TMC Labs' Gateway Shootout (Part I)
As Internet telephony gateways evolve from a fringe technology to a mainstream tool,
their developers face the challenges of integration, reliability, audio quality, and
gatekeeper functions. Covering the trend, TMC Labs is shifting its gateway testing methods
from qualitative and stand-alone testing to quantitative and comparison testing. We will
test T1 and fax options of gateways in the future, but Part I of our shootout is dedicated
to the analog voice functionality of gateways from Inter-Tel, Linkon, and Nuera, based on a PC, a Sun platform,
and a black box, respectively. We also chose these gateways based on their installation
process, documentation, support, competitive pricing, and mainstream availability.
Equipment we used included the Hammer IT 2.1.3 software and the Hammer VoIP 2.0.2 test
suite, plus a TLS-4 analog line simulator from the Teltone Corporation. You can visit the
Web sites of these companies at www.hammer.com and www.teltone.com, respectively.
TOOLS & METHODOLOGY
The most important aspect of this equipment was Hammer's VoIP suite. Earlier versions of
the Hammer IT VoIP suite included Perceptual Speech Quality Measurements, but version 2.0x
is the first version to include testing for speech latency. Midway through our tests,
Hammer provided us with an even newer revision because Hammer's disconnect tone conflicted
with Linkon's handshake tone. This conflict is explained below in the Operational Testing
An important part of the prompt test is measuring latency. This works by listening for
DTMF tones that have been played and time-stamped at the originating Hammer channel. These
tones travel to the local gateway, where they are compressed, packetized, and sent across
the network. At the remote gateway, the packets are reassembled, decompressed, and sent to
the terminating Hammer channel, where they are time-stamped again. The difference between
time stamps equals the latency. This path consists of an analog line connection stemming
from one channel on the Hammer to the first gateway, a compression phase (2) at the
gateway, a network connection (3) to another gateway where the speech undergoes a
decompression phase (4), and finally, another analog connection (5) from the second
gateway to a terminating channel (final destination phone number) on the Hammer unit. This
provided us with five distinct phases that all speech would travel across: Two analog line
connections, a compression and decompression phase, and a network path connecting the two
Our goal was to examine only the latency introduced by the gateways because of
compression and decompression algorithms, so we had to minimize the latency effect of the
analog and network connections (phases 1, 3, and 5). Better gateways don't just compress
and decompress the voice, they do it as quickly as possible to minimize any latency
The IP network between the two gateways was easy to manage because we used a
"clean" network with a 10-BaseT hub connected only to a PC running Gatekeeper
and network management software. Meanwhile, the analog circuit between each gateway and
the Hammer channels depended on whether the gateway being tested used FXS ports, which
provide their own dial tone, or FXO ports, which need dial tone provided to them. Only the
Nuera gateway used FXS ports, so we connected it directly to the Hammer system. The
Inter-Tel and Linkon gateways used FXO ports, so we used the Teltone simulator between the
Hammer and the gateways to generate dial tone. For these two gateways, we also subtracted
the Teltone-induced latency (12 milliseconds for each half of the Hammer-to-gateway
circuit) from the final figures.
Another important component of the prompt test are the PSQM scores, which determine the
quality of speech according to the benchmark assessment of a group of human judges
endorsed by the International Telecommunications Union (the ITU-T P.800 and P.861
standards). Visit the ITU's Web site atwww.itu.org. The algorithm plays sound clips chosen
for the range of sounds that they produce, not for the clips' actual meaning, which tend
to have an eerie and nonsensical nature. (Examples include "I worship wooden idols
I want a minute with the inspector;" "You will have to be very quiet
There is nothing more to be seen;" and a nasal man's voice drawling over the
word "hello.") Each of the sentences is read in a male, female, and child's
voice, encompassing a range of tones. Like the DTMF tones in the latency test, the sound
clips are compressed, packetized, sent across the network, and decompressed by the remote
gateway, which plays the clip back to the Hammer. The Hammer then takes this signal and
passes it to the PSQM algorithm along with the original prompt from the voice library. The
PSQM algorithm compares the received prompt to the original one and reports a PSQM score.
To complete our testing procedure, we configured the prompt test to produce 100 test
calls with sufficient time between calls for each gateway to disconnect and reset itself
for the next call. We found that 50 calls per hour provided an adequate time-out between
calls. We performed the prompt test four times on each set of gateways: once for latency
testing and once for PSQM scores, using two compression schemes. Finally, we also ran the
PSQM test on the Teltone unit, but the results of this test are for comparison purpose
only. As Hammer engineers explained to us, the PSQM scores are non-linear and are attained
using complex algorithms. Trying to "subtract" the Teltone's PSQM scores from a
gateway's actual scores would not give an accurate figure.
7300 West Boston St.
Chandler, AZ 85226-3224
Price (approximate): 8-port, $4,500/port; 24-port, $1,000/port; call for more
Inter-Tel's Vocal'Net 1.2 is a PC-based Internet telephony gateway that supports both
voice and fax over digital T1 and E1 and analog WTI-8 and AG-8 interfaces. The gateway
offers four compression modes - GSM120, GSM40, G.723.1, and G.711 - and is compatible with
a large variety of PBXs and CO stations. Unit programming and maintenance is accomplished
through a well-designed Windows GUI, as well as several DOS-based monitoring functions.
We tested an eight-port analog AG-8 gateway, which has minimum system requirements of a
Pentium 133 or greater running Windows NT Workstation 4.0 with Service Pack 3, 24 MB of
RAM, a one gigabyte hard disk and capacity for three full-size 16-bit PCI cards. Starting
with a "clean" PC running Windows NT (but not the service pack - that comes
last), users install the AG-8 card, the AG-DSP/RT card and the TX-2000 LAN card. The three
cards are linked using a 40-pin MVIP ribbon cable. Next, users install an optional fax
card, then the Vocal'Net software, followed by the NT Service Pack.
Perhaps it was just because our test PCs were brand new, but we found the cards
difficult to install. Setting them into the appropriate slots took an unusually large
amount of physical effort. This concerned us, because we were afraid that too much
pressing on the cards could damage them. To protect against this, we took extreme caution
when pushing the cards into the slots. The AG-DSP/RT card did suffer a bit - it had a
daughter card attached, which tended to separate from the main card. We removed this card
and determined that the damage was purely cosmetic, then re-set it. It worked fine.
Finally, we connected each machine to a hub.
Configuring the Vocal'Net software was more tedious than installing the hardware. After
running the setup wizard and rebooting each machine, we connected to the server using the
default values for the IP address and password. We chose the menu option
"network-table-load" and then opened the PSTN-to-IP routing table.
Algebraic-style dial patterns and dial rules are used to determine who can call whom.
Here, users also determine the call-billing rate if one is desired, plus gateway name
descriptions, fax options, and SPID (Service Provider ID) options.
Once these options are saved on each unit, the next step is configuring options under
the Miscellaneous Options Table. As its name suggests, this menu has a variety of
features, including server password, prompt language, maximum number of simultaneous calls
and vocoder type. In particular, we needed to change the Inter-Digit Timer in order for
the Hammer unit to send the DTMF digits without a time-out message. The Inter-Tel came to
us with a factory preset of 5,000 ms. We changed this to the minimum allowable interval of
1,000 ms so we could complete our testing.
Our user's manual was still a preliminary version, but we're very impressed so far. The
manual's 11 chapters cover installing digital and analog gateways, programming, PBX setup,
monitoring software, network management, and troubleshooting. There are diagrams, charts,
and illustrations throughout. We especially like that the document is written in language
simple enough for a typical MIS manager to understand: Users do not need advanced
telecommunications degrees to configure an Inter-Tel Vocal'Net gateway. We also like that
there is significant discussion of compression schemes and the Internet telephony concept.
The troubleshooting section is equally detailed. We like this manual for its attention to
detail, and we strongly suggest reading the appropriate chapters before attempting any
installation. It can be overwhelming for a first-timer. Overall, we feel that this manual
is very good, and the complexity of the product makes the simplicity of the manual that
much more impressive.
Certain Vocal'Net features stand out, although we didn't utilize all of them. The entire
unit is programmable through text files, and every prompt is a documented .WAV file that
can easily be changed. We also can't overlook the numerous call accounting features and
fax options, many of which make Inter-Tel's gateway different from the competitors. Our
favorite feature is the monitor program. Monitor has the same no-frills environment as the
Vocal'Net programming database, and offers event logging, password protection,
configuration of network options, helper application selections, and system information.
Other features include:
- Supports caller ID and ANI data.
- Automatic database upgrade.
- Customizable IVR.
- Natural MicroSystems Fusion II support.
- ODBC-compliant SQL database.
- Oracle 7 support.
- Year 2000 compliance.
The four compression schemes that the gateway offers are the standard supplied by the
Natural MicroSystems Fusion-based line of products. These are:
- GSM120 and GSM 40, compressing speech at 13.3 Kbps.
- G.723.1 with silence suppression at 6.4 Kbps.
- Both a-law and mu-law G.711 at 64 Kbps.
140 Sherman St.
Fairfield, CT 06430
Price: 4 channels - $18,189; 8 channels - $23,238.
The LinkNet is a Sun Microsystems based Internet gateway that runs under the Solaris
operating system. The gateway is designed to function both as an IP switch and as an
enhanced services platform capable of handling fax, interactive voice response, and speech
recognition. Currently, the LinkNet has been deployed with a variety of applications
including phone-to-phone communications over satellite networks, prepaid calling card
services, Java Phone call center applications, and in Internet and Intranet configurations
- both domestically and internationally.
We received two LinkNet Gateways as fully turnkey systems, entailing two gateways and two
Sun Sparc stations for controlling the gateways. Supposedly, all we had to do to make the
gateways work was follow a few simple directions - connect all the wires for the gateways
and the Sun computers, connect them on a LAN and run a sample application. After we
completed this setup, with the use of the Teltone Simulator and two analog telephones, we
were able to make calls from a port on one gateway to another port on the other gateway.
When we put the Hammer system into the circuit, however, we encountered one of the more
interesting troubleshooting sessions in some time.
At first, it seemed as if the Hammer was unable to connect phone calls though the
gateway system. The Hammer would place calls through the Teltone unit to the first
gateway. This required multistage dialing in order to get through the LinkNet's greeting
menu, but immediately after navigating this menu, it seemed that the Hammer was unable to
make a connection.
We called both manufacturers (Linkon and Hammer) and explained our situation, and we
went through an assortment of specifications about the boards that were used in both
products. What we found was that the two devices should be working together properly, or
at least, there was no physical reason that they should not.
This prompted us to delve further and examine the log files on the Hammer PC. What we
found was the calls were being connected, but that the connections were being terminated
before the Hammer could run its first leg of the test. With this knowledge in hand, we
tried to find the command in the Prompt Test script that was causing the LinkNet to
disconnect, but we were unable to determine the cause.
Rather than waste too much time searching for the proverbial needle in a haystack, we
called some of our contacts at Hammer and began to retrace the Hammer's commands. All
seemed remarkably commonplace until our contact mentioned that the Hammer system sends an
asterisk DTMF tone to handshake. Our contact did not need to say anything more, as we had
found the answer to our dilemma: The asterisk. As it turns out, this tone had a special
function for both devices. For the Hammer, it was used to handshake and establish
connectivity. Ironically, Linkon uses the same tone for disconnecting a call. The attempt
of the Hammer to handshake was actually disconnecting the calls.
To take care of this situation, Hammer arranged to write a new beta for the VoIP suite
that did not use an asterisk anywhere during the operational testing so that it would
comply with the LinkNet gateway (and by extension, any gateway that uses an asterisk to
disconnect). A few days later, we upgraded from the VoIP suite 2.0.13 to the brand-new
2.0.20. We set the test configuration and were relieved when the Hammer was able to run
through the Prompt Test without inadvertently disconnecting the call. It is important to
note that despite our troubles in getting this test to work, we hold no one to blame as
both the Hammer and the Linkon devices were performing according to their specifications.
The outcome is that the Hammer suite has now been successfully tested with yet another
gateway manufacturer and for that, we are pleased to have played our part in getting the
Hammer to "get along" with the Linkon gateway.
Documentation for the LinkNet came in three separate compositions. Two of these are fairly
lengthy guides that seem to be exhaustive in their detail. One of these manuals covered
the installation of the LinkNet gateways and a configuration guide to help the user with
sample applications. The remainder of this manual, and the entirety of the other, deal
with programming in the TeraVox/LinkNet space and technical specifications that only a
board developer could love.
The other form of documentation came to us in an uncomplicated two-page quick-start
guide that actually provided most of everything that we testers needed to know. Following
the commands on these sheets in a top-down manner successfully set up the sample
application, even though it took us a long time looking through the larger manuals and on
the phone with the manufacturers to realize that this was the case. Since the delay was
caused by a temporary incompatibility between the Hammer VoIP suite and the LinkNet, it
turned out the documentation for the LinkNet was accurate and should be rewarded for its
The LinkNet functions as both an IP switch and an enhanced services platform, and is
capable of supporting fax, interactive voice response, and speech recognition. LinkNet can
provide both voice and fax over IP, with an IVR front-end, and a wide variety of signaling
and call control functions.
The LinkNet offers open software architecture with SS7 integration and hardware-based
echo cancellation with Lucent's TECO chip. It supports from four analog ports to a full
digital T1. It also can achieve capacities of up to 96 ports in a single-node server
Linkon achieves up to two channels worth of full-duplex transmission at the DSP level
eliminating the need to physically bridge the two channels together. So, a four-port,
four-DSP analog board can provide four full-duplex Internet phone conversations. The
LinkNet also employs multiple compression algorithms on the DSP, and comes with
compression algorithms for mu-law G.711 at 64 Kbps, GSM 601 at 13 Kbps and a sub-band
coder at 24 Kbps. The LinkNet gateway also provides echo cancellation as a standard
The Linkon gateways also support integrated enhanced services such as IVR, speech
recognition, and speaker verification. In addition, the LinkNet has the ability to
dynamically switch among available call routes (Internet, intranet, or PSTN) and switch
among the different voice compressions for selectable voice quality.
Nuera Communications, Inc.
10445 Pacific Center Ct.
San Diego, CA 92121-1761
Nuera Communications produces the Access Plus series, a gateway that offers voice and
data integration over IP, frame relay, and TDM links. We tested the AccessPlus F200IP
model. The AccessPlus F200IP's voice ports support a variety of analog and digital
interfaces and are compliant with most voice signaling protocols. This model runs voice
over IP and voice over frame relay concurrently and can switch voice calls between any
connected ports over any supported protocol. These protocols utilize dynamic bandwidth
allocation with a range of compression rates from 4.8 to 32 Kbps. In addition, the model
supports other data equipment, such as IP routers, FRAD devices, and asynchronous
We received two AccessPlus F200IP units that came to us as turnkey systems. These two
units were black boxes and are configurable through any terminal emulation software or
through HP OpenView. Because our systems were small in scale, we used Windows' built-in
HyperTerminal program by connecting a DB15 cable from each gateway to a PC. We powered on
each unit, waiting for the finish of the five-minute LED test sequence before the units
transmitted information to the HyperTerminal program. We supplied a password and IP
address for each unit, and we set the date and the time for the system clocks.
The next step we took was to install the NueraView network management program on a PC
running Windows NT. This installation went without a hitch, though NueraView seemed to
install a countless number of small sub-programs, causing the installation to be a
time-intensive process. When we had the programs running, we set up a call routing table
configured for the two gateways.
With both units configured with IP addresses, we connected the two units across a
simple LAN. We connected each unit to a network hub that we kept separate from our
corporate LAN to eliminate "bursty" network traffic from affecting the results
on this LAN. To test our setup, we connected a computer to the network hub and opened a
DOS window to see if we could ping the units via their IP addresses. What we found was
rather interesting. We were able to ping the first unit; the second unit would receive
ping replies, but many of them would give time-out messages. We found this highly unusual,
so we called our contact at Nuera and explained the situation. He explained that there was
a high demand for people to test Internet gateways, and as a result, millions of dollars
worth of equipment was making the rounds across the world. This pertained to us especially
since it seemed that the gateway's network card had been damaged during shipment. We
received the replacement network card the next day and installed it. After we switched the
two cards, we proceeded with our tests. Thankfully, we were able to ping the unit after
replacing the damaged network card.
We were pleased that the change of network cards had solved our problem, so we decided
to call our contact and mention that all was well. While we were on the phone, we
described our operational testing procedure so that he could formulate the commands that
would set up our call flow. We explained that we wanted to automate the call flow from one
port on a Nuera unit to another port on the other unit by using the Hammer unit, and as a
result, he faxed us a list of instructions to be keyed in through the HyperTerminal
program. We typed the commands for both units and tested the current setup by connecting
two analog phones to the gateways and making a call. Strangely enough, the moment we
picked up one phone, the other phone rang immediately without waiting for us to type a
phone number. Knowing that there was no way for the Hammer to work with this
configuration, we called our contact back and explained that the Hammer required a phone
number to run through the script. With this new information in hand, our contact drew up
another set of commands for us. Once again, we used the HyperTerminal program to enter
these commands for each unit. With these changes in place, the systems were set up so that
the gateways could take calls from an analog phone and transmit them over the LAN segment
to the other gateway.
The documentation for the Nuera gateway was supplied to us in Adobe Acrobat format on a
CD-ROM that came with the units. The total documentation was comprised of five such files.
We found three of these particularly useful. These were the manuals titled Getting
Started, Installation and Maintenance, and the User's Guide. Each of these dealt with a
particular area that directly impacted our setup. The remaining two files were the General
Information Manual and the Programmer's Reference.
It should be noted that each of these Acrobat files was meticulous. Basically, if we
wanted to know something about the gateways, we knew it could be found within the manual.
Finding the information was another issue altogether. Even though the files were amply
indexed, the size of the files made it difficult to find very specific information.
However, we did find specific areas where the files were invaluable, such as command
summaries so that we could configure the systems or change the compression algorithms.
The Nuera Access Plus F200IP provides flexible voice interfaces. It can be configured with
a T1 interface for 24 channels or an E1 interface for up to 30 channels. The T1/E1s
eliminate subsequent distortion due to digital-to-analog conversions. In addition, analog
interfaces are programmable with as FXO, FXS, and E&M channels. The AccessPlus F200IP
provides a wide range of voice compression technology. These include:
- Proprietary E-CELP algorithms operating at 4.8, 7.47, and 9.6 Kbps.
- Standard algorithms G.728 LD-CELP at 16 Kbps, G.726 ADPCM at 32 Kbps, and an optional
G.729 CS-ACELP at 8 Kbps.
- Group III fax support at 2.4, 4.8, 7.2, and 9.6 Kbps.
In addition to these voice compression algorithms, Nuera utilizes a number of other
technologies to minimize network bandwidth. Silence suppression minimizes the bandwidth
usage during speech breaks while maintaining a level of ambient background noise for a
natural sounding conversation. Through the use of programmable voice packetizing,
bandwidth efficiency can be improved, while data fragmentation is configurable on a DLCI
basis to minimize delay of voice traffic over low-speed interfaces. In addition, the Nuera
contains asymmetric fax channels that minimize the bandwidth usage on the return path. The
Nuera AccessPlus 200IP model can also employ one or two FRAD data interfaces in addition
to digital and analog ports. These FRAD ports can be utilized in an asynchronous method
ranging from 75 bps to 115.2 bps and as synchronous FRAD devices from 1,200 bps to 2.0
Mbps. Other features of the Nuera Access Plus F200IP include:
- A jitter buffer to minimize end-to-end speech delay.
- An Integral echo canceller that adapts from 0-49 ms to guarantee consistent voice
quality over long tail circuits.
- Voice compression rates are negotiated to provide configuration flexibility.
- Modem transparency over voice channels.
The AccessPlus F200IP provides complete call processing by switching each call
independently with full digit and interface translation. This minimizes the number of
ports needed and also minimizes the cost of using the voice network. Digit translation
provides a unified dialing scheme capable of 20-digit translation and 40-digit outpulsing.
Furthermore, the ports on the Nuera are completely interoperable with each other, and
calls can be made between any types of ports (i.e., analog, digital or FRAD).
NueraView (for HP OpenView) provides configuration, statistics, diagnostics, and alarm
management for the network of gateways. The database in NueraView assures that the
important configuration information is safely backed up while multiple network managers
can access the network and view the information held within the database.
- Inter-Tel G.723.1
The Inter-Tel gateway we tested came with a number of different compression algorithms.
These included GSM40, GSM120, G.723.1, and G.711 (both a-law and mu-law). The first
compression that we tested was the G.723.1 codec, which compresses voice at about 6.4 Kbps
with the use of built-in silence suppression. We ran the prompt test on the Inter-Tel in
an FXO port configuration, meaning that the test needed the use of the Teltone Simulator.
The numbers we received showed that over the full path, the latency of a packet was about
177.5 ms. Subtracting the 24.0 second latency inflicted by the Teltone unit left us with
153.5 ms of latency caused by the compression and decompressions schemes. This amount of
latency places the Inter-Tel G.723.1 within the lower third, behind all but the Inter-Tel
GSM40 algorithm. Even though the G.723.1 codec had a high latency, the PSQM scores show
that the sound quality of this codec is among the best we tested. In this regard, only the
Nuera G.729 compression scored better, but it should be noted that the Nuera did not
employ the use of the Teltone simulator. Without a definitive method of determining the
exact effect of the Teltone on PSQM scores, comparisons between the two test setups based
on the PSQM criteria cannot be made conclusively.
- Inter-Tel GSM40
When we chose to test another compression for the Inter-Tel gateway, we were left with a
decision between the GSM40 and GSM120 algorithms. Researching the specifications on each
showed that the two algorithms compressed voice equally (13.3 Kbps), the only difference
being the packet size. Since the GSM40 algorithm creates smaller packets, we felt that
this algorithm would stand a better chance in the real world, so we decided upon that
algorithm. What we found was that this compression suffered the worst latency of the
various compressions we tested, but furthermore, the speech quality scores also suffered.
- Linkon GSM601
After we got the LinkNet Gateway functioning with the Hammer unit, we ran through both the
latency test and the PSQM test and tabulated the data. We received data that placed the
LinkNet squarely between the Nuera and the Inter-Tel in terms of latency. We determined
that it was running on the GSM601 compression algorithm by default. PSQM scores for the
Linkon, however, were not as favorable, though we should note that to our ears, the speech
was tolerable, hardly different than audio you'd hear on an analog cellular telephone.
Unfortunately, the other compression algorithms that the LinkNet uses were not within the
range that we were testing, so we only tested the GSM601 compression.
- Nuera E-CELP
When we received the Nuera gateways, they were already set to handle the incoming voice
with their proprietary E-CELP algorithm that compresses data to 9.6 Kbps. The Nuera was
the only machine that used FXS ports, and the latency numbers we received for the ECELP
compression were already lower than all the other Nuera compressions. However, this did
not prepare us for how low the latency figure would actually be. Even after we subtracted
the effect of the Teltone Simulator, the Nuera E-CELP still only had about half the
latency of the other two gateways, making the Nuera AccessPlus our top performer in that
category. The speech quality scores also ranked in a respectable range. Testing the system
with analog phones to our ears only confirmed the tests Hammer's results produced, as the
latency was almost indistinguishable, even when looking directly at the other speaker.
- Nuera G.729
Nuera had the largest array of compressions of the gateways we tested, ranging from 4.8
Kbps all the way up to 32 Kbps. We chose the G.729 algorithm since it compressed voice to
about 8.0 Kbps and is a fairly common compression. When we tested this compression, we
found that the latency barely changed at all from the numbers for the E-CELP compression.
Even better, the PSQM scores actually improved. Since we could directly compare the G.729
algorithm to Nuera's proprietary E-CELP, we could see that the G.729 definitely performed
better. Since the PSQM scores for the Inter-Tel G.723.1/Teltone configuration were
slightly worse than those for the Nuera, even removing the effect due to the simulator
would only make them approximately equal in this regard. This left us comparing the
latency scores of the two gateways, which made Nuera the clear leader.
And the winner is
Nuera's "black box" Access Plus F200IP, running the
G.729 compression, followed by Inter-Tel's Vocal'Net 1.2 running the G.723.1 compression.
The GSM compressions we used on the Inter-Tel and Linkon gateways did not fare as well,
and although Nuera's ECELP compression kept the same low latency figure as the G.729, it
significantly increased the PSQM scores.
We hope to see more competitors in the Internet telephony gateway field come through
our laboratory in the near future. We'd also like to remind readers that we developed our
testing methods with the goal of comparing the gateways to each other in a laboratory
environment. There performance in our labs may not rate the same in a real-world
utilization, where the scale would be much larger and where other software features may be
implemented, and we also remind readers that our tests focused on testing the gateways,
not the networks and environments they sat in. For example, we didn't consider factors
like each gateway's jitter buffer, which are usually dynamic in their adjustments.
Moreover, certain gateways might be better suited for certain applications - one may work
best for straight IP voice, another may excel at IP faxing, still another could be
considered the king of frame relay video conferencing.
Our tests should be only one part of your purchasing decision. We strongly recommend
attending industry events, visiting corporate Web sites, and insisting on seeing (and
hearing) demonstrations before making any buying decisions. Consider price, scalability,
support, integration, interoperability, and ease of use as well. Regardless of which
gateway you decide to purchase, we urge all organizations to keep in mind the cost
savings, added efficiency, and enhanced services available by using the features of a
packet-switched data network instead of a circuit-switched one.