psqm scores:
The lower the better.
An independent product review conducted by
Internet Telephony magazine (March 1999).
Save thousands of dollars each month
MultiVOIP can save your company substantial amounts in long distance
charges. Even if your company uses one of the most inexpensive calling
plans, a MultiVOIP network can quickly return your investment and begin
paying you back.
long distance minutes/ MultiVOIP
locations MultiVOIP cost cost/minute line/day payback
Central Site/ $1999 $0.04 90 139 days
California mvp410 (4 lines)
Remote Site/ $1099 $0.06 60 153 days
Chicago mvp210 (2 lines)
Partner Site/ $1099 $0.08 60 115 days
United Kingdom mvp210 (2 lines)
Easy integration
With MultiVOIP, you avoid the hassle and expense of replacing your existing
routers, WAN connections or phone systems required by other VOIP
solutions. MultiVOIP simply plugs into your Ethernet network. Neither
your phone service or network is placed at risk. Minimum requirements:
• Ethernet network
• WA N connection
• IP addresses
Award-winning voice quality
With MultiVOIP, you’ll experience
consistent toll-quality voice connections.
Using the Perceptual Speech
Quality
Measurement (PSQM),
Internet Telephony magazine found
that MultiVOIP delivered exceptional
voice quality
.In
fact, MultiVOIP
outranked the competition.
Interoperability
MultiVOIP utilizes the 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, MultiVOIP also supports T.38 real-time
fax relay for interoperability among other VOIP epuipment.
PSTN fail-over
PSTN fail-over allows MultiVOIP 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 under-
mine voice quality. Utilizing user definable controls, MultiVOIP continually
checks if the LAN/WAN is threatened by packet loss, jitter or latency, or to see if
the network is completely down. If it detects a problem, MultiVOIP switches to
“survivability mode” transparently routing all calls over PSTN lines connected
to the MultiVOIP gateway. MultiVOIP continues to monitor the connection and
automatically switches back to the LAN/WAN once the conditions improve.
Advanced speech technologies
MultiVOIP 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 cancella-
tion, forward error correction, bad frame interpolation, tunable latency and
dynamic jitter buffers, further enhance voice quality.
Complete support for multiple telephony interfaces
For maximum investment protection, the MultiVOIP 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
MultiVOIP 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 MultiVOIP supports FXS and FXO interfaces, while the digital
MultiVOIP connects directly to a PBX or PSTN line via T1, E1 or PRI.
Bandwidth management
Bandwidth is used only when someone is speaking. The silence suppression/
Voice Activity Detection (VAD) feature is an option that frees unused call band-
width for data traffic. This is significant, since callers are usually silent for
60
percent of a call. When using silence suppression, MultiVOIP also offers
Comfort Noise Generation (CNG) at the receiving end so the user knows the line
has not dropped. In addition, MultiVOIP 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 voice.
No user training required
MultiVOIP 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 MultiVOIP is like using your existing phone system.
toll-free voice/fax communications
pstn voice quality
connects directly to phones, fax or pbx
turnkey solution
PBX trunk Phone/Fax or PBX extensions