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
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
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.
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
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'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
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
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.
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).
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
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 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
- 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.
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).
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
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'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 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
- 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'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
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
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
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 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.
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.
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'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
- 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' 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'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'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.