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June 1999

Carrier Class Internet Telephony Gateways

Much has been said about what it takes to deserve the moniker "Carrier Class." I think everyone will agree that, in the end, it all boils down to 2 things: Scalability and Reliability. The products featured on the following pages offer both of those attributes. You want scale? Try Lucent’s 28 T1-span PacketStar Internet Telephony Server. Or what about Transmedia’s 100,000-port capacity? Well, not every one of these products has that kind of scale, but we figured we’d list those that offered at least a single T1/E1 capacity. We’ve also included the latest development platforms from Dialogic and Natural MicroSystems for those of you interested in building your own gateways.

The interesting thing is that when considering these vendors for this roundup, it struck us that over the course of about a year and a half, since the premier of Internet Telephony, we’ve gone from talking about 4-port black box PC-to-PC gateways to mammoth Carrier Grade solutions. Amazing. Well, now that we’ve seen the advent of these packet-switching behemoths, it’s time for vendors to ramp up the development of applications and services to run on them. (They’re already doing that, just check out the Services section) One final note: This industry moves at lightning speed. We’re all witnesses to that fact. So use this list as a starting point when checking out what the vendors have to offer, and keep your eye glued to TMCnet.com’s scrolling news marquee at www.tmcnet.com. You never know when the next Carrier Class gateway will roll off the assembly lines.

-Greg Galitzine

PacketStar Internet Telephony Server 3.0
Lucent Technologies, Inc.
600 Mountain Ave.
Murray Hill, NJ 07974

Lucent’s PacketStar server merges data and voice as part of the PacketStar Internet Telephony System. Geared toward carriers, ISPs, CLECs, and cable operators, the PacketStar provides capacity of up to 28 T1s and 21 E1s. The system comes with a gateway, a service access manager, and an administration manager, and the latter two may reside on the gateway or independently. The gateway enables phone, fax, and PC-to-phone voice transmission through a data network.

The PacketStar utilizes multiple voice compression through Lucent’s elemedia software, as well as a customizable API. The gateway supports single and two-stage dialing, as well as integrated routing and billing functions. It comes with a 24/30 channel configuration, and offers high scalability in a desktop or rackmount configuration. The PacketStar runs on Unixware, and supports IP, TCP, and RTP network layer protocols. For more information, visit Lucent’s Web site at www.lucent.com.

Vanguard 8500 Series Multiservice Concentrators (MSC)
Motorola, Inc
20 Cabot Blvd.
Mansfield, MA 02048

The 8500 MSC series from Motorola adds high-speed connectivity to the Vanguard product family by allowing multiple T1/E1 and T3/E3 interfaces. The products can support transfer of data, voice, and images through ATM, frame relay, TDM, X.25, ISDN, and cell-based connections. The MSC features a modular architecture so that processing capacity may be added as needed. Port expansion is also convenient, and features hot-swappable cards and scalable connection modules that will support up to 1,000 ports.

Other features of the concentrators include a 1.2 Gbps switch fabric for maximum throughput, high-quality integral packet voice switching, and an integral PRI interface. The series offers real-time performance results, and forwards events if user-defined statistics and thresholds are exceeded. The 8550 model is marketed for large sites or those needing expansion, and can scale up to 16 processors. The 8510 can handle three processors, and is suited for regional branch concentrations with multiple T1/E1 access. For more information, visit Motorola’s Web site at www.motorola.com/networking.

TrunkPack VoIP 200/400
AudioCodes, Ltd.
4 Hahoresh Rd.
Yehud, Israel 56470

The TrunkPack VoIP communication boards from AudioCodes are available in 60 or 120 ports, in PCI or CompactPCI format respectively. The 200 model uses the RTP/RTCP media streaming protocol to interface between an on-board NIC and an IP network. A dual E1/T1 interface may be added for connection to the PSTN, as well as on-board signaling support. The board utilizes 12 DSP chips from AudioCodes to perform all voice band functions, including voice compression through G.723.1, G.729A, and the proprietary NetCoder codec.

The 400 model is hot swappable, and supports streaming voice and fax over IP. Packet-streaming functions are handled by two dedicated processors, which function through an on-board 100BT LAN switch. Features include adaptive jitter buffer management, and T.38 fax relay support. For applications that require separation of management traffic, voice traffic, and gateway control for security reasons, two separate 100Base-T interfaces may be implemented. For additional information, visit the AudioCodes Web site at www.audiocodes.com.

TransMedia Communications, Inc.
20 Great Oaks Blvd., Ste. 210
San Jose, CA 95119

TransMedia’s MMS-1600 carrier switch merges data with traditional voice at the carrier network edge for low-cost voice/data/video offerings. The switching platform can integrate PSTN, IP, and ATM for transport speeds of DS0 up to OC-48. It can also enable media adaptation and signaling across multiple networks, while providing the characteristics of traffic and the network that particular service carriers want to offer. The switch should be located at a point of presence or the carrier’s central office.

The MMS-1600 uses TransMedia’s MultiPath technology to provide network, media, and speed functionality, as well as media adaptation, switching, and quality of service (QoS) traffic management. The high-density, programmable design allows carriers to offer full line-rate voice compression, as well as voice VPN services across packet networks using ISDN PRI and CAS. An additional feature is a high-density optical broadband service card for dynamic bandwidth grooming on a per call basis. Multipath utilizes an efficient packet-processing function of microcoding to pipeline packets through the switch, as well as dynamic de-jitter buffer, for toll-quality voice with less than 50 ms of latency. The switch is scalable up to 100,000 ports, and provides bandwidth from five to 10 Gbps. For more information, visit TransMedia’s Web site at www.trsmedia.com.

VocalTec Telephony Gateway Series 2000
VocalTec Communications, Ltd.
35 Industrial Pkwy.
Northvale, NJ 07647

VocalTec’s DSP-based Gateway Series 2000 offers a built-in networking switch and router for processing as many as 1440/1152 calls (E1/T1). Packaged as a high-capacity, carrier-class solution, the gateway is part of the multiservice VocalTec Ensemble Architecture platform of voice, video, and data convergence. It supports up to 16 E1/T1 trunks, and features 480/384 ports per shelf, as well as three terminals per cabinet.

Features include advanced voice packet handling with jitter buffer, bad-frame interpolation, and an adaptive echo canceler. The Series 2000 offers easy management through a GUI, which can be launched on a single console through the VocalTec Network Manager. Call detail records, accounting, and billing information are also managed through the product’s gatekeeper intelligence. The gateway is H.323 Version 2-compliant, and features industry-standard codecs for interfacing with third-party systems. The product will ship in Q4. For more information, visit VocalTec’s Web site at www.vocaltec.com.

CallLogiX Voice/Fax Gateway
Infosoft Technologies
26 Avalon Dr.
Montville, NJ 07045

The CallLogiX Gateway from Infosoft offers H.323-compliant transmission of voice and fax, aimed at carriers, service providers, and the enterprise market. The gateway supports analog and T1/E1 interfaces, as well as industry-standard signaling protocols, and is scalable up to 150 ports within a single rackmount chassis. The CallLogiX monitors QoS and adjusts call control, and will also route calls from an inbound port to either a predetermined outbound port, or based on the dialed number.

The gateway features a simple GUI configuration, as well as remote management through SNMP. Alarms are monitored using the Web-enabled CallLogiX Gateway Monitor, which receives alarm events from the digital interface via an SNMP trap. The gateway also features high-density DM3/IP Link voice boards from Dialogic Corporation. For more information, visit Infosoft’s Web site at www.infsft.com.

Clarent Carrier Gateway
Clarent Corporation
700 Chesapeake Dr.
Redwood City, CA 94063

A combination of DSP and host-based voice processing comprises Clarent’s Carrier Gateway, for transmitting toll-quality voice and fax calls over the Internet or a TCP/IP intranet. The gateway uses industry-standard echo cancellation, and interacts with the billing, call routing, and network administration features found in the Clarent Command Center package. The H.323-compliant gateway supports standard codecs, in addition to Q.931 signaling.

The Clarent Carrier Gateway features eight- to 120-port scalability, and gateways may be stacked for desired density. The product supports transmission via T1/E1, analog, PBX, SS7/C7, and PRI-ISDN connections. Network management is performed via SNMP, MIB-II, and a LanManager, and the gateway also comes with a service editor and an IVR toolbox. Other features include a single Pentium II processor running at 350 to 400 MHz with 128 MB of RAM, and a 10/100 Base-T network interface card for Ethernet connectivity. For more information, visit Clarent’s Web site at www.clarent.com.

elemedia H.323 Gateway Server Platform
elemedia (a Lucent Technologies company)
101 Crawfords Corner Rd., Rm. 1C412
Holmdel, NJ 07733

The H.323 Gateway Server Platform software from elemedia runs on DSP boards with an API to the host processor to keep the software independent of the DSP board, thus providing a simple, modular architecture. The gateway supports PSTN to IP protocol mapping, real-time fax demod/remod, and offers an H.323 protocol stack. It supports a variety of speech compression formats, and offers optimal use for bursty traffic from PC-to-phone or phone-to-phone traffic modes over the Internet or managed IP networks.

The platform also comes with an optional host-based Pentium-only version of the software, for users who do not require high-density solutions — eliminating the need for expensive DSP boards. The application API is the same, whether the host-based or DSP-based implementation is used. Fax features include the ability to automatically recognize fax traffic, and switch between speech compression and fax handling for dynamic sharing of DSP resources. The gateway platform supports Group 3 facsimile protocols. The software is currently available as executable code for the Brooktrout TR2001 and Analogic TAP800 DSP boards. For more information, visit elemedia’s Web site at www.elemedia.com.

InterXpress 2000 IWU
Siemens Telecom Networks
900 Broken Sound Pkwy.
Boca Raton, FL 33487

The InterXpress 2000 Interworking Unit (IWU) from Siemens is a stand-alone or central office integrated device that allows voice and fax transport between the PSTN and IP-based networks. The gateway enables PC-to-PC, phone-to-phone, PC-to-phone, phone-to-PC, IP-to-PSTN fax, and PSTN fax-to-IP transmissions. The InterXpress can also store and play voice prompts and international dialing plans, as well as process modem calls for remote access services. It is designed for remote and central offices that need high-density deployment of Internet telephony for wide areas.

The solution is modular and scalable to 312 ports for T1 systems, and 390 ports for E1 systems. It also supports a variety of networks including CATV, IP, PSTN, and the open Internet. The InterXpress provides several billing and customer care features, including authentication/authorization, and billing. Relevant information is logged through the SNMP network management system, and remote software upgrades are supported with minimal service interruption. For more information, visit Siemens’ Web site at www.icn.siemens.com.

Inter-Tel Vocal’Net 3200S
Inter-Tel, Inc.
7300 West Boston St.
Chandler, Arizona 85226-3224

Inter-Tel’s complete gateway solution integrates management, network access, and support capabilities. The Vocal’Net 3200S enables full-duplex voice and real-time fax through open, standards-based software and DSP voice processing hardware, on the Windows NT platform. The gateway supports T1/E1, analog sign line, and ISDN interfaces, and integrates with PBXs, central office equipment, and other circuit-switched telecommunications platforms.

The product is available in 8-port analog or 24-port digital configurations, and both setups are scalable up to hundreds of ports. It supports open database control and the Microsoft MS-GSM vocoder compression standard, and is H.323-compliant. Features include caller ID, automatic database updates, call progress analysis and detection, network status polling, and fax capability. The gateway also offers jitter buffer management, centralized administration, dynamic routing algorithms, and detailed report generation. For more information, visit Inter-Tel’s Web site at www.inter-tel.com.

Nuvo 500
Mockingbird Networks
10050 Bubb Rd.
Cupertino, CA 95014

Mockingbird offers a "hybrid" IP telephony switch architecture with the Nuvo 500, which integrates with the PSTN switch fabric and interfaces with SS7 call control as well as IP and data networks. Instead of sitting on the edge of the network, this Class 4 tandem switch sits behind the PSTN Class 5 switch for optimum integration with the existing network. The Nuvo provides all of the features of an edge gateway, including real-time H.323-compliant transmission of voice, fax, and data through T1/E1 and IP interfaces.

Nuvo 500 features include industry-standard voice compression, silence suppression, and echo cancellation. The Solaris-based platform features an UltraSPARC CPU with 20 PCI slots for easy scalability from 48 to 576 ports per shelf. Additional gateways may be stacked within a rack for up to 10,000 simultaneous calls. For more information, visit Mockingbird’s Web site at www.mockingbirdnetworks.com.

Salix Technologies, Inc.
904 Wind River Lane, Ste. 101
Gaithersburg, MD 20878

The first member of the ETX family of "class-independent" switches, the ETX5000 combines voice and data switching for large-scale voice, fax, and VPN connectivity. Offering broadband network interfaces, the ETX5000 utilizes high-performance switching and high network availability. The switch features redundant equipment and hot-swap support for all Universal Service Modules (USM), as well as a compact design.

The switch can use DS3, E3, STS-1, OC-3, or OC-12 SONET interfaces to connect to the PSTN, and supports 100Base-T Ethernet interfaces for connecting to backbone routers. The product also utilizes OC-3c and OC-12c SONET interfaces for ATM and packet-over-SONET data protocols. Features include voice compression, echo cancellation, silence suppression, and comfort noise generation, as well as support for standard codec algorithms. The ETX5000 also offers remote and local management through SNMP and a command line interface, in addition to FTP and Telnet support. For more information, visit the Salix Web site at www.salix.com.

Linkon Corporation
140 Sherman St.
Fairfield, CT 06430

The LinkNet gateway from Linkon acts as an IP switch and an enhanced services platform, and runs on Solaris for high scalability and performance. LinkNet offers voice and fax over IP, an IVR system, speech recognition, and signaling and call control functions. The platform integrates with SS7, and offers echo cancellation using a chip from Lucent.

Other LinkNet features include enhanced call detail recording with billing and bandwidth reports, a network management feature set, and an open architecture for dynamic application portability. The gateway is available in Sun PCI and Sbus configurations, and can support multiple T1/E1 digital circuits. It also offers the ability to switch among call routes and voice compressions for selectable voice quality. For more information, visit Linkon’s Web site at www.linkon.com.

NetBlazer 8500
Digi International
1101 Bren Rd. East
Minnetonka, MN 55343

Digi’s VoIP gateway converts voice signals into TCP/IP for transmission over standard IP networks. The NetBlazer 8500, based on open standards, uses an integrated high-performance switching architecture for connection from an ISDN PRI interface to a DSP-based voice compression card. Scalable from 30 to 120 ports, the NetBlazer supports dial-in from remote clients through ISDN, GSM, and modems. It also supports voice calls using analog phones.

Several NetBlazers may be cascaded to form a single POP, for added redundancy and fault tolerance. The gateway supports standard voice codecs, for interoperability with other vendors’ products as well as integration into existing backbone networks. Connection to authentication, application, and accounting servers is possible through the RADIUS standard. Other features include line management through a LAN interconnection, and dynamic and static assignment of IP addresses. For additional information, visit Digi’s Web site at www.digi.com.

Nokia IP Telephony Gateway
555 Legget Dr., Ste. 400
Kanata, ON K2K 2X3 Canada

Nokia’s IP Telephony Gateway works with the Nokia Call Processing Server and Services Control Node to provide high-quality voice and fax over IP. The gateway enables phone-to-phone, PC-to-phone, Web browser-to-phone, and fax-to-fax transmissions over IP networks. VPN connectivity is also available through Nokia’s Toll Bypass Application, which provides an IVR system, dial plans, and call routing. The gateway may be stacked with other Nokia gateways to provide hundreds of ports, for use by thousands of users. Gateways in a single point of presence may also be configured to balance incoming calls across all platforms.

The gateway offers real-time G3 fax, and features packet reconstruction for lost or delayed packets. Other features include auto-detection for voice and fax, silence suppression, G.165-compliant echo cancellation, DTMF detection and generation, and a programmable IVR. It also offers user authentication, accounting, and authorization through Nokia’s Services Control Node, as well as dynamic routing of calls, call load balancing, and network configuration and monitoring through the VCTool. For more information, visit Nokia’s Web site at www.nokia.com.

Telecom 2000-CCS
e-Net, Inc.
12800 Middlebrook Rd, Ste. 200
Germantown, MD 20874

The T2000-CCS (Carrier Class Switch) from e-Net, part of the Telecom 2000 series of gateway products, is a NEBS-compliant, port-dense VoIP switch designed for co-located carriers. This open, programmable switch offers 4,096 ports and is compliant with H.323, SS7, and SNMP. Featuring a non-blocking switching matrix, the T2000-CCS can provide up to 17 universal port/service call slots, as well as optional redundancy for all systems.

Additional features include a comprehensive set of Integrated Intelligent Call Services — software resources for reduction of application development time, simplified network management, and real-time protocol generation. The gateway features a high-density network interface, as well as service resource cards for enhanced density price and performance. It offers various voice compressions, and features an SNMP agent and a management information base for local and remote resource management and configuration control. For more information, visit e-Net’s Web site at www.datatelephony.com.

Digitcom Corporation
2202 Broadway Ave.
Santa Monica, CA 90404

The IntraVoice Internet gateway servers from Digitcom convert, compress, and packetize voice into IP using DSP-based compression, for full-duplex transmission over data lines. Echo cancellation, along with the built-in compression, produces minimal latency and near toll quality voice. The IV-2400 model features a 24-port T1 digital interface, as well as digital tone generation and detection, and a digital vocoder.

The model is available as a rackmount PC with hot-swappable dual power supplies, as well as disk mirroring. It also features Windows NT and comes with Internet gateway routing software, and call accounting/billing utility software. Other features include single TCP call control sessions for all calls, generation of a transaction log, and exact billing. Global network management is also available for efficient call routing, as well as call statistics, prompting, and call announcement. For more information, visit Digitcom’s Web site at www.digitcom.com.

Network Exchange 2550
NETRIX Corporation
13595 Dulles Technology Dr.
Herndon, Virginia 20171-3424

This multi-service switching platform from NETRIX transfers voice and data over ATM, frame relay, X.25, TDM, and ISDN. The Network Exchange 2550 acts as a carrier edge switch or an enterprise backbone, with scalability up to 2,400 ports and port speeds from 600 bps to 45 Mbps. The gateway maximizes bandwidth through dynamic sharing of frame, packet, and circuit traffic, as well as the combination of data, compressed voice, and image traffic.

Delay-sensitive voice and SNA traffic are easily transmitted along with other types of data through processing, priority, and fairness algorithms. Logical port capabilities also enable multiple types of traffic to travel a single virtual circuit, and a bandwidth-on-demand feature allows capacity to be added automatically through ISDN or other dial services. The gateway is administered by a network management system for monitoring, configuration, and control of the Network Exchange. For more information, visit NETRIX’s Web site at www.netrix.com.

Tempest Data Voice Gateway (DVG)
Franklin Telecom
733 Lakefield Rd
Westlake Village, CA 91361

Franklin’s Tempest DVG transmits voice and data over existing network infrastructures and the public Internet. The Tempest can transfer up to 24 channels of voice traffic over T1 trunks or analog lines through FXO/FXS ports, while an Ethernet LAN port enables the connection to the data network infrastructure. It may be based in a central office, or on the customer’s premises, and can be dialed into from the PSTN or an office PBX.

Available in a rackmount chassis, the Tempest comes equipped with a unique DSP board with multiple vocoders for auto detection and control of voice and fax calls. The boards feature 12 channels of voice compression and a variety of vocoders, including the G.729b for silence suppression, and the G.165 for echo cancellation. The Tempest also offers fax relay and detection, and is T.38 compliant. It utilizes an Intel PCI Bus PC interface, and offers processing power of 40 Mips per DSP or 480 Mips per board. For more information, visit Franklin’s Web site at www.ftel.com

ServiceXchange SX-10
ACT Networks
188 Camino Ruiz
Camarillo, CA 93012

This multiprotocol gateway from ACT Networks offers voice, fax, and data transmission over IP and frame relay networks. The ServiceXchange SX-10 features a modular design, with support for up to 96/120 (four T1/E1) simultaneous calls in a single chassis. The gateway is H.323-compliant, and enables PC-to-PC, phone-to-PC, phone-to-phone, and G3 fax connectivity through the public Internet, frame relay, or corporate intranets via IP.

The ServiceXchange offers the ACTview 2000 management system, for network administration through SNMP, as well as a point-and-click GUI for easy navigation. It is also available with the Intracell Service Management system for call accounting and billing, as well as least cost routing functions. Other features include four frame-relay WAN ports, and a 10/100 Base-T LAN port. The gateway also features voice compression of 6 or 8 Kbps, echo cancellation, silence suppression, and comfort noise generation. For additional information, visit the ACT Networks Web site at www.acti.com.

Hypercom Network Systems
2851 West Kathleen Rd.
Phoenix, AZ 85053

The IP.tel gateway from Hypercom offers up to 960 voice/fax channels or 32 T1/E1 circuits on a single node for packetized voice delivery over IP, frame relay, VSAT, and ATM. The gateway lets users leverage their incumbent billing and calling card applications by generating call detail records that integrate with many different billing systems. A switch/gateway/IVR function also enables access control and billing capability.

IP.tel uses DSP technology for clear voice quality, and allows users to control options for echo cancellation, signal quality monitoring, and dynamic buffering. The gateway also features hot-swappable, redundant processor modules and power supplies for minimal downtime. Other features include T1/E1 voice /fax processors, and a voice network interface that can handle T1/E1 CAS/CCS, ISDN PRI, and analog transmissions. For additional information, visit Hypercom’s Web site at www.hypercom.com.

Fusion 3.0
Natural MicroSystems
100 Crossing Blvd.
Framingham, MA 01702

The new Fusion 3.0 gateway platform from Natural MicroSystems offers 60 ports of VoIP in a single slot. The product supports calls from many different Internet telephony clients, including VocalTec’s Internet Phone, Microsoft NetMeeting, the VoxPhone from VoxWare, and the WebPhone from NetSpeak. The gateway is H.323-compliant, and also supports the MGCP, SGCP, and SIP protocols.

Fusion 3.0 offers an IVR, as well as T.37 fax ability, to allow developers to deliver high-value messaging solutions. It includes dual 10/100 MB Ethernet interfaces, as well as real-time fax support. The gateway also supports a variety of industry-standard vocoders. It is made up of the Alliance Generation (AG) DSP board, and AG Realtime/2 daughtercard for vocoding in real time, and a TX Series board for IP routing and data protocols. The product supports a T1 in two ISA slots, and can scale up to four T1s per chassis. For more information, visit the Natural MicroSystems Web site at www.nmss.com.

eFusion, Inc.
14600 N.W. Greenbrier Pkwy.
Beaverton, OR 97006-5764

The eStream enhanced Internet services application gateway from eFusion provides a variety of connectivity options targeted at the carrier class. eStream enables PC-to-phone direct dial, phone-to-phone connectivity through VPNs and a PSTN bypass, and phone-to-PC, PC-to-phone, Web-to-phone, and Web-to-Agent transmissions. eFusion’s Push to Talk technology allows users to initiate a phone call with a customer service agent directly from a Web site.

The eStream open platform is H.323-compliant, and easily interfaces with the current network infrastructure for administration, operations, maintenance, and provisioning. PSTN connectivity is provided through T1, E1, and ISDN lines, and each gateway can support up to 96 T1 and 120 E1 lines. The gateway also supports Internet phone clients that use the G.723.1, G.729a, and SC8300 codecs, and features an Internet Call Assistant for registering users with the gateway and setting up Push To Talk icons. For more information, visit eFusion’s Web site at www.efusion.com.

MultiVoice for the MAX
Ascend Communications, Inc.
One Ascend Plaza
1701 Harbor Bay Pkwy.
Alameda, CA 94502-3002

This integrated VoIP solution from Ascend allows voice traffic to travel from the PSTN to IP or frame relay networks, intranets, and extranets. MultiVoice for the MAX is part of the MultiVoice family, and is available for the Ascend MAX 2000, 4000, and 6000 WAN access switches. When integrated with the 6000 model, the MultiVoice can provide up to 48 ports of T1 connectivity and 60 ports for E1.

The MultiVoice supports several different levels of voice compression for reduction of data throughput requirements, and also emulates the PSTN through DTMF tone detection and generation. It supports phone-to-phone and phone-to-PC connections, and works with an Access Manager to establish and terminate calls. Other features include backup for existing voice trunks, and simplified management for support of small and branch offices through a single packet network. For more information, visit Ascend’s Web site at www.ascend.com.

PulsePoint Enhanced Application Platform
PulsePoint Communications
6307 Carpinteria Ave.
Carpinteria, CA 93013

The open architecture and standards-based design of PulsePoint’s Enhanced Application Platform allows service providers to upgrade and provide new features to customers without interrupting their current services. The platform delivers 99.996% reliability, and is developed to ensure almost no system downtime from component failure. Upgrades may be done while the system is online, and surveillance and alarming for multiple systems can be shown on a single screen, through SNMP and HP OpenView.

Installations and upgrades to the platform may be performed over the network or by CD-ROM, and low level maintenance and recovery can be done remotely, either through dialup RS-232 access or PC Remote. Visual Basic scripting may also be used to simplify repetitive tasks, and problem resolution is simple with nested exception reporting, which is customizable in any language. The platform is available in three configurations of 96–240 ports, 240–480 ports, and 480–1,920 ports. Components may also be scaled independently. For more information, visit PulsePoint’s Web site at www.pulsepoint.com.

Sonus IP Telephony Suite
Sonus Networks, Inc.
5 Carlisle Rd.
Westford, MA 01886

The IP Telephony Suite from Sonus includes the Sonus Gateway Switch (GSW), and the Sonus Open Services Architecture, an open platform for IP telephony services development. The GSW was conceived and built exclusively as a carrier-class solution, and provides the interconnectivity, capacity, and quality to meet the most stringent demands of the public carrier networks. The suite also includes the Sonus SS7 Signaling Gateway, which provides a high-performance interface to the carrier SS7 network.

Features of the suite include 99.999 percent availability with full redundancy and hot-swap capability; high scalability (a single GSW shelf can support more than 8,000 calls, while a fully configured GSW can support more than 64,000 calls); one-stage dialing; and toll-quality voice. The product is geared toward converged-network and incumbent carriers who need to introduce new, interoperable services in a timely manner. For more information, visit www.sonusnet.com.

Cisco Systems, Inc.
170 West Tasman Dr.
San Jose, CA 95134

Cisco’s AccessPath-VS3 is a fully integrated dial access system that merges universal Internet telephony, switching, and high-speed backhaul routing into one scalable architecture for quick and reliable transmission of toll-quality voice and fax services over data networks. The system is equipped with assembled components and an easy-to-use Web-based configuration tool for quick deployment. ITSPs and corporations can use AccessPath-VS3 for intracompany phone calling and faxing, phone-to-phone communications through PBXs and key systems, real-time fax to fax exchanges, and PC-to-phone transmissions.

Other notable features include built-in CDR, an integrated H.323 gatekeeper; a 48- to 1008-port capacity for T1 configurations (60 to 1,260 ports for E1); availability of 99.99998 percent; and support for 42 to 168 ISDN Primary Rate Interfaces (PRIs). Components include the Cisco AS5300 platform with voice/fax feature cards, Cisco IOS software, routing and switching products, a system controller and fully tested system configurations, and optional Cisco 3640 gatekeepers. For more information, visit www.cisco.com.

Castle Networks, Inc.
One Executive Dr.
Chelmsford, MA 01824

The Castle Networks C2100 Services Mediation Platform (SMP) is a next generation central office (CO) platform built from the ground up to switch both voice and data traffic. The C2100 SMP unbundles voice switching from Class 5 equipment and allows service providers to quickly expand into regional markets offering integrated voice and data services, while providing a platform for peering with long-distance providers and efficiently feeding data traffic to ISPs.

By applying the economies and efficiencies of data networking technology to the PSTN, the C2100 SMP employs a distributed processing model that unbundles core voice switching from the Class 5 mainframe, while still leveraging its feature-rich software. And, with support for data inherent in its architecture, the platform takes advantage of the ubiquity and reliability of the PSTN to migrate carriers and end users from today’s circuit-switched environment to tomorrow’s converged networks of circuit, packet, and cell technologies. Base system pricing starts at $101,500 and includes a chassis, SCE, CPE, and channelized DS-3 trunk/access module (DXM). For more information, visit www.castlenetworks.com.

DM3 IPLink, Release 3
Dialogic Corporation
1515 Route Ten
Parsippany, NJ 07054

Dialogic’s single board DM3 IPLink combines phone system and data network interfaces with mediastream processing for real-time voice and fax support over IP. The board features T1 and E1 PSTN interfaces, and also supports Euro-ISDN, Robbed Bit, and other protocols. Features include multiple vocoder support, as well as variable jitter buffer control to compensate for IP traffic delays.

The DM3 solution supports as many as eight boards per system, and offers up to 240 channels per chassis. Other features include T.38 real-time fax support, and H.323 interoperability with an H.323 stack built onto the board. The board also supports SNMP for system status updates, as well as S.100 call processing support with Dialogic’s CT Media software. The product also provides call progress updates, DTMF direction and generation, voice recording, noise tolerance, and volume control. For additional information, visit Dialogic’s Web site at www.dialogic.com

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