specifically targeted at businesses looking to reduce toll charges
between frequently called sites. It is comprised of voice over
IP gateways that integrate seamlessly into your data network and
operate alongside with your existing PBXs, or other phone equipment
to simply extend voice capabilities to remote locations. is
designed to help you maximize the investments you've already made
in your data and voice network infrastructure.
Integration. With ,
you avoid the hassle and expense of replacing your existing routers,
WAN connections or phone system required by other VOIP solutions.
into your Ethernet network. Neither your phone service or network
is placed at risk. Minimum requirements: Ethernet network, WAN connection,
Thousands of Dollars Each Month. can
save your company substantial amounts in long distance charges.
Even if your company uses one of the most inexpensive calling plans,
a network can
quickly return your investment and begin paying you back.
With , you'll
experience consistent toll-quality voice connections. Using the
Perceptual Speech Quality Measurement (PSQM), Internet Telephony
magazine found that 's
MultiVOIP gateway delivered exceptional voice quality. In fact,
it outranked the competition.
H.323 and SIP protocols to provide complete interoperability with
other Internet telephony solutions. The inbound IP call protocol
is automatically detected and the voice channel is dynamically configured
to match. The outbound IP call protocol is configured with the phone
number allowing you the flexibility to call H.323 or SIP devices
from the same port. In addition,
also supports T.38 real-time fax relay for interoperability among
other VOIP equipment.
Fail-over. PSTN fail-over allows to
automatically route calls over the PSTN network when the IP network
is congested or completely down. This feature heightens reliability
and augments QoS when conditions threaten to undermine voice quality.
Utilizing user definable controls,
continually checks if the LAN/WAN is threatened by packet loss or
latency, or to see if the network is completely down. If it detects
switches to ˇ§survivability modeˇ¨ transparently routing all calls
over PSTN lines connected to the gateway.
monitor the connection and automatically switches back to the LAN/WAN
once the conditions improve.
Speech Technologies. supports
the Differentiated Services (DiffServ) Quality of Service (QoS)
protocol which sets priorities for voice and fax traffic and allows
transparent delivery. DiffServ helps move time-sensitive voice traffic
across even low-bandwidth WAN connections, like 56K and ISDN, with
the priority and quality required by voice. Other features such
as adaptive echo cancellation, forward error correction, bad frame
interpolation, tunable latency and dynamic jitter buffers, further
enhance voice quality.
Support for Multiple Telephony Interfaces. For maximum
investment protection, the 's
two, four and eight-port models accommodate changing communication
needs by providing a programmable FXS/FXO and an E&M interface
for each port. This allows to
connect directly to a phone, fax machine, key phone system or PBX.
It automatically detects whether the incoming call is a voice or
fax call. The single port supports
FXS and FXO interfaces, while the digital connects
directly to a PBX or PSTN line via T1/E1 or PRI.
Management. Bandwidth is used only when someone
is speaking. The silence suppression/Voice Activity Detection (VAD)
feature is an option that frees unused call bandwidth for data traffic.
This is significant, since callers are usually silent for 60 percent
of the call. When using silence suppression, also
offers Comfort Noise Generation (CNG) at the receiving end so the
user knows the line has not dropped. In addition, supports
voice compression standards like G.729 (8:1) and G.723 (10:1). These
standards help minimize the bandwidth required for voice. G.723,
for instance, is the maximum compression rate and requires only
5.3K bps (plus an added 7-8K bps for IP overhead). Even at maximum
compression, your VOIP solution will still provide toll-quality
is easily managed
locally using a windows-based software application or remotely by
the central office with a web browser or SNMP. also
includes its own SNMP management software which provides central
site configuration, management and call monitoring for all gateways
on the network. It utilizes a Windows interface that makes it easy
to view events like usage tracking, live use reporting, call history,
and voice quality statistics. In addition, it eases administration
by automatically e-mailing call logs based on volume or time.
User Training. provides
single stage dialing by utilizing a Uniform Dialing Plan that is
consistent with the E.164 (PSTN) standard numbering plan. This includes
automatic appending and stripping of digits to dialed numbers to
ensure that users will not require additional training to make VOIP
calls. In fact, placing calls with is
like using your existing phone system.
H.450 supplementary services to provide for call transfer, call
forwarding, call hold, call waiting, and name identification. It
also supports Q.SIG, an inter-PBX signaling protocol, for networking
PBX supplementary services in a multi- or uni-vendor environment.
In addition, supports
SIP extensions providing call forward and call transfer capabilities.
Small Office Media Gateway Solution. can
also provide an affordable small office media gateway solution that
delivers the features of Avaya™'s Communication Manager software
to the branch offices of large corporations. The gateway,
with integrated gatekeeper, cost-effectively extends the call features
and networking benefits of a centralized Avaya™ Media Server
to small branch offices, utilizing traditional analog devices, over
an IP infrastructure. also
renders local office survivability, in the case of a LAN or WAN
failure, by providing local, reliable PSTN trunking.