TMC Labs Innovation Awards – Part I

By: Tom Keating (News - Alert)

Innovation is defined as the introduction of a new idea into the marketplace in the form of a new product or service, or an improvement in organization or process. TMC (News - Alert) Labs has been testing, examining, and reviewing products since 1994 and the favorite part of our job is seeing new products and technologies. It has been truly remarkable to witness the evolution from proprietary PBXs with CTI (News - Alert) links to PC-PBXs, then from PC-PBXs to IP-PBXs, and finally to unified communications platforms. As such, one of the most exciting parts of the year is to review the most innovative and unique products and services in IP communications for the TMC Labs Innovation Awards. While others have tried to copy the premise of these awards recognizing the true innovators in IP communications, only the TMC Labs Innovation Awards goes back 9 years. Further, TMC Labs takes great pride in the stringent selection process for this award which typically only grants a couple dozen recipients as winners. 2008 was no exception with only 24 winners for this prestigious award.

This year marked several strong contenders in these specific areas: mobile, testing tools, video, and unified communications. In the unified communications space we had winners from Iwatsu, Mitel, Siemens, and Toshiba. We had some interesting HD (high-definition) video offerings from LifeSize (News - Alert) and Polycom.

TMC Labs is proud to announce our 24 picks for this year’s TMC Labs Innovation Awards, 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 Adtran this month and ending with Polycom. Next month, we start with Radware and end with Zed-3 (News - Alert).

Adtran, Inc.

The NetVanta 1335 Multiservice Access Router is a unique platform that combines multiple networking devices onto a single compact platform. The NetVanta 1335 is the first product to integrate a modular IP access router with a 802.11a/b/g wireless access point, a 24-port Layer 3 Power (News - Alert) over Ethernet (PoE) 802.3af switch, firewall, VPN appliance, a DSU/CSU, and a SIP ALG (Application-level gateway). The uber-comprehensive NetVanta 1335 features VQM (voice quality monitoring), QoS, NAT, and advanced firewall functionality. Some of the wireless security features include the ability to disable SSID broadcasts, 802.1x Authentication, and WPA/WPA2 for protected key exchange. In addition to serving as a wireless access point, the NetVanta 1335 acts as a WiFi (News - Alert) controller for up to eight NetVanta 150 WAPs

It also sports Gigabit SFP/1000Base-T Ethernet uplink port(s) for stacking capability allowing you to stack up to 16 switches using a single IP address to manage and up to 8.8 Gbps switching capacity that is non-blocking. Importantly, it includes both a “techie” CLI (command line interface) that networking experts are familiar with, as well as a web interface for techies and non-techies alike. Another unique feature is Voice Quality Monitoring, a graphical interface that allows network administrators to graphically monitor the voice traffic on their networks. Voice traffic can be identified via graphs and charts down to the individual call level.

Another innovative feature is that it has the ability to diagnose 10/100/1000Base-T twisted-pair Ethernet cables. This Cable Diagnostics feature can determine the length of the cable, if the Ethernet

cable has an Open or a Short (based on pair), and can even detect the location of the Open or Short within a meter.

Most importantly, by unifying all of these network elements onto a single platform, you simplify administration and reduce the TCO making this an optimal solution for the SMB market. Additionally, it’s a single vendor-solution and a single point of contact for service and support. ADTRAN (News - Alert) told TMC Labs, “ADTRAN’s NetVanta products address the needs of the SMB and SME in a unique way. These products are specifically designed for this market. Hence, they are not larger more expensive products that have been scaled down to address this market. They have the features and functionality needed to address SMB/SME applications. Again, there may be platforms with more functionality on the market, but is that functionality really needed and can the SMB/SMEs afford to pay for something they are not using?”

AudioCodes (News - Alert)

The Mediant 1000 MSBG (Multi-Service Business Gateway (News - Alert)) is truly an innovative product — targeting SIP Service Providers and large distributed enterprises that use SIP to interconnect their services or remote sites. AudioCodes Mediant 1000 MSBG is a unique, all-in-one and modular multiservice access solution for Enterprises and Service Providers offering managed services such as IP-Centrex or SIP trunking. The Mediant 1000 MSBG is based on AudioCodes’ Media Gateway technology, combined with enterprise class Session Border controller, data & voice security, data routing, LAN Switching and WAN access.

The key innovation for the Mediant 1000 MSBG is the combination of a full-featured and widely interoperable media gateway, data router, enterprise-class Session Border Controller and application platform in one compact device. Additionally, customers can start with TDM trunking, and then add SIP trunking while keeping the TDM trunking as a survivable back-up or local access point for E911.

The optional Open Solutions Network (OSN) application server allows Windows or Linux-based applications to operate within the Mediant 1000. Essentially, you can install Asterisk, Microsoft (News - Alert) OCS, or other IP communications apps right on the device. In fact, the Mediant 1000 is compatible with several SIP software applications, including Microsoft OCS and Exchange, Broadsoft, Sylantro, Avaya, Nortel, 3com, Genesys, and Interactive Intelligence (News - Alert). Adtran claims that the Mediant 1000 MSBG is the first MSBG with a truly modular architecture and to include an option for a completely open partner application server. This allows customers to install on-premise software within the OSN server for on-site IP-PBX (News - Alert) functionality, messaging or survivability features, etc.

AudioCodes told TMC Labs, “For the service provider, combining these devices into one package solves a real challenge they are facing with controlling the point of demarcation, plus it eliminates a lot of separate devices and simplifies the installation process.” They added, “Another very unique aspect of the product is the migration path — allowing customers to start with a standard Mediant 1000 media gateway, then using modules and software options, add the functions needed within the same chassis.” Lastly, the latest version, 5.4 Release also added IP-to-IP mediation (for IP transcoding) and the new Enhanced G.711 (EG.711) from Global IP Sound (News - Alert).

Challenger mobile

Mobile Business Challenger MBC AB, a company based in Sweden, offers a unique mobile VoIP solution called ‘Challenger mobile’ targeting mobile operators (MNO and MVNO) and related service providers (ISP, WISP), as well as retailers. From an end-user perspective Challenger mobile could be compared with Skype (News - Alert), Fring, or Truphone, but they are end-user services and do not offer operators a white label platform. “Challenger mobile develops and markets its white label Mobile VoIP platform that empowers mobile operators, wireless ISPs, and other service providers to deliver their own branded Mobile VoIP services to subscribers, with a rapid time-to-market and low barriers to entry,” stated a Challenger mobile spokesperson.

Challenger’s white label platform allows subscribers to place calls from their mobile phones over the Internet, using a WiFi, GPRS or 3G connection. In addition to core Internet telephony capabilities, Challenger supports remote mobile handset configuration, call detail record (CDR) support, call termination, and flexible CRM & billing system integration.

Challenger mobile explained, “Our platform allows operators to configure their end-users’ phone with a “virtual SIM card” (just like you need a SIM card to make GSM calls) that enables their end-users to make mobile VoIP calls. In order to offer mobile VoIP, operators or service providers need to integrate the end-users handset into a VoIP network. We help them do this, by configuring the mobile device as a standard SIP client, which then uses the operator’s data network to send and receive SIP-based VoIP data packets. SIP-based mobile VoIP can utilize any IP-capable broadband wireless network connection such WiFi or 3G.”

They also pointed out the power of mobile VoIP to break into new markets — “When operators make a decision to launch mVoIP, time will be essential in defending their market from VoIP and mVoIP erosion. Also, if an operator wants to go in to a new geographical market, Challenger’s mVoIP platform can be used as a tool to break in to a market without having a GSM or 3G license. The mVoIP customers can, in a second stage, be converted to GSM/3G customers.”

The process is very straight-forward for users. The user registers for the service, then receives an SMS (text message) that helps configure and provision the handset. Once the phone is provisioned, the user can either make free mobile VoIP calls to other SIP-based phones, or make calls to any other international/long distance number at low rates.

Lastly, they told us they are developing a SIP client for downloading into mobiles that don’t have a native SIP client, to enable a large number of mobile handset models to be able to use Mobile VoIP service. The client will also include integration with their call-back service, enabling low cost rates for international calls without a WiFi connection. The SIP client should be available by the time you read this.

Citrix Online

GoToMeeting enables SMBs to have a cost-effective managed online meeting service that makes communication with employees, partners, customers, and prospects easier and more efficient. GoToMeeting users can collaborate on documents, deliver presentations, perform product demonstrations, and securely share confidential information from anywhere. Its screen-sharing technology allows presenters to share PowerPoint slides, financial spreadsheets or any other PC application with all participants in real-time. Users meet with others online to share, discuss and edit any document. Attendees do not need their own account to join a session and they do not need to pre-install any software.

The latest innovation with the launch of GoToMeeting 4.0, is the introduction of VoIP to go along with their PSTN capabilities. VoIP opens the door to inexpensive multi-national meetings as well as the convenience factor of not having to dial a conference bridge phone number. One nice feature is that an organizer can see who is talking and even selectively mute an attendee. Importantly, there will be no anticipated increase in the price for web conferencing with VoIP, while some competing providers charge for VoIP. GoToMeeting is built using patented bandwidth-adaptive compression technology, ensuring fast performance and real-time screen updates along with true 24-bit color representation. Powerful features include sharing of keyboard/mouse, drawing tools, and desktop recording & playback, Mac support, and Office integration.

Digium

Switchvox is a comprehensive IP-PBX based on Asterisk (News - Alert) with some unique features perfectly suited for the SMB market with a very attractive price-point. The latest version of the software, Switchvox 3.5 integrates with key CRM applications such as Salesforce.com and SugarCRM (News - Alert) through Switchvox’ SMB’s Switchboard interface. One “innovative feather” in Digium’s cap is that Switchvox is the first product to incorporate sales and customer service “mashups” directly into an IP PBX system. This certainly improves the ways companies interact with customers, potential leads and outside parties. Customers can easily create custom mashups that leverage other Web applications, including Web 2.0 tools such as Google (News - Alert) Maps.

According to Digium, “Its ease of use, ability to be customized to support each customer’s business processes and workflow, and support for mashups make Switchvox 3.5 the ideal choice for any SMB phone system.” They continued, “The ability to integrate Web 2.0 tools through mashups is a key element that had been missing in the SMB telephony industry and allows SMBs to take full advantage of communications technologies available to them in one simple package.”

Switchvox is very easy to setup and manage even for non-technically staff, which results in lower TCO. Switchvox’s Switchboard is Web-based, so it runs on any OS (Windows Mac or Linux) and is easily updated across the whole system at once. Newly added or improved features include autoprovisioning of phones, multi-level administration with security levels, and batch direct inward dial (DID) mapping which allows administrators to quickly map all of the company’s extensions to direct dial numbers rather than keying them into the IP PBX one at a time. Similarly, you can perform batch caller ID configuration which lets you specify per extension outbound caller ID for all extensions with one simple setting. Finally, Switchvox is backed by Digium, the company that founded the popular open source Asterisk movement and includes their excellent technical support staff.

EMC Corporation

EMC Smarts VoIP Performance Manager delivers performance data and reporting to help ensure high call quality and reliability. Using VoIP Performance Manager, you can manage, monitor, and diagnose VoIP services with its built-in intelligent alerting, deep diagnostics, and extensive reporting.

VoIP Performance Manager currently supports Avaya Communications Manager 3, Cisco (News - Alert) Unified Communications Manager 5, Cisco Unified CallManager 4, and Cisco CallManager 3.EMC Smarts VoIP Performance Manager simplifies the task of managing large IP telephony deployments by providing you with one view across Avaya (News - Alert) and Cisco technology. According to EMC, “VoIP Performance Manager saves global enterprises and large MSPs time and money by eliminating the need to purchase and maintain multiple tools, reducing the time and expense involved in training staff to use a variety of solutions, and enabling delivery and measurement of common service levels across multiple technology platforms.”

VoIP Performance Manager delivers real-time information about phone extensions, phone calls, voice quality, availability of the telephone service, and interconnections to telecommunications providers. Importantly, the solution reduces the total cost of ownership (TCO) by giving you the performance data you need to optimize equipment overhead, bandwidth, and trunk capacity, perform capacity planning and avoid making unnecessary new purchases.

Monitors, measures, and manages the availability and performance of your VoIP services and systems as well as the network services supporting them through integration with EMC Smarts VoIP Manager and EMC Smarts VoIP Performance Reporter. VoIP Performance Manager alerts you if a route pattern has degraded or failed, and lets you drill down to determine which trunks or gateways have caused the problem. Additionally, it lets you map any of its thousands of metrics to your service-level agreements (SLAs). With these performance metrics VoIP Performance Manager can alert you to potential or current SLA breaches.

Iwatsu Voice Networks

Iwatsu has embraced SIP in a big way and not for the reason you’d expect. Previously, the Iwatsu Enterprise Suite used analog cards as the interface between the voice processing server and the UC (unified communications) server. The major revision over the last 12 months involved completely removing the analog interface and using a pure IP connection; in this case, SIP. Iwatsu Voice Networks designed a SIP connection to provide an access path for assets within the Iwatsu Enterprise Suite. By using SIP it decreased hardware costs by eliminating analog hardware in the Iwatsu Enterprise Suite resulting in approximately 25% cost savings to the customer.

The target market for the Iwatsu Enterprise Suite (ES) 1.0 ranges from 20-400 users, which is a wider range than most phone systems. It can hit both the low-end (20) and the medium-sized business (400) using the same product. Usually, when phone systems scale you need a different product, often one that is more expensive per port or extension. The Iwatsu ES is unique in that it uses a SIP interface and therefore system expansion is a simple matter of additional software licenses. Other systems, for example, can reach physical capacity as the business grows, necessitating additional hardware. SIP removes this limitation.

The Iwatsu Enterprise Suite consists of a full featured UC (unified communications) server and the voice processing server. The full featured UC server is the Iwatsu Enterprise TOL, and the voice processing server is the Iwatsu Enterprise-CS. Together, they make the Iwatsu Enterprise Suite. It’s unified communications capabilities include speech recognition to route calls appropriately, the ability to send and receive faxes from email, voicemail-to-email with full synchronization, and it includes tailored presence management.

Iwatsu proudly claims, “Other manufactures can begin to approach these features, but they do so piecemeal: each application requires a separate server. Needless to say, this can be expensive and an integration nightmare. The Iwatsu Enterprise-Suite combines all aforementioned features and adds the ability to manage all UC applications using a mobile phone. So for review: one platform, complete UC using SIP as the communication path.”

Iwatsu Enterprise Suite will now support GoogleTM Apps. This will allow small businesses that do not require an on-premise Microsoft Exchange environment can use Gmail — a fully-feature email component of Google apps—to deliver UC features. The Iwatsu Enterprise Suite uses IMAP to integrate with Gmail with full synchronization. Additionally, you can get innovative click to dial functionality from Google contacts, Gmail, Google Talk or Google Docs. This integration also enables message lighting on phones when using Gmail. Future versions of Google integration will allow for full featured presence capability, including using Google Calendar integration to provide user access to their daily agenda over the phone via text-to-speech technology. Iwatsu told TMC Labs this could not be possible without using SIP.

Junction Networks

Junction Networks targets the SMB market with an advanced IP phone system without the large costs associated with one. Its hosted onSIP Hosted PBX solution enables workers at various locations, including teleworkers to simply use their IP phones and features over broadband Internet connections as if they all were physically attached to an on-premises PBX. Junction Networks offers traditional PBX enhanced functions that is available on an à la carte basis allowing customers pay only for what they need, not a fixed collection of functions. Junction Networks cuts SMB telecom costs further by charging nothing for intra-company, extension-to-extension calls or calls to SIP phones anywhere.

Importantly, Junction Network’s onSIP hosted PBX offering combines basic SIP calling and PSTN gateway services with a comprehensive bundle of advanced PBX features, for one discounted price per customer, not per line or user. This is unique since many competitors do charge per line or user. While typical hosted PBX services charge $29, $39 or $49 per month per extension, onSIP is not priced by number of lines, but by usage of enhanced features and off-net calling.

The customer premise need only have any SIP-compliant phone or softphone for all phone functions to operate, and a web browser to be able to access the onSIP administrative interface. From the web tool you can make changes to call routing, pick up voice mail as email, and set up or remove new extensions and features. The hosted location holds the onSIP platform, built by Junction Networks using open-source software components, including the use of Open Source SIP servers for routing and Asterisk for enhanced services.

Customers can set the hosted PBX service up in minutes, add or drop lines or features themselves, and rely on Junction Networks to handle its customer support directly. Another unique feature is that it allows several IP phones at multiple locations to ring simultaneously at one number, so end users can place or pick up calls from home, office, or any other location.

In February on 2008, the onSIP service added the Inbound Bridge, an accessory service that ties in third-party providers of international and domestic DID numbers. Inbound Bridge saves money for onSIP customers by allowing them to find the best per-minute price for inbound VoIP calling minutes in their chosen geographic regions.

LifeSize Communications

Videoconferencing is becoming more and more popular, especially as a fuel and travel costs continue to rise. But simply using a desktop webcam for a professional business meeting won’t suffice. Step in the new LifeSize Express with the new LifeSize Focus integrated HD camera and microphone array claims to be the world’s first high definition video communication system with a retail price below $5000. The system delivers an immersive, high definition telepresence experience over existing broadband networks. LifeSize Focus features a high definition video camera that delivers full HD 1280x720 video (720p) at 30 frames per second. Its high-end camera provides outstanding image quality in a wide range of light conditions. It also features an integrated two-microphone array with beam-forming technology for superb sound quality. By eliminating external microphone pods and integrating the microphone array in the camera housing, LifeSize Focus saves space on conference room tables which is often at a premium.

LifeSize has several innovative firsts, including being the first company to deliver a high definition video conferencing system, LifeSize Room, in 2005. They were also the first to bring the price of high definition below $10,000, with LifeSize Team, in 2007. LifeSize Express with Focus is very compact — about the size of a wireless router and it provides HD video at 1MBps. It is fully interoperable and standards-compliant with H.264 HD and H.263 Standard Definition video conferencing systems. The device can be connected directly to any HD monitor and in addition to H.264 compliant HD video it also features wideband audio.

Mitel

The Mitel Communications Suite, a pre-integrated unified communications solution, allows Mitel’s flagship 3300 IP Communications Platform (ICP) software and other related applications to be integrated as a single seamless communications solution for business, resident on a standard Sun X4150 SunFire server.

Features include support of up to 5,000 users per server and SIP Application Integration for unified messaging, conferencing, and Microsoft Office Communications Server 2007. In addition, you can use a single software stream for deployments from 10 users up to 65,000 users in a clustered configuration. The platform has HTML and XML support for business process integration at the desktop telephone and SIP Trunking capabilities. It can be deployed on the customer premise or offered as a hosted service. If hosted, the Mitel Communications Suite is remotely located providing connectivity and service over an IP-based MPLS or broadband connection.

Mitel told TMC Labs, “The tight integration of call control and communications applications into a single package offered on a commercial server for the medium business market, we believe, is an industry first. Mitel is the first voice partner for Sun Microsystems (News - Alert). This is the first IP voice solution offering from Sun which is ‘Powered by Mitel’ branded.”

The Mitel Communications Suite delivers simplified system installation and management, business process improvement, and a reduction in operating costs by consolidating low-power consumption telephony and server hardware platforms for IT and unified communications applications. Mitel will add its Mitel Applications Suite solutions base to the same high performance Sun Fire X4150 server that carries the rich features of the Mitel call control. This will allow customers to consolidate applications such as Mitel NuPoint Messenger IP, Mitel Live Business Gateway, Mitel Teleworker Solution, Mitel Mobile Extension, and Mitel Audio and Web Conferencing all on a single server.

This solution also provides SIP / CSTA gateway connectivity for Microsoft’s Office Communications Server 2007 and allows for the extension of Microsoft presence capability to a host of Mitel applications including NuPoint Messenger IP, Mobile Extension, Customer Interaction Solutions, operator console, and more.

Phybridge

The Phybridge UniPhyer is a Layer 2 Ethernet appliance that enables unique voice and data network convergence with Power over Ethernet (PoE) and guaranteed Quality of Service (QoS). It actually leverages your existing phone cable network and your existing data cable network to create what Phybridge calls a “Parallel Voice Network (PVN)” to deploy IP telephony with no network downtime. Essentially, the appliance leverages your existing copper phone wiring (RJ11) to run the voice IP packets over, thus separating the VoIP packets from the data packets that run over traditional RJ45 network cables. This results in two separate networks — one for voice and one for data, which will result in better quality of service (QoS) since even a heavy load on the LAN won’t affect the separate VoIP network. They claim the appliance is quick and easy to install, offering centralized PoE, and a dedicated voice path to each desktop. The UniPhyer is IP PBX and phone agnostic and deployment is very fast.

Here’s how the process works, you simply:

• Disconnect the existing TDM PBX.
• Install the chosen IP-PBX on the LAN.
• Connect the Phybridge UniPhyer to the TDM amphenol connectors.
• Attach the UniPhyer to the LAN through one of the UniPhyer’s Gb. Ethernet connections.
• Attach the UniPhyer PhyAdaptor to the RJ11 telephone jack at the user’s desk.
• Attach the IP telephone of choice to the RJ45 jack on the UniPhyer PhyAdaptor.

Phybridge has invented and filed a patent regarding carrying both PoE and signaling over two wires. As a result business can leverage their existing legacy telephony infrastructure to deploy IP telephony. Phybridge told TMC Labs, “The current methods require a leap of faith in the pre-deployment phase — issues can include cabling, Quality of Service (QoS) and constant monitoring of the network. Phybridge is the first to address these issues and offer hardware appliances that solve deployment challenges and enable unified communications and converged networks, all at a fraction of the cost.”

Polycom

The Polycom HDX 4000 is an integrated desktop visual communications system offering UltimateHD — comprehensive HD video, HD audio and simultaneous HD multimedia content collaboration. The HDX 4000 allows users to talk face to face and share virtually any type of PC content in high definition, simultaneously. The HDX 4000 works seamlessly with Polycom telepresence solutions, as well as standards-based HD, SD and other video conferencing systems from Polycom and other manufacturers with its support for H.320, H.323, and SIP.

What’s unique about the HDX 4000 is that it is designed as a complete video collaboration solution, meaning you don’t need a separate display and it can be easily moved as needed. The Polycom HDX 4000 solutions are designed to fit on an executive desktop as a personal system, at a small table, or in a small conference room to support up to four people on camera. HDX 4000 systems contain all necessary components including an integrated 20-inch 16:9 display that can double as a PC monitor (up to 1680 x 1200 resolution); a built-in HD camera with pan-tilt-zoom capability; integrated dual HDX microphones that are immune to GSM interference (mobile phones, PDAs), high-fidelity speakers and subwoofers; a powerful, standards-based HDX video conferencing codec; and a stand with an integrated keypad that allows users to dial or answer video calls just like a telephone. It also features HD voice (22kHz stereo audio) and simultaneous HD multimedia content sharing capabilities (dual HD streams for video and content; content shared in native resolution).

The HDX 4000 features Polycom Lost Packet Recovery (LPR) technology, a forward error concealment technology that improves the quality of video communications over IP networks that may experience packet loss. This technology also helps make HD video collaboration more viable over sub-optimal network environments that may experience packet loss or network congestion when communicating with corporate offices. Polycom LPR technology delivers outstanding video, voice, content quality with up to 10% packet loss.

Perhaps the most innovative feature is that it can deliver ‘People On Content’, or chroma key technology (a.k.a. green screen), that allows people to become part of their content during a call. This enables users to do that with any type of content running behind them (video, static images, CAD drawings, animation, etc.). IT

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