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Special Focus
May 2000

Greg Galitzine The Right Tool For The Right Job

BY GREG GALITZINE


A carpenter is only as good as his tools. You can't possibly make a straight cut without using a true square and a sharp saw. Likewise, you can't drive railroad spikes with a 12-ounce rubber mallet. And I defy you to use a claw hammer to rip a piece of wood to an exact measurement (I've tried, with varying degrees of success, and I don't recommend this to anyone).

Likewise in the field of developing Internet telephony solutions. The proper tool for the proper job is a must. In this brief roundup, we take a look at some of the development tools and platforms available today. One thread running throughout the whole collection of companies offering products in this space is an adherence to standards, whether the standard is H.323, SIP, MGCP or some other offshoot.
I urge the developers reading this article to take a close look at the vendors mentioned herein. These tools are designed to provide you with the platform you need to begin building the next generation of Internet telephony products and services. But choose wisely; pay attention to detail; and (oh no, here he goes again!) do your homework. Visit the Web sites, call the vendors, and test their products. You'll quickly find the tool that best fits your needs. In my former life as a carpenter, there was no substitute for a good and trusted tool. I'd like to go into more detail about my love affair with a 24-ounce Estwing framing hammer, and how it actually saved my life one day, but that's another tale for another time. For now, familiarize yourself with these products, and keep a constant eye out for updates, upgrades, and new offerings. This industry is dynamic, and it moves quickly. Don't be left behind without the proper tools.

BOARD-LEVEL PRODUCTS
Anatel Communications Corporation, an Analogic company Anatel's TAP-810, the first CompactPCI member of the company's telephony applications processor family, is a "hot-swappable" carrier-grade VoIP platform specified for 120 channels of G.723.1, G.729a, and other standard and proprietary algorithms. Specifically designed for PSTN-to-VoIP connectivity applications, the new board combines high-density I/O and DSP power to facilitate large system implementation.

AudioCodes
The TrunkPack-VoIP/400-cPCI VoIP media streaming board is a high-density, hot-swappable, CompactPCI resource board with a capacity of 120 ports, supporting all necessary functions for voice and fax streaming over IP networks. The board is powered by AudioCodes' VoIP DSP chips, and supports 120 independent and concurrent channels with the most demanding algorithms, such as G.168-compliant echo cancellation, G.723.1, and G.729A.

The AC4804A-C signal processor combines toll-quality low bit rate voice compression, G3 fax relay, and other voiceband processing functions. The AC4804A-C features four independent full-duplex voice channels with user-selectable G.729A, G.723.1, G.727/G.726, and G.711 voice codecs and supports packetized voice communication. Many other built-in features support use of the AC4804A-C in a wide range of voice-over-packet networks. The features include silence compression, adaptive echo canceller, and DTMF relay.

Blue Wave Systems
Blue Wave Systems CPCI/C5420 ComStruct digital signal processor (DSP) platform, based on Texas Instruments' TMS320C54x DSP Generation, provides telecom system providers with the very high levels of processing power they need to deliver carrier-class voice, fax and data services to their enterprise and service provider customers, while also shortening their development cycle time and risk. The board, featuring 20 programmable TMS320C5420 DSPs and four PowerQUICC microprocessors, is designed to run Telogy's Golden Gateway software package, providing more than 120 channels of simultaneous voice, fax, or data over IP calls in one CompactPCI slot.

Brooktrout
Brooktrout's TR2001 is a powerful, high-density platform for IP voice and fax applications and gateways. The TR2001 platform can operate in a dual Pentium configuration that helps to minimize host processor loads and provides a high degree of scalability. In addition, for developers who want the same degree of reliability and scalability in a single chassis, the TR2001 now features multi-board synchronization, which allows four boards in a single chassis to handle up to 240 simultaneous calls. Furthermore, The TR2001 SDK provides high-level APIs and a complete development environment for rapid creation and ongoing enhancement of IP telephony products.

Commetrex
OpenMedia is Commetrex' implementation of the MSP Consortium's M.100 recommendation. M.100 is a media-processing API that separates media-processing software from the underlying hardware, fostering independent competition and development in these value-adding layers. OpenMedia is an open multi-stream, media-processing software environment that supports IP media-stream processing for VoIP and fax-over-IP (FoIP), as well as traditional computer telephony circuit-switched applications. OpenMedia reduces the cost of developing and porting media-processing products and supports straightforward media software deployment on multiple hardware platforms.

Dialogic (an Intel company)
Dialogic recently announced version 4.0 of DM3 IPLink. This latest IPLink release allows developers of IP telephony solutions to build robust applications for both the enterprise and public network.

Version 4.0 enables new IP telephony application developers to significantly reduce development time because of its single board solution architecture, and reduces system complexity for all developers. This allows the deployment of high-performance, high-density, IP-enabled enhanced services systems for the service provider, ISP, and voice portal market segments. The cost of deployment goes down even more for servers that require media processing, since there is no need for separate resources for individual features. With IP protocol flexibility, IPLink also allows easy interoperability with third-party softswitches and IP telephony access devices. DM3 IPLink Version 4.0 also supports SNMP and CompactPCI, allowing for high availability and remote maintenance.

DSP Research
The VIPER(NGEN) combines the power of 32 Texas Instruments 'C5420 DSPs with eight telephony I/O ports, an Ethernet interface, and a powerful CPU to provide telecom systems integrators and OEMs with a fully-integrated, voice-over-packet (VoP) application board. This VoP board provides up to 320 simultaneous channels of H.323 (G.723 and G.729A).

Natural MicroSystems
The CG 6000C offers 240 VoIP ports per CompactPCI slot, a mature and comprehensive software development environment, worldwide trunking certifications, and high-availability features.

A tightly integrated hardware/software design, the Convergence Generation supports the rapid, simplified development of IP media gateways, IP media servers, and enterprise communications. The CG 6000C hosts Fusion 4.0, Natural MicroSystems' VoIP software platform for the development of advanced real-time media streaming applications. Fusion 4.0 features a streamlined API set that delivers a powerful architecture for rapid development. The CG 6000C is also supported by Natural MicroSystems' Natural Access development and runtime environment.

Odin TeleSystems Inc.
The Odin Telecom FrameworX (OTX) adapter family from Odin TeleSystems Inc. is a collection of re-usable software and hardware components (building blocks), each implementing certain telecom functions. The OTX framework allows the rapid implementation of a variety of computer-based telecommunications systems. Examples include network elements (protocol converters, SS7 nodes), customer premises equipment (ISDN terminals, frame relay access systems), and test systems (traffic simulators, protocol analyzers). The OTX enables short product development cycles and cost effective end products. The platform features a software toolkit for protocol implementations and a modular hardware adapter family. The adapter family offers various network interface cards as well as generic processor daughterboards and application-specific daughterboards.

Performance Technologies, Inc.
The CPC376 is designed to provide telecom equipment manufacturers and OEMs with an integrated hardware/software product that expedites the development of telecom equipment applications for current and next-generation networks. The CPC376 is a two-port T1/E1 line interface for CompactPCI-based systems. Telecom features include stratum level 3 clock inputs and outputs for locking to selected off-board references, alarming and health monitoring, hot-swappable hardware, and high availability drivers and operating system support. Hardware features of the CPC376 include one Motorola MPC860MH PowerQUICC processor and 32 MB of DRAM.

A complete suite of WAN protocol products is available for the CPC376, including SS7, SS7 MTP-2, frame relay, LAPD, HDLC, and X.25. Operating system support includes Solaris, Windows 2000, Windows NT, and Linux.

PIKA Technologies
PIKA VoIP is a hybrid host-based and AllOnBoard DSP software solution that can be added to any of the PIKA hardware platforms. This approach utilizes the power available on today's PC host processors so that a low-cost, low-density VoIP application can be realized. Benefits of the PIKA approach include:

  • Can be added to All PIKA platforms now or later in the field;
  • Fax support;
  • VoIP without an expensive DSP resource card; and
  • Low per-port pricing.

A four- to eight-port solution, PIKA VoIP is targeted at small businesses. PIKA VoIP can be combined with any IVR, auto attendant, and voice-mail CTI application that exists today. PIKA AllOnBoard DSP algorithms are available on all PIKA hardware.

Telogy Networks, a Texas Instruments company Telogy recently unveiled its next-generation GoldenPort product, a Universal Port solution for the Remote Access Server (RAS) market. This new TMS320C5440 DSP-based implementation of GoldenPort delivers a 4x reduction in area per channel and a 2x reduction in power per channel over its nearest competitor, enabling equipment developers to significantly increase their platform densities while decreasing their cost per port. The new GoldenPort solution delivers the ability to run any application on any channel at any time. This advance allows ISPs and service providers to deliver new services and to achieve operational savings through the use of a single GoldenPort-enabled platform capable of transmitting voice over packet, fax relay, fax termination, modem relay, and modem termination services.

TESTING
Agilent
Agilent Technologies, Inc., recently announced that the Agilent Firehunter Internet service-assurance solution can retrieve and display important measurements obtained by Cisco networking equipment running Service Assurance (SA) Agent, a feature of Cisco IOS software. Firehunter now has the ability to configure SA Agent measurements and to incorporate SA Agent test results into Firehunter's customizable service models. These models simplify Internet service analysis by aggregating data into tree-structured graphical environments that allow ISPs to manage services proactively, deliver end-to-end Quality of Service (QoS), and demonstrate compliance with service level agreements (SLAs).

Catapult
The DCT product line is a comprehensive software-based telecom test system that combines power, versatility and ease of use to help customers rapidly develop, test, and deploy telecom products. DCT can simulate and monitor telecom network entities in a flexible, multi-user, multi-protocol, programmable test environment. DCT is designed to be scalable and leverages the strengths of Sun workstations by offering a modular software and hardware architecture. Catapult provides an extensive software library of protocol modules (including H.323, MGCP, SIP, and RTP/RTCP encoders and decoders) and a wide variety of high-performance physical interface options.

Hammer Technologies
Hammer Technologies recently revealed release 2.0 of their Voice over IP Test System (VoIPTS). Release 2.0 represents one of the first products for carrier-class gateways that integrates up to 480 ports of full-coverage ITU standard voice quality analysis (PSQM) with SS7, ISDN, and fax in a single, integrated test platform. Release 2.0 incorporates many enhancements to Hammer's VoIP test suite. In addition to SS7, improved PSQM density, and fax, this release incorporates new levels of user programmability, additional ISDN variants, enhanced scheduling, remote control, reporting, and monitoring features as well as the Call Profiler introduced with the Hammer DS3 test system.

Ixia
Ixia's introduction of its unique QoS Performance Tester with integrated Global Positioning System (GPS) technology is targeted both at the ISP and carriers who must be able to precisely and repetitively validate network performance issues to verify the QoS metrics that form the basis of their SLAs. The QoS Performance Tester is capable of measuring latency and jitter through the wide area network (WAN). Packets generated at location A have time stamps inserted from the GPS. The packets traverse the network and arrive at location B where another time stamp is inserted from the local GPS receiver. The Ixia technology calculates the latency between locations A and B on a packet-by-packet basis. The integrated GPS technology synchronizes to within 150 nanoseconds, time stamping at both the transmitting and receiving side.

RADCOM
RADCOM offers a complete Test Suite consisting of the following elements:

  • The H.323 call generator, which enables developers and service providers to benchmark, load test, and verify proper protocol implementation in voice-over-IP equipment.
  • The AudioPro Voice over IP analysis package, which provides comprehensive analysis of both voice/video streams and H.323 signaling messages.
  • The Internet Simulator, which emulates the behavior of WAN links and networks, enabling efficient testing of real-time applications before deployment on the network.
  • A latency and loss application that measures the delay introduced by any network device or segment by correlating data streams captured on either side of the device; and
  • Jitter analysis, which quantifies continuous, real-time media performance by measuring the inter-arrival time of packets in a stream and calculating average delay and jitter.

Shomiti
Shomiti's Multi-QoS software plug-in to the company's Surveyor 3.0 product enables the capture, analysis, and summarization of a broad range of QoS factors associated with H.323 and related multimedia IP traffic (voice, video, data). Multi-QoS delivers a rich set of reported and calculated data to validate QoS parameters presented by IP phones, PSTN/IP gateways, IP switches, and IPBXs on a call-by-call, or channel-by channel basis. Multi-QoS also supports SIP, Cisco's SSP, MGCP, and SGCP protocols. Multi-QoS detailed metrics and measurements also help to clarify the capability of the network infrastructure; discover problems affecting user quality such as delay, jitter, and loss; and identify when to increase network capacity.

SIMULATORS
Gordon Kapes
The System 930 Telephony Simulator is essentially a digital telephone network in a box. Among its many features, the system offers a choice of ISDN PRI protocols, robbed bit T1 protocols, or both. Two 1.544 Mbps digital interfaces permit an "end-to-end" simulation of a PSTN/VPN. The System 930 can emulate the network or user side of T1 or NI-2 compliant ISDN-PRI and 4ESS, 5ESS, DMS100/250, and GTD5 switches can be emulated as well. The System 930's software facilities accommodate Q.931 public or private networking protocols.

The System 930's internal bulk call generator can be programmed to initiate up to 23 or 24 simultaneous calls on a repetitive basis and the call generator scripts can be created to approximate myriad "real-world" events.

Teltone
Teltone's Telecom Simulation Platform (TSP) gives developers both T1 and POTS simulation in one compact unit. Developers can build any size simulation by connecting multiple units to a single PC, and can mix and match the hardware modules that plug into the TSP base chassis, thus changing the configuration lab simulations. There are four slots at the rear of the TSP into which the user can plug in the following available Network Interface Modules:

  • TSP T-1 Module
  • TSP Dual T-1 Module
  • TSP 4-Port POTS Module
  • TSP 8-Port POTS Module

CC&T
EMIP-1 is a turnkey solution designed to emulate Internet behavior. The user can easily configure EMIP-1 with the desired network delay, drop rate, and maximum bandwidth (up to 50 Mbps). Various distribution functions or even actual network trace data can be used to model the network delay and drop rate. There is no need to change the application software or network protocols. EMIP-1 is an excellent tool for network-related product developers and evaluators.

Some of the EMIP-1's technical specifications include:

  • Network Interfaces: One 10/100Base-T;
  • Operating System: Linux 1.35;
  • Memory: 64MB; and
  • Disk: 4.3 GB.

Shunra
Shunra Software's The Cloud 2.0 enables development engineers to degrade network traffic on any LAN, emulating a WAN, in order to evaluate any application, product, or technology on any WAN configuration. The Cloud 2.0 offers upgrades to its latency, packet loss, and bandwidth emulation features: Emulating jitter is now even simpler thanks to the addition of linear dynamic latency; granularity of degradation can be pre-set down to .01 percent, a hundred times finer than the previous version of The Cloud; and users can now add multi-byte overhead to packets in order to emulate different data link layers such as PPP (point-to-point protocol) and DSL.

PROTOCOL STACKS
dynamicsoft
The dynamicsoft SIP User Agent provides a set of development tools for equipment manufacturers and service providers, including multi-level APIs in Java and C++. The SIP User Agent manages the basic connection between the call originating party and the call terminating party. It can support the delivery of enhanced services while providing carrier-class reliability. With a wide range of functionality, the SIP User Agent is designed to immediately enable products to communicate using the SIP standard. Working closely with various industry groups, dynamicsoft will continue to add additional functionality to the SIP User Agent. Currently, SIP-T (SIP for Telephony Interworking), previously known as SIP+, is available as an option.

elemedia
The elemedia H.323 Protocol Stack product line implements the ITU family of H.323 protocols and makes them available through two distinct lines of H.323 protocol stacks that provide a wide range of product options for developers. First, the PX3230S Classic H.323 Protocol Stack, elemedia's next release of its Classic Protocol Stack, will be newly-rearchitected to take advantage of a lower-level C core while utilizing a higher-level C++ Application Programming Interface (API). This combination will support a variety of applications -- from general-purpose processors and desktop applications to carrier-class solutions. Also, elemedia's EX3230S Embedded H.323 Protocol Stack is specifically architected for embedded processor systems that typically rely on lightweight operating systems and require minimal footprint, modularity, and integration flexibility.

Mastermind Technologies
Mastermind's MasterVox for NT v3.0 features fully integrated VoIP support on Natural Micro-Systems' Fusion series of software and hardware products.

Features of MasterVox VoIP include:

  • Ability to combine telephony switching, IVR, voice compression, IP gateway, and other CTI capabilities in a single system;
  • Ability to perform IVR functions across IP networks, including play and record of voice and DTMF/MF digits;
  • Support for the complete range of analog and digital telephony line interfaces (e.g., T1, E1, ISDN, Loop Start) on the PSTN side of IP applications;
  • H.323 call control and G.723.1, G.711, and GSM audio encoding;
  • Vocoding is off-loaded onto the high-speed NMS DSP boards, minimizing CPU load and latency; and
  • Support for up to 96 IP calls in a single PC.

Netergy Networks (formerly 8x8;)
Netergy Networks' Veracity Voice over IP (VoIP) Software Stacks implement industry standard VoIP protocols. H.323 v2, MGCP, and SIP are available today; these stacks share an industry standard POSIX-compliant operating system, TCP/IP stack, comprehensive network services, and complete audio library including ITU G.711, G.723.1, G.726, G.728, and G.729A/B with audio support services.

The Veracity software stack is interoperable with over 30 H.323 v2 endpoints, gatekeepers, and gateways including: Cisco GateKeeper (Cisco 3600); Cisco Gateways (Cisco 2600, Cisco 5300); and RadVision Gateway and GateKeeper (RadVision L2W-323).

RADVision
RADVision's H.323 core protocol enables the creation of real-time voice and video H.323 calls over IP networks. All H.323 entities (terminals, gateways, gatekeepers, multipoint controller units) require an embedded H.323 protocol stack. The RADVision H.323 protocol toolkit is available for numerous operating systems including various Unix versions, real-time operating systems for embedded systems, as well as Windows, including Windows NT and Win CE. Recently, RADVision announced Version 2.6 of its H.323 protocol toolkit, optimized for demanding carrier-grade applications and embedded systems. The company also announced the availability of MGCP toolkits, designed to provide the building blocks for next-generation media gateways and media gateway controllers.

Trillium
Trillium's MGCP software solution is designed to enable communications equipment manufacturers to develop high-performance, carrier-class IP telephony and PacketCable equipment that meets the performance, capacity, and scalability requirements of next-generation networks. Recently announced, Trillium's MGCP offering shatters the industry performance barrier by demonstrating more than 23,000 messages per second on media gateway controllers and more than 24,000 messages per second on media gateways in a code footprint of less than 200 KB. The solution consists of a gateway control protocol (GCP) implementation of MGCP for both media gateways and media gateway controllers.







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