Black Box Multiport Analog VoIP Gateway Router VOE231A - Manual (Page 3)

3 of 5 1/11/2008
#26894
VoIP.
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, 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.
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.
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.
VoIP doesn’t interfere with other network services either,
so you can surf the Web while making a VoIP call.
Portability
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
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.
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.
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.
724-746-5500 blackbox.com
Technically Speaking