First Quarter 1998

Internet Telephony Gateways

By now, you've undoubtedly heard that Internet telephony (aka Voice over IP, IP Telephony, etc.) is the wave of the future. In fact, as is often the case with Internet technologies, the future is here. You've done some research, read some analysts' opinions on the future of the market, choosing to side with believers who make claims like "The Internet telephony market will approach $7 million trillion by the time you finish this article." You think to yourself, "How can I get in on this?" To begin with, you'll have to convince the suits that hold the purse strings that it's high time to buy into this burgeoning technology. Next, you'll need a tool to help you sell your idea. This article is the perfect place to start. Now, who do you turn to to buy your hardware? Or does it make more sense for you to build your own? In the next 12 pages or so, you will be introduced to the companies that have already come out with the products and solutions you're looking for.

These products, known as Internet Telephony Gateways, are available in a variety of configurations, ranging from single-user gateways to full-blown 96-port systems. Some are already compatible with the latest standards, such as H.323, the rest are planning to comply in the near future. All claim to offer the clearest quality on calls from here to Timbuktu. Some do, others... not quite yet. But, check them out for yourselves. Do your homework. And keep on it. Last year, you could count on one hand the number of companies building gateways. Several months ago, maybe 15-20. We've gathered up just over 30, but if we did this same feature every month, that number would certainly continue to grow and grow.

The playing field is so hot right now, and so rife with competition, that this is by no means a complete listing. Many companies are just buying in to this technology, alliances are being forged, new players are appearing on the scene every week. And not just small-time start-ups. Networking giants, telecommunications providers, fax companies, and others are jumping on the Internet telephony bandwagon, fearful to be left behind. There's just too much money to be made. Traditional telephony board vendors are doing brisk business in Internet telephony development platforms too.

We have broken down our listing into several categories: Development platforms; "Traditional" gateways (how can something so new be called traditional?); Fax gateways; Video conferencing gateways; and a category we, interestingly enough, call "Other."

This list is to be considered a starting point for those who are interested in purchasing or creating a gateway of their own. Read on. Get some ideas. And whatever you do, contact the vendors. Visit their Web sites for more information. And hurry up. This is after all, "Internet time" we're talking about.

- Greg Galitzine
ggalitzine@tmcnet.com

Dialogic Corporation
1515 Route Ten
Parsippany, NJ 07054
www.dialogic.com

Dialogic's DM3 IP link servers are designed to enable individuals to communicate directly over data networks in a variety of ways: phone-to-phone, PC-to-phone, phone-to-PC, fax-to-fax, and Web browser-to-phone. Based on Dialogic's DM3 mediastream resource architecture the DM3 IPLink family of server platforms provides support for standard protocols, such as H.323 and G.723.1, both specified by the IMTC (International Multimedia Teleconferencing Consortium) voice over IP forum. Furthermore, the IPLink platform supports standard coders, such as G.729a, G.711, GSM, and Voxware RT24. It provides an open development framework, which enables custom system configurations and field downloads of software upgrade modules.

A recent addition to the IPLink family of products, Dialogic's DM/IP2000 is an enhanced IP telephony platform with an integrated on-board high-speed Ethernet interface, which results in greatly reduced latency, while increasing scalability and system reliability for developers. Since all data/voice packets are handled directly on the board, there is a reduction in end-to-end delays. Scalability is enhanced because the TCP/IP and H.323 stacks (which run on-board) reduce the host CPU load factor, also enabling reduced call setup times and a smaller host software footprint. The platform has proven to be compatible with leading H.323 applications such as Microsoft's NetMeeting, VocalTec's Internet Phone, and Intel's Internet Video Phone.

TAP-804 DSP Resource Board

Analogic Corporation
8 Centennial Drive
Peabody, MA 01960
www.analogic.com

Analogic's TAP-804 DSP resource board is an open DSP-based computer telephony processor that is designed to deliver a full trunk's worth of audio compression - up to 30 channels - in a single PCI slot. The board is specifically designed for advanced CTI applications, such as Voice over IP. The TAP-804 delivers efficient system performance by taking advantage of the fast PCI bus system, yielding greater available bandwidth for CTI applications. Up to twelve, 60-Mhz floating point processors provide as much as 720 MFLOPS of compute power enabling the TAP-804 to support up to 30 channels of voice compression, and that includes G.723.1, G.729a, and SX7300.

The TAP-804 DSP resource board supports two industry standard TDM data buses: the Multi-Vendor Interface Protocol (MVIP) and Signal Computing System Architecture (SCSA). Through the board's TDMRouter, developers are able to port their applications to the TAP-804 to access the entire computer telephony market. Analogic offers a comprehensive SDK including, libraries, sample programs, and debug, load, and other utilities.

MetaVoice

Voxware, Inc.
305 College Road East
Princeton, NJ 08540
www.voxware.com

Voxware is a developer of digital speech and audio technologies for IP telephony, multimedia software, electronic devices and wireless communications. Among its core technologies, Voxware offers MetaVoice, the company's voice-coding technology. Unlike conventional methods which condense the audio waveform to reduce bandwidth, MetaVoice models every element of the human voice, including resonance, pitch, timbre, timing, and character. This allows accurate voice reproduction without distortion, while consuming low levels of bandwidth, storage, and processing power. The MetaVoice codecs were designed specifically for the human voice, allowing for compression ratios of up to 53:1. This level of compression allows voice to transmit more smoothly, with less choppiness and delay in bandwidth-constrained environments.

Another product from VoxWare is VoxPhone, a full-duplex Internet telephone that enables users to make real-time calls over the Internet, as well as initiate full-duplex audio conferences of up to five people without the need for a conference server. VoxPhone is H.323 compliant. and allows users to communicate with any other H.323-compliant phone, including Netscape Conference.

TR2000 Series

Brooktrout Technology, Inc.
410 First Avenue
Needham, MA 02194
www.brooktrout.com

Brooktrout's recent introduction of a complete IP telephony line of hardware and software products includes software development kits, as well as boards, like the TR2000 series, designed to enable vendors to develop standards-based IP voice and fax solutions in a common software and hardware environment. The TR200 series are DSP resource boards designed to support the transport of voice and fax traffic over IP packet data networks. Based on 32-bit DSP chips, the TR2000 provides all of the power needed to support 24 channels of speech or fax processing. A daughterboard with a digital T1 interface lowers system costs and saves system board slots. The TR200 comes with SpeechPac voice compression software which provides users with multiple standard vocoders, supporting G.723.1 and G.729a in particular. These vocoders are part of a highly integrated compression system that includes echo canceler, DTMF and VAD capability, and can be selected at the time of the call.

The TR2000 boards support up to 24 channels of IP voice based on ITU standard vocoders and up to 36 channels based on Lucent's efficient SX-7300 vocoder. Brooktrout boards support both industry standard TDM buses: MVIP and SCbus, which gives users the ability to operate with any digital network interface board that has these interfaces. To speed development, Brooktrout provides the BTGateway Kit, which includes the TR200 IP telephony board with 24 channels of SpeechPac software based on G.723.1, 24 channels of FaxRelay software (for simultaneous fax over the same data network used for voice), BTStackH.323 (H.323 protocol stack), and the Netaccess PRI-ISA24 T1 network interface board.

Fusion 2.0

Natural MicroSystems
8 Erie Drive
Natick, MA 01760
www.nmss.com

Fusion 2.0 from Natural MicroSystems, is the evolution of their Fusion platform. Recent advances include support for the H.323 standard, enabling IP telephony gateways built with Fusion to support calls from a wide range of IP telephony clients including Microsoft NetMeeting, Netscape Conference, VocalTec Internet Phone, VoxWare's VoxPhone, and NetSpeak's WebPhone. Fusion is a hardware and software platform that provides developers of PC-based telecommunications solutions with the scalability, performance, and support for industry standards necessary for building server-based IP telephony applications. Fusion incorporates a broad range of vocoders, including G.723.1, G.729a, Microsoft's GSM, and VoxWare's MetaVoice RT24 algorithm, utilized by both Microsoft and Netscape.

The Fusion IP telephony platform consists of an Alliance Generation (AG) DSP board for PSTN network interface and media processing, an AG Realtime/2 daughtercard for real time vocoding, and a TX Series board which handles IP routing and data protocols. Fusion supports a full T1 in a two ISA slots, and up to four T1s in a single chassis. Furthermore, Fusion supports Natural MicroSystems' TX3000, which enhances scalability by providing up to four T-spans in only 5 ISA slots. Fusion also integrates with CTAccess, Natural MicroSystems' open and extensible development environment, making it simple for developers to quickly bring applications to market.

Array Telecom, Inc.
1120 Finch Avenue West
8th Floor
North York, Ontario M3J 3H7
CANADA
www.telecom.array.ca

ATG is a complete suite of software products that provides comprehensive voice and fax communications over any IP network, including intranets and Internets. The ATG and associated servers enable service providers and corporations to offer high-quality, reliable telecommunications services to their clients. The Array suite provides turnkey inbound, calling card and prepaid services, as well as normal long-distance calling and PC-PBX capabilities. This is accomplished with client/server software on Windows NT servers.

TELEGATE 2.0 is Array's newest generation of its Internet telephony software solution. Enhancements to the gateway software include DSP-based voice and fax processing, using the same codec as in previous versions. Each DSP card will support a maximum of 30 simultaneous voice conversations or fax connections. This represents a 6-fold call volume increase over previous TELEGATE versions. Automatic switching means each channel can be used for fax or voice on a dynamic basis. TELEGATE gateways utilizing DSPs support a variety of interface cards, including T1, E1, ISDN PRI, analog loop, and station interfaces. Version 2.0 provides greater flexibility with new fax capabilities including real-time fax.

Minute Bartering, a TELEGATE service, allows one company's gateway to be configured to allows other organization's gateways to terminate calls into the local dialing area of the first company's gateway for "credits." The Minute Bartering system allows gateway owners to cooperate over the Internet, terminating calls in the local dialing areas of each of the participants at no cost.

Tempest Data/Voice Gateway (DVG)

Franklin Telecom
733 Lakefield Road
Westlake Village, CA 91361
www.ftel.com

Franklin's Tempest DVG supports conventional phone operation. Franklin's Tempest is designed to terminate telephone voice circuits from a telco and carry voice information as data packets over an IP-based data network. Up to 24 channels of voice traffic can be connected to a Tempest gateway, utilizing T1 digital trunks or analog lines via FXO/FXS ports. An Ethernet LAN port provides connection to the data network infrastructure. The voice packets can be routed, bridged, or tunneled through another network to a remote Tempest where the voice traffic is sent to a station, central office, or PBX.

Depending on the number of simultaneous connections expected, the Tempest can be connected to an IP network with an access line as small as 8 Kbps or as large as a 1.5 Mbps T1. Dial into the Tempest from the PSTN or from an office PBX extension. The gateway compresses voice to IP packets in an 8:1 ratio. The Tempest DVG supports industry standard G.729 voice compression and encoding.

eBridge Interactive Web Response (IWR)

eFusion, Inc.
15236 New Greenbrier Parkway
Beaverton, OR 97006
www.efusion.com

The eBridge IWR system supplies call centers with an Internet telephony platform to connect customers directly to live agents, where they can simultaneously talk and share visuals over a single analog phone line. eFusion addresses the lack of QoS standards over public networks with its DirectQuality technology, which enables a voice and data connection over the PSTN to the call center. Call center managers would have the option of either a low-cost connection through the Internet with variable voice quality, or a reliable and secure connection using eFusion's DirectQuality service.

An adjunct open architecture system, the eBridge server integrates with any ACD and supports existing IVR applications. The server also utilizes DSP technology to provide expandability in increments of 24 (T1) or 30 (E1) ports that can expand to multiple server systems with thousands of ports. Hardware features of the server include:

• Two or four 200 MHz Pentium Pro processors per server unit
• G.723-compliant DSP vocoder boards
• 128 MB memory, expandable to 1 GB
  

Software features include:

• Operation on Windows NT 4.0
• Connectivity support for H.323, T.120, SNMP, HTTP, ODBC, TCP/IP, and RTP
• Multimedia scripting through Visual Basic, Visual C++, and Java.

Rockwell Internet Gateway

Rockwell Electronic Commerce Division
300 Bauman Court
Wood Dale, IL 60191
www.ecd.rockwell.com

Rockwell's Internet Gateway bridges the gap between the Internet and the call center by enabling Web users and agents to talk and interact directly over the Internet without the use of a telephone or a second phone line. The gateway server also gives companies an alternate way to route inbound and outbound telephone campaigns over the Internet instead of higher-cost networks. Rockwell's Internet Gateway server is bundled with NetSpeak's WebPhone, a software application allowing Windows-based multimedia PCs to perform standard telephone functions from an on-screen graphical user interface.

All parties using the system need an MCI-compatible sound card and headset. Agent system requirements include a 120 MHz Pentium processor, 16 MB RAM, 3GB hard drive, and LAN connection. Gateway system minimum requirements include a 166 MHz Pentium server, Windows NT 4.0, 32 MB RAM, 3 GB hard drive, network access cards, and Dialogic's Antares 2000/50 high-speed DSP card. Caller system minimum requirements include a 486 66 MHz processor (75 MHz to take advantage of DSP Group's TrueSpeech), Windows 3.x/95/NT, 8 MB RAM, 5 MB free hard drive space, and a 14.4 Kbps modem with error correction.

Cisco 3600 Voice Module

Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
www.cisco.com

The voice module of the Cisco 3600 series routers enables enterprises to integrate voice, fax, and data across existing data infrastructures while reducing costs. With Cisco's voice over IP solution, companies can off-load branch-office traffic from the PSTN and route is over the company's intranet, eliminating toll charges. The voice modules interface with legacy phones, fax machines, key systems, and PBXs, making the network telephony process transparent to users.

Voice processing latency is minimized by Cisco's router-based design and quality of services features in Cisco IOS software, such as Resource Reservation Protocol (RSVP) and Weighted Fair Queuing (WFQ). The dedicated DSP architecture provides processing power for voice compression, echo cancellation, silence suppression, and jitter buffer management without burdening the router. The voice modules support toll-quality compression scheme such as G.711 for high-bit rate applications and G.729 for WAN applications. WAN modules support ISDN (BRI and PRI), integrated digital modems, and T1/E1. LAN modules support Ethernet, Fast Ethernet, and Token Ring. The framework within the IOS software also offers support for H.323 for interoperability with third-party telephony software. The vocoder can compress standard 64 K voice down to 8 K. List prices for the voice modules start at $625 per port.

WebPhone Gateway Exchange (WGX) Server

NetSpeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487
www.netspeak.com

The NetSpeak's WGX server brings calls placed from conventional telephones into an IP-based data network, and calls placed using a WebPhone application into an existing telephone system. The WebPhone application provides toll-quality, point-to-point communication over TCP/IP networks. In order for WebPhone users to communicate with someone using a traditional phone on the PSTN, the WGX server is needed.

The WGX server consists of hardware and software that run on the Windows NT OS. Two industry standard interfaces are supported - conventional circuit-switched telephone systems and data network systems. The server takes audio and signaling information from the PSTN, compresses or decompresses it, and delivers it to the TCP/IP network. Two interfaces are supported to the PSTN: analog, standard two-wire loop connecting directly to a line from a telco or a line to an existing PBX; and a T1, high speed, digital interface for connecting directly to a telco central office, PBX, or ACD. Analog and digital boards from Natural MicroSystems are used by NetSpeak. NetSpeak also uses the RSA Public Key Cryptosystem to provide secure communications.

VocalTec Telephony Gateway 3.1
VocalTec Communications, Inc.
35 Industrial Parkway
Northvale, NJ 07647
www.vocaltec.com

Building on VocalTec's Internet Phone technology, the Telephony Gateway bridges the gap between the PSTN and the Internet or intranets to enable long-distance calling from phone-to-phone, fax-to-fax, PC-to-phone, phone-to-PC, and Web browser-to-phone. The VocalTec Telephony Gateway uses full-duplex echo-canceling algorithms and a management system which optimizes audio behavior over different types of networks, and the gateway reconstructs packets which may have been delayed or lost during transmission.

Other features include:

• Simple Dialing Plan: The Gateway auto attendant asks the caller for the destination phone number, and the caller enters the number just as they ordinarily would. The local Gateway automatically determines which remote Gateway should handle the call.
• Call Progress Detection: An auto detection mechanism prevents disconnects during long periods of silence, ensuring conversations are not prematurely cut off.
• Interactive Voice Response: IVR serves as the interface between the PSTN/PBX and the Internet or intranet. Voice prompts make the calling process user-friendly, and the system can be customized so prompts can be recorded in any language.
• T1/E1 capability.

The VocalTec Telephony Gateway is designed for easy management. An intuitive graphical user interface GUI), security functions, and third-party billing support are standard. The GUI allows viewing of group connectivity, and each line can be configured separately as an internal extension, analog line, or digital trunk line with information for up to 30 voice and 8 fax lines.

Internet Telephony Server

Lucent Technologies, Inc.
211 Mt. Airy Road
Basking Ridge, NJ 07920
www.lucent.com

Lucent's Internet Telephony Server (ITS) is a server-based solution that places voice and fax calls over TCP/IP networks using voice compression software developed by Bell Labs' new division, elemedia. ITS supports industry standard codecs, including those provided as part of the H.323 protocol stack. ITS works with DEFINITY ECS and most existing phone systems and is connected to the PBX via a T1/E1/PRI tie line interface. The ITS is based on an industry-standard components, including:

• Intel-based Pentium server
• Windows NT 4.0 OS
• Administration from NT screens or web browser
• Standard NICs: Ethernet 10Mb/s and 100Mb/s
• Standard DSP cards
• Standard T1/E1/PRI or analog loop-start interface
• H.323 protocol

ITS connects a company's switches or PBXs together and routes calls over the Internet or an intranet, or the customer private or PSTN networks, depending on customer pre-configured criteria. Up to 24 sessions can take place simultaneously, each including either a single voice or fax call, per each T1 card in the ITS. The ITS uses a dynamic and transparent routing algorithm for its operation (no routing decisions on the part of the customers are needed). In addition, if the quality of the IP network falls below pre-specified threshold, backup to PSTN is automatic.

Clarent Gateway 2.0

Clarent Corporation
1900 Broadway
Suite 200
Redwood City, CA 94063
www.clarent.com

The Clarent Gateway 2.0 is a suite of software programs that converts a telephone conversations and fax into digital information and compresses it 3:1 for transfer over the Internet or TCP/IP intranets. This version brings users improved telephony connections, enhanced scalability, and expanded management features that making billing and reporting easier. Applications developed in conjunction with Voxware, a leader in voice compression, squeeze more calls into less bandwidth - from 4-24 calls per gateway.

Clarent's gateway is powered by single or dual 200 MHz Pentium Pro processors, runs on Windows NT 4.0, and requires 64 MB RAM. The telephone interface cards are Natural MicroSystems', and Clarent supports analog, T1, E1, and ISDN connections. New features offered by version 2.0 include real-time fax, more sophisticated management capabilities (such as RSA encryption of customer information), redundant architecture for fault tolerance, Simple Network Management Protocol (SNMP) support, and support for SS7 and HP OpenView.

iGate

TTM & Associates, Inc.
271 East Imperial Highway
Suite 641
Fullerton, CA 92635
www.ttminc.com

TTM's iGate is a combination Internet gateway, voice service bureau, and unified messaging solution that uses the Internet to deliver telephone conversations in real-time will toll-quality voice. The iGate server is accessed from any touch-tone phone. TTM addresses Internet toll-bypass as a business application. Online prepaid calling cards and credit card payments are integrated into the server.

Two services are incorporated into the iGate server: iCall and iOffice. iCall features are services normally associated with a voice mail system, such as "follow me" service, voice mail, and pager notification. iOffice uses an automated attendant, multiple extensions, individual voice announcements, and "fax through" integration. Both services can use debit and credit card purchasing through a Web interface and via telephone.

iGate servers are available in configuration from 8 to 100 ports per server. Multiple servers can be combined to create any size system. TTM is a development partner with Natural MicroSystems, and uses Fusion in creating its servers. TTM will modify the iGate servers for custom gateway applications on a project by project basis.

LinkNet IP Telephony Gateway

Linkon Corporation
140 Sherman Street
Fairfield, CT 06430
www.linkon.com

The LinkNet IP Telephony Gateway is designed as an open-system platform with a full set of enabling technologies supporting a variety of IP communication applications, including:

• Phone to phone,
• PC to phone,
• Phone to PC,
• Real-time and Store-and-Forward IP Fax,
• Voice over Frame Relay, ATM, and Cable,
• SS7 integration.

At the heart of the hardware architecture, developers will find the foundation-switching card, a carrier card that interfaces with the host bus, providing connectivity to internal computer telephony bus structures (MVIP, SCSA). Current cards are available for ISA bus and Sbus, with CompactPCI and VME implementations planned for the coming year. Most recently, Linkon has announced a gateway for Sun Microsystems' PCI-based Netra T 1100 and Sun Enterprise 450 computing platforms. The new card is designed to run under the Solaris 2.6 operating system, provides data transfer rate of up to 8 Mbps, and support for up to 48 channels per voice slot.

Based on Linkon's universal port Maestro System technology, the gateway also offers H.323 support, which comprises data format, call control, and both audio and video compression. At present, the LinkNet IP Telephony Gateway supports from 4 analog ports up to multiple E1/T1 digital circuits, thus providing up to 90/96 channels for full-duplex digital communications. According to the company, the product is designed to address the most difficult technical issues of IP telephony, namely full-duplex communications, scalability, echo cancellation, efficient audio compression, and low latency.

Hicom Xpress Telephony Internet Server

Siemens Business Communication Systems, Inc.
4900 Old Ironsides Drive
Santa Clara, CA 95054
www.siemenscom.com

Siemens Business Communication Systems' Hicom Xpress Telephony Internet Server (TIS), enables end users to place voice and real-time fax calls over a corporate intranet or the Internet. Siemens' TIS, incorporating components from Vienna Systems Corporation, allows voice and real-time fax calls initiated from a PBX to be routed over an IP network. Siemens also plans to provide phone-to-phone, real-time fax-to-fax, PC-to-phone, phone-to-PC, and PC-to-PC communications over the Intranet/Internet infrastructure.

Transparent access to the IP trunks of Siemens' new Hicom Xpress TIS is made possible by the automatic route selection software on Siemens' Hicom 300 E communications server. The Siemens Hicom Xpress TIS is fully non-blocking and can handle up to 46 calls simultaneously. It delivers toll-quality voice with an end-to-end delay of only 100 msec and a greatly reduced bandwidth CODEC (coder/decoder) using only 7.3 Kbps per voice channel. The product supports standard T1/PRI interfaces to a PBX and most of the popular LAN interfaces to the IP network. E1 is anticipated for delivery in early 1998. Delivering real-time fax service on any port on a call-by-call basis, the Universal Voice/Fax Line feature of the TIS allows any line to start out as a voice call and switch to fax. PSTN fallback, the ability to reroute calls back through the PBX transparently, is supported when the IP network is congested at the time of call setup.

Access Plus F200ip

Nuera Communications
10445 Pacific Center Court
San Diego, CA 92121
www.nuera.com

Supporting phone to phone connections over frame relay, Internet, and private IP networks, Nuera's Access Plus F200ip integrated Voice over IP gateway and Voice/Frame Relay Access Device (FRAD) features E-CELP (Enhanced Code Excited Linear Prediction) along with Ethernet-based IP transmission and serial frame relay trunks supporting T1/E1 throughput. The F200ip offers software configurable analog voice interfaces, digital T1/E1 PBX interfaces, and capacity of up to 30 voice and fax channels per unit for users. Nuera's F200ip routes calls based on either user selection or system manager configuration.

As for scalability, Ethernet support allows F200ip units to be scaled up to configurations supporting from tens to hundreds of T1/E1 connections and thousands of voice channels as packet voice applications grow. The product includes an embedded SNMP (Simple Network Management Protocol) agent that can be managed under any SNMP-based manager as well as Nuera's HP OpenView for Windows workstation manager. An F-Series configurator is designed to facilitate installation and configuration of the various Nuera F-Series products in public and private network applications.

V/IP Phone/Fax IP Gateway

MICOM Communications, Inc.
4100 Los Angeles Avenue
Simi Valley, CA 93063
www.micom.com

MICOM'S V/IP (Voice over IP) product family allows companies to digitize, compress, and route intra-company voice and fax traffic over their enterprise IP networks. V/IP connects to PBX and key telephone systems via a comprehensive line of analog (E&M, FXS, FXO) and digital (T1, E1) voice interface cards (VICs), expandable from 4 to 24 channels for T1 and to 30 channels for E1. V/IP installs in any shared 486, Pentium, or similar PC with NetWare, MS-DOS, Windows 95, or Windows NT. MICOM's offering uses very little processing power, allowing the PC to be used for other tasks.

V/IP uses MICOM's ClearVoice technology with ITU standard G.729 voice compression, silence suppression, background noise regeneration, echo cancellation, and robust voice switching. ClearVoice runs on the VIC's 40-MIPS digital signal processors (DSPs), providing toll-quality voice with low bandwidth requirements. In fact, V/IP connections average less than 3 percent of a 56/64K WAN link. V/IP converts a voice call into a digital signal at 8 Kbps. IP overhead increases this to 17 Kbps. MICOM's silence suppression technology (which only uses bandwidth when a person is talking) then reduces the voice signal to an average of 7 Kbps. As for flexibility, V/IP Gateways transmit over any LAN/WAN via Ethernet, FDDI, frame relay, or ATM.

InfoTalk; InfoGate

InnoMedia, Inc.
4800 Great America Parkway, Suite 400
Santa Clara, CA 95054
www.innomedia.com

InnoMedia's InfoTalk Internet telephony device reroutes standard long-distance calls over the Internet without requiring a PC. Designed for an individual user and based on InnoMedia's patent-pending compression technology the product plugs in between the handset and the wall connection. After establishing a normal phone connection, all the InfoTalk user needs to decide if they want to pay for the long-distance call or use the Internet instead.

InnoMedia has also recently announced InfoGate EP and InfoGate IP - a pair of Internet gateways for voice and fax applications, designed for small- to medium-sized corporations. Each gateway is an internal PC card that allows up to four simultaneous users to call long-distance over the Internet to any destination using a standard phone. The gateways offer the following features:

• Four PSTN lines (EP version) for local call-in and dial-out capabilities;
• Four internal extensions (IP version) allowing business users to realize cost saving features;
• Four-card capacity per system, allowing up to 16 ports in a single PC system;
• Backup ISDN connection ensures high-quality, low-cost voice connection during peak Internet traffic periods.

InfoGate offers dynamic least-cost routing, full-duplex voice echo cancellation, interactive voice response, and is compliant with the ITU's H.323 standard.

Vienna.way Gateway

Vienna Systems Corporation
555 Legget Drive, Suite 400
Kanata, Ontario
Canada K2K 2X3
www.viennasys.com

The Vienna.way Gateway allows users to place voice and fax calls over the Internet, private IP networks, or the PSTN using conventional phones and fax machines. Key features of the product include echo cancellation, compression, and transport optimized for IP networks; PBX functionality at the desktop, such as hold, transfer, forward, and conference; real-time fax transmission and immediate confirmation; scalable system that can handle up to 2,000 simultaneous calls and 800 trunks; and centralized network administration.

The Vienna.way Gateway consists of Network Interface Cards (NICs) and Digital Signal Processor (DSP) cards installed in either a standard PCI/ISA tower or a resilient rackmount chassis for high availability applications. The Gateway interfaces directly to telephone networks or PBXs through either ISDN BRI or T1 interfaces. Programmable DSP cards ensure that voice processing and packetization occur directly on the card, leaving Gateway resources available for call processing functions such as billing, routing, and call authorization.

Vienna's my.way client application turns a multimedia PC with microphone and sound card into a multi-line phone with PBX functionality. Users can simultaneously access their corporate voice and data networks, attend virtual meetings, and collaborate on documents with their coworkers through a 28.8 Kbps or faster dial-up connection from anywhere in the world. Coworkers at the main office can reach remote workers by simply dialing an extension.

Vocal'Net IP Telephony Gateway

Inter-Tel
120 North 44th Street, Suite 200
Phoenix, AZ 85034
www.inter-tel.com

Using Internet Protocol (IP) Inter-Tel's Vocal'Net IP Telephony Gateway serves as a bridge between the PSTN or PABX and an IP data network. It enables organizations to communicate full duplex over IP networks such as the Internet or a private intranet for potential cost savings compared to standard long-distance phone service. The Gateway converts voice to data which consumes less bandwidth. The Vocal'Net Telephony Gateway is scalable from 8 to hundreds of ports depending on the needs of small to large businesses. Packetization and real-time compression take place on high-performance DSP cards, with no significant impact on the CPU. Minimum platform requirements for the Vocal'Net include Microsoft Windows NT operating system, a Pentium 100 processor, 10 MB of disk space, and 16 MB of RAM. The system works with any manufacturer's PBX as well as any existing computer network.

The current version of Vocal'Net allows local or remote configuration over the IP data network for programming multiple locations worldwide, and also offers a call accounting package (InsideTrack), which provides the ability to define and run custom reports as well as several predefined reports. Future enhancements to Inter-Tel's telephony gateway include:

• Increased capacity of up to 96 ports per server;
• Support for H.323, G.723.1, G.729a, Voxware, and elemedia;
• Fax gateway, fax broadcast, and video gateway.

Network Exchange Vodex Voice Gateway

NETRIX Corporation
13595 Dulles Technology Drive
Herndon, VA 20171
www.netrix.com

NETRIX has announced Vodex Voice Gateway software for its Network Exchange 2210, a Voice/Data/Fax over IP Gateway Switch. Vodex simultaneously delivers high-quality voice over IP and voice over frame relay with the ability to gateway between the two. The Network Exchange 2210 is a multi-service switching platform that combines WAN switching, switched compressed voice, and multi-protocol support in a single platform. The Network Exchange connects between the corporate PBX using either a DS1 or E1 interface and the LAN using a standard Ethernet interface. The existing LAN carries voice/data/fax traffic between the PBX and the egress to the WAN. Along the way, voice traffic is compressed down to 5.5 Kbps from 64 Kbps. Each Network Exchange has one IP address per box, greatly reducing network administration. Multiple units can be deployed within the enterprise to handle PBXs, analog handsets, and even ISDN-based PBXs.

The functionality of the Vodex gateway is transparent to network users, helping to ensure its use and realization of the associated cost-savings. The Network Exchange can adapt to any dialing plan and reduce administrative costs typically associated with an off-premise extension, a tie line or special inter-location numbering plans. The Network Exchange can operate with any WAN technology, such as public Frame Relay or ATM services, leased lines, satellites, and IP Virtual Private Network (VPN) services. The extensive statistics-gathering capabilities of the Gateway provide real-time accounting information that is useful for management and billing.

Internet Phone Fax Gateway

MicroCall (US office)
160 East 56th Street
New York, NY 10022
www.micro-call.com

MicroCall's Internet Phone Fax Gateway (IPFG) is designed to deliver real-time voice and fax communications over IP networks. Developed by IBM-Haifa Research Lab, the IPFG is based on MicroCall's hardware and software. The Internet Phone Fax Gateway utilizes industry standard protocols: TCP, RTP on UDP, H.323, and G.711 to provide excellent voice quality, minimum delays, and instant connectivity at greatly reduced cost. The IPFG is available in 8- and 16-port analog configurations, as well as T1 and E1 configurations (24/30 ports, respectively). Other PSTN interfaces include 2-wire loop start, 2-wire E&M, and ISDN PRI/BRI.

The gateway's billing management functions include a full range of features such as: number of calling subscriber; number of called subscriber; length of call; and time and date. An RS232 interface is also available to interface with external billing systems. Special features of the gateway include:

• Built-in IVR system;
• Individual voice mailbox;
• PABX-like features, including call forward, hold, and conference;
• Unified messaging.

Network management features include local and remote management of the gateway, as well as a network interface via SNMP.

Fax Gateways

Panasonic Office Products Company
Two Panasonic Way
Secaucus, NJ 07094
www.panasonic.com/office

Panasonic's Panafax UF-770i is designed to reduce phone charges for organizations that send frequent global and/or long-distance faxes. The UF 770I is a hardware-based solution that sends and receives documents, pictures, hand-written messages, and e-mail over the Internet by simply pressing a one-touch key. Instead of dialing a telephone number, the user enters an e-mail address on the keypad. The UF-770i works by first scanning a document and compressing the binary image. It wraps the compressed image with TIFF headers and tags and then encodes the TIFF file in base64. The data is then attached to a MIME file (multipurpose Internet mail extension) and passed through a predetermined mail server with SMTP (simple mail transfer protocol).

The Panafax UF-770i is also a standard G3-compatible fax device, and can be configured with a variety of memory and paper cassette options for practically every office environment. The UF-770i can track and monitor department by department fax usage - allowing up to 24 departmental codes to be used. Photos and text can be reproduced on the same page with clarity and sharpness due to Panasonic's Facsimile Image Processor chip, which features auto Picture/Text Recognition, Edge Enhancement, and a 64-level gray scale.

FAXfree Portal 500

TAC Systems
1035 Putman Drive
Huntsville, AL 35816
www.tacsys.com

TAC Systems' FAXfree Portal allows legacy fax equipment to be connected to the Internet and practically freed from long-distance charges. The FAXfree Portal 500 joins a growing list of TAC products that enable Internet faxing from a single PC, a LAN-based PC, or a standalone fax machine. Connecting the FAXfree Portal is as simple as connecting an answering machine - the product comes with a built-in Ethernet port for direct connection to a LAN, as well as a dial-up PPP connection for connecting to an ISP.

The basic operation of the fax machine remains the same. Upon entering the destination phone number (or e-mail address), the FAXfree Portal 500 scans a list of up to 500 previously entered phone numbers and e-mail addresses for a match. If a match is made, the fax goes out over the Internet to a matching e-mail address, without the associated long-distance charges. A front-panel LCD display monitors the progress of the call, and lets the user know if the call will appear on the next monthly phone bill or if it went out over the Internet.

PASSaFAX

RADLinx
900 Corporate Drive
Mahwah, NJ 07430
www.radlinx.com

RADLinx' PASSaFAX PF-1M fax gateway transfers faxes between the Public Switched Telephone Network (PSTN) and IP backbones (Internet/intranet). The gateway connects a standard facsimile device to a company's LAN. Users are then able to send faxes over their intranet or over the Intranet, to save on toll costs, instead of using the PSTN. Fax machines can either be attached locally to the PF-1M or attached remotely over the PSTN. Thus, both "on Net" and "off Net" fax machines are able to enjoy the benefits of toll free faxing via the Internet.

Most recently, RADLinx has announced the release of PASSaFAX Fax and e-mail bidirectional integration tools designed to provide fax to e-mail delivery, e-mail to fax delivery, and store-and-forward fax delivery as a fallback option to RADLinx' OND real-time Internet fax transfer. The Fax and e-mail integration follow such standard protocols as SMTP and POP3, so that the fax user can send or receive faxes from any e-mail user over the Internet. Additional features include the ability to broadcast faxes to several destinations simultaneously.

Alcom IP Fax Service

Alcom Corporation
1616 N. Shoreline Boulevard
Mountain View, CA 94043
www.alcom.com

Alcom IP Fax Service utilizes Alcom technology that acts as a gateway between fax hardware and a Web browser. With Alcom's IntraFax and LanFax NT server acting as a hub for all incoming and outgoing faxes, all faxes pass through the Alcom location on their way to their intended destination. The main features found in Alcom's IP Fax service are:

• Send/Receive faxes from anywhere in the world, from any computer, with any Web browser with no extra software requirements.
• E-mail to fax capability allows you to send faxes without logging on to a Web account (S/MIME support).
• Secure login connection to your account: full security and confidentiality.
• No need for expensive fax equipment or additional phone lines.
• Broadcast fax capability.
• Compose new faxes on-the-fly; use built-in spell-checker; attach MS Word or other documents.

Optional other features include e-mail notification of incoming faxes and failed outgoing faxes at no extra charge, delivery of faxes as e-mail attachments at no extra charge, numeric/alphanumeric pager notification, and personal 800 fax number.

w.FAX Web Fax Server

SoftTek, Inc.
30555 Trabuco Canyon Road
Suite 100
Trabuco Canyon, CA 92678
www.sftek.com

SoftTek's w.FAX Web Fax Server product is the company's entry into the Enhanced Fax Services market. To support this new service, SoftTek is planning to roll out POPs (points of presence) into major international markets, initially to include Japan, Holland, Germany, and the United States. The primary need for these enhanced fax services is from international customers who send large volumes of faxes to the United States. The enhanced services are built on SoftTek's core technology of fax server products to offer low-cost faxing and convenience of access by users that want to use their fax machine, a browser (Netscape Navigator), or any SMTP e-mail package.

By using the Netscape Navigator browser as the user interface, SoftTek's w.FAX is supported by 16 different client platforms that Netscape supports. On the server side, w.FAX is built and supported on commercial-strength Sun platforms which provide the ability to scale from small systems to a distributed network of Intranet fax servers. Users can attach PostScript, text, or popular PC file formats to the cover page of their fax to simplify transmission of information in its original format. Additionally, users are notified of an incoming fax through the Netscape Mail feature provided with the browser. An optional OCR package deciphers the "To:" line on the fax cover page and auto-routes the fax to a user's mailbox.

Video Conferencing Gateways

VideoServer, Inc.
63 Third Avenue
Burlington, MA 01803
www.videoserver.com

VideoServer's Encounter NetGate enables audio-visual and multimedia conferencing between ITU standards-based H.323 and H.320 terminals. The NetGate emulates the communications interfaces of H.323 and H.320 terminals, which lets each terminal communicate with the NetGate server as if it were another terminal of the same type, hiding the complexity of the gateway function. Encounter NetGate also supports G.711, G.722, G.728, and G.723 audio.

Encounter NetGate supports H.261 and H.263 video. Selection of the video protocol is dynamic and depends on the capabilities of the terminals in a gateway session. T.120 data conferencing is also supported to allow for maximum collaboration. The switched circuit interfaces available for this module include quad-port BRI, dual port T1/PRI, and dual port E1/PRI. Encounter NetGate connects to an IP network via 10/100 BaseT interface modules.

The system is available in two chassis configurations: the Workgroup and the Enterprise chassis. The Workgroup chassis can support 4 or 8 gateway sessions and the Enterprise chassis supports 8 sessions, scalable to 32.

OnLAN L2W-323

RADVision, Inc.
575 Corporate Drive
Suite 420
Mahwah, NJ 07430
www.radvision.com

The L2W-323 is a self-contained, standalone gateway the translate between H.323 and H.320 protocols, and converts multimedia information from circuit switch to H.323 IP packets. On the PSTN side, it supports voice in addition to H.320 video conferencing sessions. Depending on your WAN, the L2W-323 lets users exchange audio, video, and data in real-time at 64 Kbps, 128 Kbps, 256 Kbps, or 384 Kbps. Voice over IP gateway features include IVR, echo cancellation, and DTMF support. The gateway also supports up to 8 concurrent voice calls or 4 concurrent video calls between users on different networks. Full end-to-end T.120 support is also provided to users of a L2W-323-enabled network, provided all terminals support the standard.

In conjunction with an embedded RADVision H.323 Gatekeeper, the L2W-323 provides the functionality of a multimedia PBX - allowing internetwork calls, direct inward dialing, call forward and transfer, and custom call control. By-pass routing between network segments is provided. Optional modules perform transcoding between G.723 to G.711, and G.728 to G.711. A RS232 serial/modem port is standard for remote configuration.