterogeneous configurations (not a 100 percent Alcatel network), the QSIG multi-vendor protocol standard can be used to connect to legacy PBXs. In single site configurations it scales up to 5,000 users, and by networking up to 100 Call Servers it can scale up to an impressive 50,000 users.

As if the OmniPCX didn�t have enough features rolled all in one, it also features a unique in-building wireless functionality that is integrated with the OmniPCX. The radio base station connects directly to the OmniPCX using digital cards instead of going through a gateway. In addition, the OmniPCX also works with Symbol�s wireless VoIP phones (another 2002 TMC Labs Innovation Award winner).

The OmniPCX has an extensive list of optional call center modules, including ACD/IVR functionality, outbound dialer for campaigns, agent login/logout, discrete call listening for supervisor, multimedia skills-based routing, and extensive reporting and statistics. Its multimedia capabilities include Web call back, Web call through (VoIP), text chat, e-mail, and collaborative browsing. TMC Labs loved the Alcatel OmniPCX 4400 right from the start and we wish we had one in the lab right now just so we could tinker with it.

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MediaStore IP
 So your company just bought one of those newfangled IP-PBXs. Certainly one of the purchasing factors was that IP-PBXs allows for a far simpler installation making moves, adds, and changes much simpler. Unfortunately, if your organization is a call center or any other company that implemented an IP-PBX and requires call recording � you were out of luck. There were no call recording products that could record VoIP conversations traveling across the IP network.

We should point out that the recording of traditional PBXs required the connection to proprietary telsets or telephone trunk lines along with proprietary CTI links. Simple software configuration of the business rules for recording are just some of the benefits of VoIP recording. The complexities and costs associated with CTI begin to disappear as voice and data converge onto a single network allowing managers to record, monitor, and analyze customer interactions, including the agents� screens. In theory the ability to record both voice and data over IP is great, but until recently the technology to record VoIP calls did not exist. Now, with Eyretel�s MediaStore IP product, a comprehensive, enterprise-wide VoIP and screen recording solution is available.

MediaStore IP integrates with the Cisco IP Contact Center (IPCC) supporting Cisco�s SCCP (�skinny� protocol) and other IP telephony solutions, enabling contact centers to use the same IP infrastructure to process both voice and data. Using MediaStore IP, contact centers will be able to record all the various customer interaction touch points. In addition to recording the voice, MediaStore IP will also record the agent�s screen to capture such important activities as Web chat, agent assisted Web browsing, desktop computer activity and e-mail. MediaStore IP provides the ability to monitor and analyze all of the recorded information via a single system. One unique feature of MediaStoreIP is its ability to record conversations in stereo, allowing for on-screen playback that visually displays each side of the conversation. Also, calls can be queried and retrieved and then replayed through a browser interface, allowing managers or agents with appropriate security rights to view calls from anywhere. Overall, TMC Labs was very impressed with Eyretel�s MediaStore IP product, which establishes new ground for IP telephony.

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Black Rocket

Last year Genuity launched a TV ad campaign that featured a model of a black rocket at some point in the commercial. One of them was a boardroom with a black rocket sitting on the conference table and a twenty-something geek getting the old board members to somehow perform the �wave� by raising their hands. If you were like us, you were probably dumbfounded what the heck the black rocket did. The commercial seemed to elicit some strange magical powers upon the �black rocket� � as though it were the solution to all your problems. Although not as annoying as the SuperBowl�s nebulous mLife commercials, it certainly did its job in branding the Black Rocket name. One year later when TMC Labs received an online application from Genuity applying for the TMC Labs Innovation Awards we instantly flashed back to their commercials. Thus, their brainwashing, err marketing messages sure did the trick!

We checked out Black Rocket on Genuity�s Web site and determined that Black Rocket is actually made up of several disparate modules often completely unrelated to one another. For example, Black Rocket Voice is VoIP-related and Black Rocket Storage is backup, restoration, and archiving of data, with the ability to restore entire systems. Yet another module is Black Rocket Hosting, which is Genuity�s solution for secure, enterprise hosting infrastructures. With Black Rocket having such a varied product line, our confusion about what exactly made up Black Rocket only seemed to worsen. However, after careful examination and research we figured out just what Black Rocket is all about. We could best sum up Black Rocket as a solution that provides voice/date convergence on a managed backbone network that offers enhanced services such as disaster recovery, VoIP with quality of service guarantees, managed Web hosting, managed security services.

We focused our attention on Genuity�s Black Rocket Voice module, which is a Voice over Internet Protocol (VoIP) communication solution for enterprises that integrates voice and data traffic onto a single, multi-protocol IP network infrastructure utilizing Genuity�s Tier 1 IP backbone. Black Rocket Voice is designed to supplement or replace existing enterprise voice solutions for both inbound and outbound calling including intra-company (�on-net�) voice and data communications, and also for inter-company (�off-net�) communications anywhere in the world. Genuity can work in conjunction with legacy PBXs to deliver enterprise-wide IP telephony service while protecting an enterprise�s investment in existing PBX and telephony handset equipment.

Genuity claims that their Black Rocket Voice solution is the first convergence solution to deliver end-to-end prioritization of voice traffic from the customer premise across a backbone network. To back up their claim of voice quality, Genuity is the only company that we are aware of that offers voice quality metrics for service level guarantees. In fact, Genuity provides a proactive service level guarantee for the Black Rocket Voice services and a reactive service level guarantee for service availability. Genuity guarantees a Service Availability of 99.97 percent, Network Latency of 55ms, Packet Loss of up to 0.25 percent, Jitter of 10ms, and Voice Quality � as measured by the Perceptual Analysis Measurement System (PAMS) scale of 4.0 (carrier quality) for the G.711 codec and 3.7 (business quality) for the G.729 codec.

Cisco and Verizon (a minority stakeholder) are two of the first customers for Genuity�s Black Rocket Voice. It�s part of the Black Rocket eBusiness Network Platform for enterprise Internet access, hosting and IP transport over Genuity�s Tier 1 managed OC-192 backbone, delivering 50 times the capacity of a frame relay network to the customer site. The Black Rocket Voice service extends the Black Rocket platform to now include next-generation business communication services and enables business continuity for enterprises with a potential savings of up to 40 percent in total data and telecommunications costs.

�By deploying Black Rocket Voice to 41 domestic sites, Cisco expects to streamline internal communications, using the 35,000 IP phones we�ve currently deployed,� said John Bruno, vice president, information technology, at Cisco. �Genuity�s Black Rocket Voice service is regarded by Cisco as an important milestone in our long-range plans to migrate many of our voice and data services to outside service providers and the key to ensuring the continuity of our business communications. Genuity�s network management expertise allows us to focus on our core business.�

For providing a converged voice/data solution for the enterprise while offering QoS guarantees and measurable service-level agreements, TMC Labs commends Genuity and we grant our TMC Labs Innovation Award without reservations.

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Jasomi Networks, Inc
PeerPoint Enterprise Edition
408-252-VOIP (8647)

Any administrator relying on a VoIP solution to deliver voice communication between an employee desktop and the Internet no doubt understands the Internet telephony idiom: �NAT issues.� While Network Address Translation (NAT) is a proven method used to secure data networks by converting the address of each LAN node into one external WAN IP address (and vice versa) preventing a direct path to internal hardware, it�s not quite as effective a system for sending and assembling voice packets correctly. Since most enterprises use NAT-enabled firewalls, many VoIP devices lose quality, partial functionality or are rendered entirely inoperable from behind such enterprise protection.

Enter Jasomi Networks� PeerPoint Enterprise Edition. The PeerPoint Enterprise Edition is a SIP-to-SIP gateway that translates VoIP between an internal, private IP address to the external global address space, enabling calls to pass into and out of the organization across the NAT boundary. Jasomi accomplishes this by employing what they call a Back-To-Back User Agent (B2BUA) combined with an integrated media proxy; one side of the B2BUA handles internal call flows on the private address space, while the other side handles external flows on the global address space. PeerPoint essentially divides each VoIP call into two halves, the voice communication up to the firewall, and the internal communication behind the firewall.

This results in what looks like call termination at the box, in much the same way a IP-to-PSTN call is terminated, however while maintaining the external, incoming voice transmission, PeerPoint simultaneously begins another internal SIP session, communicating the voice information to its destination while performing all protocol translations and directing media streams between the both halves of the call. In this way PeerPoint Enterprise Edition uniquely and innovatively handles �NAT issues� providing the appearance of a single, uninterrupted VoIP call.

Contrary to what you may be thinking, Jasomi Networks says that deployment of PeerPoint Enterprise Edition does not impose an additional load on an incumbent firewall affecting its traffic, security, fraud protection, or topology hiding. Using B2BUA allows an enterprise to use VoIP without gating to a circuit-switched network, thereby minimizing costs related to equipment, services and telco charges. Since it�s an out-of-box solution, Jasomi also says �PeerPoint Enterprise Edition can be installed without controversy, and configured in a matter of minutes.� Engineered for housing within an enterprise subnet, or DMZ, the unit also allows companies to utilize Communication Application Service Providers (CASPs) that otherwise would not have access due to equipment limitations.

Jasomi Networks has defined SIP-to-SIP solutions on several different levels. In addition to the Enterprise Edition, they�ve created a PeerPoint for carrier and service provider networks to manage and control SIP streams that, among many additional features, have the ability to perform Communications Assistance for Law Enforcement Act (CALEA) legal intercept. Since all signaling streams pass though PeerPoint hardware, the device has the capability to examine, intercept, duplicate, and forward copies of call information to a central collection point.

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Mediatrix Telecom, Inc.
After TMC Labs tested the APA III-4FXS product for the Jan �02 issue of Internet Telephony magazine, we realized first hand what a truly unique product Mediatrix had developed. Though voice gateways alone aren�t quite as innovate as they were three years ago, add a four-port FXS interface, remote management, PSTN fallback; pack it all into a small box and employ full-duplex and silence suppression along with industry-standard compression codecs, and this device is in a space all its own.

The APA III-4FXS is a standalone device compliant with all the major voice protocols: H.323, SIP, and MGCP. This telephony adapter connects up to four analog terminals to a LAN providing a gateway for packetized audio communication. It additionally supports the features provided with Analog Display Service Interface (ADSI) phones such as the Nortel Vista 350. Along with voice, G3 and Super G3 fax transmissions are also supported at speeds up to 33.6 Kbps. Real-Time Fax Over IP (FoIP) with T.38 protocol stack can be used and automatic fax mode detection is available on all ports. Similarly, all ports support standard codecs: G.711, G.723, and G. 729. The unit�s software is also upgradeable via TFTP and supports DHCP.

The APA III-4FXS also has a fifth RJ-11 jack used for connection with a PSTN line. During normal operation the line is �switched out� or excluded from the circuit. When the power is disconnected from the unit, or the power fails, the relay setting is restored to a connected state and the PSTN line can be used as an emergency bypass line.

When power is restored to the unit, the APA III-4FXS does provide PSTN tones, just as if a user were on a POTS line. On any of the four RJ-11 ports dial tone is provided when a connected handset is lifted from its cradle. Similarly, call progress tones such as ring back or a busy tone are also provided.

As a SOHO gateway with four FXS ports employing H.323, SIP and a PSTN fail-safe the APA III-4FXS appears to be a unique solution. Add upgradeable software, DHCP support, and G3 fax support and the unit is definitely innovative. Couple this feature set with the Mediatrix IP Communication server compatibility and you�ll be hard pressed to find a standalone SOHO gateway that can be deployed in various network configurations as a PBX-to-IP solution, an enterprise CLEC solution, used as an IP-PBX, or to connect branch offices.

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NetIQ Corporation
VoIP Manager Suite

The VoIP Manager from NetIQ is a software solution that actively monitors VoIP call quality across IP telephony networks. VoIP Manager provides comprehensive real-time monitoring, management, and reporting on system health, call quality, in addition to the performance and availability of critical IP telephony servers, such as the Cisco Call Manager.

NetIQ�s VoIP Manager is doing far more than simply assessing and monitoring throughput and response time. While these criteria are effective for assessing and maintaining data networks, voice requires a different method. NetIQ approaches network assessment and management through provisioning of its nine modules dedicated to uncovering a VoIP network�s golden mean. NetIQ targets the areas that have the most effect on a system�s readiness to handle VoIP traffic: codecs deployed, one-way delay, jitter, and types and degree of packet loss; and further uses these factors to calculate the MOS. Additionally, instead of exclusively using SNMP polling NetIQ incorporates Intelligent Agents that run directly on the network and interface with native Windows management capabilities.

Since VoIP Manager is entirely software-based, it does not require any special-purpose probe hardware, or simulation equipment used to collect network statistics. Instead, NetIQ uses what they call Performance Endpoints: small executable files designed to run in the background on any computer connected to the network. The Endpoints simulate traffic based using a manager�s choice of codecs to calculate a MOS.

NetIQ has also developed Connectors, which allow customers to use their existing management products in conjunction with VoIP Manager to correlate events on their network. The Connectors have been verified by solution manufacturers such as Cisco, to assess their VoIP products. Additionally, NetIQ offers connectors for Micromuse NetCool/Omnibus, Dell OpenManage, and HP OpenView Network Node Manager. For example, VoIP manager allows users to proactively monitor and manage Dell PowerEdge hardware performance when used (as an IP telephony servers) such as UPS battery levels, network interface card errors, and computer temperature.

VoIP Manager is a powerful, easy to deploy, out-of-box solution that enables administrators to assess, provision, and continually monitor VoIP on their networks. Incorporating automated VoIP management, problem resolution, automatic alert management, in addition to powerful reporting and charting tools make it an easy selection for a 2002 TMC Labs� Innovation Award.

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Symbol Technologies
NetVision Phone

 Symbol Technologies� NetVision Phone is the first to incorporate the rapidly growing 802.11b, and the widely adopted H.323 VoIP protocol, into one wireless handset offering new levels of enterprise functionality and mobility. The handset essentially provides the feature set of an Ethernet telephone and the mobility of a wireless handset based on the range of in-building Wi-Fi access point(s), or network.
Lying face up on an office desk, the NetVision phone could easily be mistaken for a Nokia or a Samsung cell phone; the circuitry underneath its familiar looking shell however, is quite a far stretch from a GSM chipset. The NetVision is both unique and innovative in its housing of 802.11b, ITU standard H.323, and POP3 technologies, in a small and fully portable handset. Symbol also provides optional support for Cisco�s Skinny (SCCP) protocol and enhanced H.323 solutions from Mitel and Nortel. Partnering with Ericsson, Nortel, Mitel, and Cisco, Symbol has designed the NetVision as an extension of offerings from these major IP PBX manufacturers, able to function with all the capabilities you�d expect from a native VoIP handset: peer-to-peer, extension, name, and speed dialing, call hold, Caller ID, park, transfer, conference, forward, call waiting, and do not disturb features.

The NetVision phone handles all the processing, compressing voice via G.711 and G.729a, packetizes and sends via an 11Mbps Wi-Fi network connection using the CSMA/CA wireless access protocol to either an H.323 gateway, or IP-PBX with built-in gateway functionality. The NetVision also houses embedded wireless communication features such as Symbol Wireless Voice Prioritization, which gives voice packets precedence on the wireless LAN, preemptive roaming to maximize bandwidth and sustain voice quality, and up to 10 active calls per wireless access point.

The phone itself stands about five inches in height, 1� inches in width, with a �-inch diameter. At about 5.5 ounces, it features a small three-line, 12-character display, and carries a rechargeable Lithium Ion battery. With three hours talk time, NetVision has its user�s choice of ringing options while offering vibrating call notification, text paging, intercom functionality and user configuration options as well.

Symbol also produces a similar product called the NetVision Data Phone, which addition to VoIP functionality, has integrated bar code scanning and Web-client data capabilities. Symbol plans to sell the NetVision Phone directly to its core customers and indirectly to its community of wireless VoIP telephony and channel partners through OEM and reseller agreements.

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Communications Suite

Working in conjunction with their Application Switch, the Communications Suite from Sylantro targets service providers interested in providing SIP-based, IP-hosted communications infrastructure. The Communication Suite provides a comprehensive set of hosted PBX and IP-Centrex applications that allow providers to offer many different degrees of functionality while allowing end users to forego high installation, hardware, and management costs. The suite consists of the following components: c-Business, ComCierge, ComTraveler, ComMerchant, ComRIO, and ComOffice.

c-Business handles the hosted PBX and IP Centrex functionality while additionally allowing a service provider to provide customers with a Web portal along with new directory and browser-based features: individual call logs, click-to-call, and single number support. e-Business also provides Find Me/Follow Me and other advanced features of the product.

A service provider and notable adopter of Sylantro�s solution, GoBeam�s DashBoard offering is based on the Communication Suite of products. TMC Labs took GoBeam�s product for a �First Look� application test in the April issue of CIS magazine http://www.tmcnet.com/cis/0402/0402lab2.htm Through our testing, we had the opportunity to interact firsthand with many components in the Suite, such as ComCierge. ComCierge provides call control, personalized call treatments, and CallerID, allowing end users to decide how their incoming calls should be managed.

ComMerchant focuses on distributed call center capabilities for SMBs. The ComMerchant module features a network-hosted ACD, one of the groundbreaking features of this product. The browser-based GUI makes it simple for users to check in or out of queues, display caller information, and balance call distribution.

Mobility is also factored into the Suite supporting travelers and remote workers through the implementation of ComRIO (Remote Instant Office) and ComTravler. Both armed with the provider�s offering and branding, ComRIO targets the remote worker by extending single-number and single-mailbox functionality, while ComTravler extends portal capabilities with WAP-enabled mobile phones allowing users to change reach me or click to call settings from their mobile phones � another feature we found to be quite unique.

The Communication Suite also offers support for legacy business phone sets. Sylantro promises to breathe new life into existing hardware by supporting LCD displays and assignment of features and speed dial to legacy phone feature keys. Sylantro is also using SOAP, part of Microsoft�s .NET initiative to provision click-to-call from any Web page and well as interworking with Microsoft Outlook, which helps comprise their ComOffice offering.

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Plexus 9000

While there are literally dozens of softswitch manufacturers, making it seem as though they are a �dime a dozen,� there are softswitches on the market that are clearly head and shoulders above the rest. Telica�s Plexus 9000 is a very flexible and highly scalable softswitch-based switching platform that certainly carries the mark of a premier softswitch manufacturer. Their solution is a softswitch-based packet switch for TDM, VoATM, and VoP (Voice over Packet) switching, supporting traditional and next-generation services.

Service providers deploying the Plexus 9000 can cap their investment in traditional circuit switches and eventually migrate to a converged switching infrastructure. They can use the Plexus 9000 today as a high-capacity tandem or Class 5 switch for traditional services such as 3/6/7/10 Routing, International Dialing, CLASS, E800, E911, LNP, and CALEA. Then when market conditions are right they can offer next-generation services such as IP-based voice services including IP-Centrex.

It features a fault-tolerant NEBS Level-3 certified architecture that Telica claims has an estimated reliability rate of 99.99994 percent. Also, the Plexus 9000 can easily handle a dynamic mix of circuit- and packet-switched traffic for Class 4 and Class 5 applications. The Plexus 9000 supports a comprehensive suite of interfaces for both subscriber and network access and hundreds of services for Class 4 and Class 5 switching applications The platform includes all the elements of a softswitch-based solution: Media Gateway, Media Gateway Controller (Softswitch) and Signaling Gateway.

Supported signaling protocols include SS7, ISDN, CAS, Session Initiation Protocol (SIP) for VoIP, Broadband Loop Emulation Services (BLES) for VoATM, and GR-303 for DS1 and Voice Frequency (VF) interfaces. It supports a variety of interfaces and protocols, including DS-1, E-1, DS-3, STS-1, OC-3, STM-1, OC-12, STM-4, OC-12c and 10/100/1000 Mbps Ethernet. Telica�s solution includes VoIP subscriber side access through industry standard SIP signaling enabling the deployment of new services such as IP-Centrex. It also includes support for SS7, MEGACO, M3UA/SCTP, and BICC. Additionally, ISDN, CAS, and GR-303 signaling interfaces allow the Plexus 9000 to leverage the large installed base of subscribers connecting to hundreds of existing TDM access devices such as PBXs or Digital Loop Carriers to deliver business and residential voice services.

One key advantage of the Plexus 9000 is its I/O density, which Telica claims is an industry-leading 137,088 TDM ports or 24,192 VoIP calls per shelf (13RU) supported and 411,264 per rack. It also features low power consumption (0.3 watts/DS1), which is very important in the carrier space. Its components are hot-swappable and you can configure �hot spares� for increased redundancy. It includes the required CLASS services such as calling number delivery, calling name delivery, call waiting, call forwarding, speed dialing, anonymous call reject, caller identification, distinctive ringing, call add-on and others. In fact, Telica recently added software to support call forwarding, call waiting, call reject if caller ID is blocked, speed dialing, distinctive ring, operator barge-in and the ability for law enforcement agencies to wiretap phone calls. These are necessary features to offer residential phone services that are traditionally delivered via Class 5 switches. Telica will also be supporting AIN 0.2 that enables Plexus 9000 to tap databases in existing phone networks to add even more Class 5 features. The Plexus 9000 is the perfect solution today to perform �toll bypass� while at the same time this platform is prepared for the future to offer enhanced services. By offering a solution for today�s and tomorrow�s needs, along with impressive density, scalability, and reliability, TMC Labs is proud to bestow our TMC Labs Innovation Award to Telica�s Plexus 9000.

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Voyant Technologies, Inc.

 Voyant�s MobileMeeting solution allows carriers to deploy a turnkey, instant conferencing solution connecting disparate endpoints across separate networks. Voyant has partnered with OZ and Odigo to create a wireless point of presence conferencing solution. Uniting OZ�s ICS 2.0, Odigo�s Open IM Server (OIMS) and Presence Management Server with Voyant�s ReadiVoice platform has resulted in a unique, reservation-less conferencing infrastructure for wireless carriers and service providers called MobileMeeting.

Utilizing the Innovox conferencing media server with ReadiVoice conferencing solution along with the aforementioned offerings from both OZ and Odigo, Voyant�s MobileMeeting allows users to initiate a full-duplex group voice session from any mobile phone, PDA, or PC on-the-fly. This capability represents a shift in traditional voice conferencing, bringing instant conferencing to subscribers anywhere and via any device. This solution also utilizes a carrier-grade, centralized contact and presence management system providing presence information and thereby identifying who is available to initiate a point-to-point or group conference.

Open APIs enable service providers to integrate MobileMeeting with an existing solution, perhaps to fortify an already branded offering and allow further service customization by integrating voice conferencing with presence management. Utilizing XML and speech recognition from Nuance, MobileMeeting is offering some unique conferencing capabities to subscribers: SMS or e-mail conference notification, personalized �welcome� conference greetings, as well as a Web portal and WAP access.

MobileMeeting speech recognition offers �hands free� conference initiation from any mobile phone utilizing conversational speech to invite and join participants into a conference. Via a WAP interface, mobile users can simultaneously dial out to multiple participants by selecting individuals from a contact list. The more detailed address book and group applications can be edited from the Web portal. The Web portal additionally allows members to dial out and control calls. Additionally, when dialing into or creating a call, the conference host has the option to alert participants via e-mail, or SMS notification.

The uniquely innovative MobileMeeting platform offers anytime, anywhere voice collaboration to the end user featuring speech recognition and multiple points of presence using standard devices. Deployed by a carrier or wireless provider, this solution may also offer an enhanced revenue opportunity through its distinctive service offering.

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Webley Systems, Inc.
CommuniKate Unified Communications

Webley�s Media Switching Platform (MSP) supplies CommuniKate with a foundation for providing both traditional PSTN services and IP-powered services to carriers. With Web-based tools, XML, speech technology, standard SIP protocol, and scalable architecture, these offerings foster the carrier and service provider deployment of Webley�s voice-enabled unified communications suite. The switch and CommuniKate together provide a platform and services that not only converge IP and PSTN communication, but also their respective services.

Communi-�Kate� is the voice-enabled, phonetically-British, hands-free virtual assistant tasked with helping its users navigate through hosted PBX-functionality, voice mail, e-mail and fax, calendar, PIM information, as well as setting up conference calls, and adding contact information. CommuniKate�s unified messaging service offers integrated communication between voice mail, e-mail, and fax retrieval. Listening to e-mail messages over the phone, or accessing your voice mail via a browser are two reasons why carriers such as WorldCom have deployed Webley�s solution. Other messaging features allow users to send, respond to, or forward messages to anyone else on the CommuniKate powered network with either the touch of a key or voice command. Additionally users can simultaneously broadcast any message type one or a hundred recipients while using the address book and pre-established distribution lists.

Conferencing and Calling Services provide the means to connect directly with individuals in different ways. Conferencing-On-The-Fly, for example, allows a user to conference in a phenomenal 32 parties, without having to set up the call ahead of time. While traditional conferencing options are available, Webley also offers a Calling Party Pays service, which charges each party equally for the call. Though conferencing is effective when people know where to reach you, Calling Services can track you if you�ve strayed from your office: One Number Reach Me, Find Me Forwarding, Call Blast (transfers four calls to one destination), and Transfer Rules.

Webley�s CommuniKate is also integrated tightly with PIM management tools such as an Address Book capable of storing up to 2,000 contacts and a Calendar that will remind you about its documented events via phone, e-mail, pager, or simultaneously notify using all three methods. A simple voice prompt such as, �Call my contact �Jack Walsh,�� instructs CommuniKate to initiate the call using the phone information stored in your address book.

Webley assigns the user a toll-free number, which when dialed has the capability of �call blasting� the user at five possible endpoints. These endpoints can be any traditional phone, such as a cell phone, office phone, etc., but the real beauty of Webley is that it supports SIP endpoints. Ultimately, this allows both PC-based phone and presence features to exist on users� desktops while enabling call origination and reception without the use of a traditional handset. Thus, a user could simply enter their SIP address as one of their possible endpoints and Webley would attempt to reach the user on any SIP client, such as MSN Messenger or PingTel�s SIP phones.

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Empirix Hammer NXT

Since the TMC Labs Innovation Awards are granted to products that are truly �unique� or innovative, we had yet to award the same company with a second TMC Labs Innovation Award. After all, it�s difficult enough for a company to develop just one innovative product � much less two. Thus, when we gave Empirix an Innovation award for their H.323 Call Generator in the July 2000 issue, we did not expect to grant Empirix a second Innovation Award any time soon. Well, Empirix proved us wrong when they launched Hammer NXT, an integrated carrier class TDM/IP test platform running of a single 21-slot CompactPCI chassis that can generate up to an impressive 50,000 simultaneous calls using a mix of TDM over DS-3 interfaces using SS7, CAS and ISDN signaling, and SIP/IP with a real RTP media. Multiple NXT platforms can be combined to achieve traffic volumes of over 250,000 simultaneous calls.
Running on an embedded dual 1-GHz single board along with a blade-based, modular architecture, performance and expandability are non-issues. The system also supports the Hammer Voice Quality Test Suite (VQTS) option, which provides TDM-to-TDM and IP-to-TDM voice quality testing. It measures QoS with real voice, including packet loss, latency, and jitter. It also performs all the standard voice quality scoring algorithms such as PSQM and PAMS. We�re told subsequent Empirix NXT releases will support MGCP and MEGACO signaling in addition to STM1 and OC3 interfaces.

We asked Empirix whether or not the Empirix NXT was simply the �old� Empirix software ported to a more scalable hardware platform. We were informed that the Hammer NXT utilizes base designs from industry leaders such as NMS Communications and Advantech that designed and developed custom versions of their components to address the challenging requirements of Empirix�s carrier-class converged test platform. In fact, the Advantech 11U height CPCI chassis was designed especially for the Empirix NXT.

In addition, Empirix mentioned that they are using enhanced versions of the TestBuilder User Interface and Hammer Test Server that are uniquely designed to address high-density performance testing and deliver converged test capabilities. We should also mention that the NXT also integrates with the PacketSphere network-degradation emulator, for simulating real-world conditions. The Hammer NXT enables engineers to isolate and test network edge devices with just a single system rather than a plethora of separate test equipment. The Hammer NXT�s modular, blade-based architecture allows systems to be configured with flexible mixes of TDM, IP and signaling capabilities.

With the Hammer NXT, equipment manufacturers and service providers can verify multiple next gen applications such as toll bypass, and Class 5 enhanced services. Empirix has set a new bar with their high-density, scalable, carrier-class NXT test platform. Empirix is our first two-time Innovation Award winner and we couldn�t have selected a more worthy product.

[ Winners' List ]
[ Return To The July 2002 Table Of Contents ]

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