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Multiport Analog VoIP Gateway Routers
Connect up to eight voice/fax analog calls to VoIP.
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Technically Speaking
VoIP.
Portability
Voice over Internet Protocol (VoIP) is a cost-saving
alternative to traditional telephone service that enables voice
data to be transported over IP networks, like the Internet,
instead of the public switched telephone network (PSTN) or
a cellular network.
VoIP doesn’t tie you to one phone or to a single location.
Anywhere you find high-speed, reliable Internet access, you
can use VoIP. Your phone number stays the same wherever
you are—office, home, hotel, or even traveling overseas.
Standards
VoIP, which operates strictly over IP networks, can connect
to other VoIP nodes or traditional phone lines. The IP network
used may be the Internet or a private network.
In either instance, the actual data-transport portion of this
network can still be made up of the full gamut of network
services: high-speed leased lines, Frame Relay, ATM, DSL,
copper, fiber, wireless, satellite, and microwave signals. VoIP
simply digitizes voice data and adds it to other information
traveling along the same network.
With this technology, a phone call can be placed between
two PCs, between a PC and a standard telephone, between
a PC and an IP phone, between an IP phone and a standard
telephone, or between two IP phones. It will take a long time
for the PSTN to support this technology seamlessly, but this
seems to be the direction in which phone systems are headed.
Although the ITU standards for VoIP have evolved
significantly in the last few years, VoIP is still suffering from
a lack of generally accepted interoperability standards.
H.323, a standard for real-time audio, video, and data
communications across IP-based networks (including the
Internet), is almost universally accepted as the primary
standard for VoIP call setup and signaling. It’s actually a
collection of standards that works together for sending
multimedia and data over networks that don’t provide
guaranteed Quality of Service (QoS).
The H.323 standard includes:
Real-Time Transport Protocol (RTP) specifies end-to-end
network transport functions for applications transmitting real-
time data such as video. RTP provides services like payload
type identification, sequence numbering, time stamping, and
delivery monitoring to real-time applications. Plus, it works
with RTCP.
Real-time Transport Control Protocol (RTCP) works with
RTP to provide a feedback mechanism, providing QoS status
and control information to the streaming server.
Registration, Admission, Status (RAS) is a gateway protocol
that manages functions such as signaling, registration,
admissions, bandwidth changes, status, and disengage
procedures.
Benefits of VoIP
Because VoIP is inexpensive, has a worldwide reach, and
operates on a few simple principles, it’s exploded in popularity
recently—especially among both small and large businesses
that incur significant long-distance telephone expenses.
Savings
Without question, the primary benefit of a VoIP system
is decreasing or eliminating long-distance telephone charges.
Organizations with a high volume of long-distance voice
traffic stand to save quite a lot of money by implementing
a VoIP system. However, this factor alone may not warrant a
full commitment to VoIP for some companies.
Q.931 manages call setup and termination.
H.245 negotiates channel usage and capabilities.
H.235 provides security and authentication.
As VoIP product manufacturers began conducting
interoperability tests for more complex operations, they
recognized that they needed a simpler and more adaptable
standard for call handling and signaling protocol.
To this end, the IETF developed the Session Initiation
Protocol (SIP). SIP is built with less computer code than H.323
is, so it’s less cumbersome. Because SIP is similar in nature to
HTML—it uses ASCII text for configuration—users can adapt
it more easily for specific VoIP systems. In contrast, modifying
H.323 for VoIP applications requires a knowledgeable
computer programmer.
Setup fees for VoIP are usually quite low so your
organization can generally start saving money after only a
month or two of service. And with the wide variety of VoIP
products and services on the market, it’s easier than ever
to set up a VoIP phone system over your network.
Convenience
VoIP can be set up in a way that enables you to use phone
numbers in exactly the same way as you did before VoIP. Most
of the services you get with traditional phone service—Voice
Mail, Call Waiting, and Call Routing, for instance—are also
available with VoIP.
Both H.323 and SIP are considered “thick clients,” where
intelligence is maintained in the end devices such as IP
telephones. In this respect, H.323 has a head start, although
most VoIP systems today support both H.323 and SIP.
VoIP doesn’t interfere with other network services either,
so you can surf the Web while making a VoIP call.
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Regulation vagaries: Much of the government regulation
of VoIP is still being worked out. The U.S. government hasn’t
decided whether VoIP is going to be regulated as phone
service or whether to tax it. VoIP isn’t available in all countries.
Compatibility: Although older VoIP equipment may still
have some compatibility issues, current VoIP products from
different vendors generally work together.
Cost: For all the popular talk about VoIP being free, it
isn’t truly free. Any VoIP system has costs associated with its
implementation—equipment, high-speed Internet access,
and gateway service.
Providers
Despite the fact that VoIP standards are still developing,
providers are already flooding the market with products and
services while forming partnerships and matching expertise
to strengthen their position in this new market. The biggest
of these players and alliances—the ones who have the size
and experience to grasp technical issues and quickly build
infrastructures over which to offer VoIP services—are able
to keep up with (and often influence) the continual changes
in this market and keep rolling out new services.
Components
QoS: VoIP depends on having a fast, reliable network
to operate. A fast network connection with guaranteed
bandwidth is not a problem in a corporate intranet where
you have complete control over the network. However, if
you’re using the Internet for VoIP, you’re using a public
network that may be subject to slowdowns that cause drop-
outs and distortion. You may find that your high-speed
Internet connection is faster than the actual Internet and that
the quality of your connection is generally unacceptable or is
unacceptable at times when Internet usage is high.
A VoIP system depends on devices that connect your
traditional phone or phone system to an IP network.
Components that you’ll see in a VoIP system include:
• End-user devices
• Gateways or gatekeepers
• IPBXs
• IP Networks
End-user devices are usually VoIP telephones or PCs
running VoIP software. End-user devices have their own IP
address and make a direct connection to the IP network.
A gateway is a device that converts circuit-switched analog
voice calls from a traditional PBX into VoIP packets and
transmits them over an IP network either to another gateway
or directly to an end-user device.
A gateway can have additional features such as voice
compression, echo cancellation, and packet prioritization.
Because VoIP-enabled end-user devices can communicate
directly with each other over an IP network, a gateway is not
a required component of a VoIP system as long as the VoIP
devices are connected directly to the IP network.
An IPBX is a PBX with a built-in gateway. IPBX systems are
equipped for hundreds of telephone ports, with WAN support
for trunk connections to the PSTN, and with high-speed IP
WAN links. In addition to VoIP features, these systems usually
include other features typical of traditional PBX systems such
as music on hold, auto-attendant, and call management.
Often, they include Ethernet ports to support VoIP
telephones.
Emergency services
If you subscribe to a VoIP gateway service that enables you
to use your VoIP phone like a regular phone, be aware that
you may not be able to call 911 for emergencies. If 911 service
is important to you because you don’t have an alternative way
to call 911, shop for a VoIP provider who provides this service.
Consider, too, that VoIP needs both working Internet
access and power to work. If you lose your Internet service,
your phone goes, too. And, unlike regular phone service that
can keep basic telephones working when the power goes out,
VoIP needs power—if you lose power, you lose your phone.
Moving forward
Before VoIP technology becomes truly universal, the
current worldwide PSTN will have to migrate to a packet-
based IP equivalent. Industry inertia alone dictates this will
not occur instantly. The current worldwide PSTN system has
grown to what it is over a period of 125 years. Given the sheer
complexity of the existing PSTN, the migration to an IP packet
network will probably occur during several decades.
As migration from the PSTN to IP-based networks
VoIP can be set up with or without a connection to
standard PSTN phone service. You can, of course, place calls
over the Internet directly from your PC or IP phone to another
VoIP-enabled device. But what makes VoIP so versatile is that,
through the use of a gateway service, it can also be used to
call the numbers of phones connected to standard land-line
or cellular phone services. They can also receive calls from
standard telephones.
proceeds, businesses and home users will gradually discover
reasons of their own to implement VoIP. It won’t happen right
away, but we predict that VoIP will become a big part of
telecommunications in the not-so-distant future.
Although it’s not quite as convenient as conventional
phone service, VoIP can offer serious savings—particularly if
you now regularly pay for multiple overseas phone calls. Keep
in mind though, VoIP isn’t a one-size-fits-all solution. But with
a little planning, VoIP could spell savings for you!
Not all fun and free calls
There are still things to consider when you’re deciding
whether or not to invest in VoIP.
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TECH SPECS
1
Voice Services — Anonymous Caller ID block;
Call blocking;
Call forward on busy;
Call forward, selective;
Call forward, unconditional;
Call hold/retrieve;
Call return;
Call transfer, blind;
Call transfer, with consultation;
Call waiting/retrieval;
Caller ID;
Conference drop;
Conferencing (3-way calling);
Distinctive ring;
Do not disturb;
Hotline calling;
Incoming Caller ID on/off;
IP URL dialing;
Message waiting indication;
Self Caller ID block;
Speed dial;
Voicemail message retrieval;
Warmline calling
Caller ID Type — FSX Ports: ⁄ FSK, ITU V.23/Bell 202 generation;
2
FXO Ports: Caller ID FSK CLI reception and relay (Bellcore/ANSI and
ETSI/ITU), Call routing based on Caller ID
Call Routing —
Virtual interfaces;
Routing criteria: Called party number (Destination), Calling
party number (source), time of day, day of week, date, longest prefix
match, wildcard match, regular expression match;
Number manipulation functions: Replace numbers, add/remove
digits, regular Expressions;
Fallback routing: Soft fallback to alternative interface or
call router table
Environmental — Operating temperature: 32–104° F (0–40° C);
Operating humidity: 5–80% (noncondensing)
Fax and Modem Support — G.711 fax and modem bypass;
T.38 Fax relay (9600 bps, 14.4 kbps)
IP Services — IPv4 router;
Static routes, ICMP redirect (RFC 792), RIPv1, v2 (RFC 1058 and 2453);
Static and dynamic NAT and NAPT;
DHCP server and client;
Access control lists;
VOE231A: rear view
IPSEC AH & ESP modes, preshared keys;
AES/DES/3DES encryption
Management — Web GUI;
CE Approval — Yes
Connectors — All: (2) RJ-45 10-/100-Mbps Ethernet;
VOE231A: (2) RJ-11 FXS;
VOE232A: (4) RJ-11 FXS;
VOE233A: (6) RJ-11 FXS;
VOE234A: (8) RJ-11 FXS;
VOE235A: (2) RJ-11 FXS, (2) RJ-11 FXO;
VOE236A: (4) RJ-11 FXS, (2) RJ-11 FXO;
VOE238A: (4) RJ-11 FXS, (4) RJ-11 FXO
Indicators — LEDs: (1) Power, (1) Run (1) VoIP Link, (2) Link, (2) 100M,
(2) Activity, (2–8) Voice Port
Power — 100–240 VAC, 50–60 Hz, external
Size — 2.5"H x 8.5"W x 7.1"D (6.4 x 21.6 x 18 cm)
Weight — 1 lb. (0.5 kg)
Industry standard CLI with local console (RJ-45, RS-232) and remote
Telnet access;
TFTP configuration and firmware loading;
SNMP v1 agent (MIB II and private MIB);
Built-in diagnostic tools (trace, debug)
QoS — Traffic classification by ACL;
TOS and DiffServ labeling, configurable TOS/Precedence bits
or DiffServ codepoints;
IEEE 802.1p/Q;
Traffic scheduling: Priority, weighted fair queuing (WFQ), hierarchical
traffic classes;
Policing of traffic classes;
™
DownStreamQoS dynamic restriction of inbound (downstream)
TCP traffic to free bandwidth for voice packets;
Improves voice quality in the receiving direction
Voice Processing — Voice codes: G.711 A-Law/µ-Law (64 kbps),
G.726 (ADPCM 40, 32, 24, 16 kbps), G.723.1 (5.3 or 6.3 kbps),
G.729ab (8 kbps);
Up to 8 parallel voice connections;
G.168 echo cancellation;
Carrier tone detection and generation;
Silence suppression and comfort noise;
Configurable dejitter buffer;
Configurable tones (dial, ringing, busy);
RTP/RTCP (RFC 1889)
Item
Code
Multiport VoIP Gateway Routers
2-Port FXS
VOE231A
VOE232A
VOE233A
VOE234A
VOE235A
VOE236A
VOE238A
4-Port FXS
6-Port FXS
8-Port FXS
2-Port FXS, 2-Port FXO
4-Port FXS, 2-Port FXO
4-Port FXS, 4-Port FXO
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