Voice over Packet (VoP), telecommunications using high-density media
gateways, is attracting enormous attention and investment. As standard phone
service migrates from the continuous-circuit public switched telephone
network (PSTN), to a packet-switched VoP technology, explaining both the
need for this conversion and its complexities from both a technical and a
financial perspective presents challenges to service providers.
From a user's standpoint, placing a VoP call that travels over a private
network or even the Internet is not all that different from placing a
traditional call that travels over the PSTN. In fact, that lack of perceived
difference is critical to the success in ultimately converting to a
packetized approach. Customers expect seamless quality of service (QoS) to
accompany successful VoP deployment. That means whenever VoP technology
replaces the PSTN and does not offer a new service or convenience, such as
the mobility of Cellular phones, the quality of VoP system service must
closely approximate that of the PSTN such that service quality and
reliability are transparent to the end user. Only by delivering the
appropriate level of quality will manufacturers and service providers
accomplish their goal of deploying voice access over a packetized network.
And that won't be easy. Because even though call transmission may be
transparent to the user, from an equipment and routing standpoint, VoP call
placement, setup, and transmission is quite different and complex. Routing a
call over a network, or over the Internet, requires the use of extensive
network equipment and software resources; and those resources are being
deployed incrementally, in part because QoS is paramount. As a result,
service providers are installing hybrid systems that utilize both
approaches.
In the near term, the "vision" of VoP systems will be supported
by these hybrid deployments that use the PSTN to provide worldwide access to
the end user -- while packetized, VoP infrastructure is developed for
long-haul services and competitive local exchange carrier (CLEC) access to
end users with the assumption that these hybrid systems will eventually
convert to full VoP deployment. Unfortunately, the implementation of total
VoP systems is impeded by today's infrastructure limitations and by the
economics that impact service providers. For example, although the
accessibility of the Internet is growing rapidly, the absence of substantive
QoS controls and standards deter user adoption of voice over Internet
protocol (VoIP) services. Because the voice quality of VoIP is inconsistent
and often unreliable, VoIP systems are today deployed primarily in limited,
private networks, or campus and enterprise-based local area network (LAN)
systems such as the IP PBX. These systems provide the advantage of a single
LAN connection for both voice and data services, but voice calls almost
always connect back to a circuit-switched PSTN for off-campus connections.
Deregulation, compounded with the proliferation of service suppliers,
adds to the mix because as suppliers announce new opportunities and
services, it is clear that there is no "one size fits all" VoP
solution. To make it even more complicated, whole new technologies that
converge voice and data are being implemented across the communication
infrastructure. For example, voice and data packets are transported through
trunking gateways, or high-capacity multi-channel access concentrators. They
provide an essential single point of access by converting voice signals from
the PSTN to packetized data and vice-versa. As long as a conversion is
required gateways are necessary regardless of the subscriber's connection
type such as the public phone system, a mobile phone, VoIP, VoCable, VoDSL,
PBX, or other system. Trunking gateways are designed to process thousands of
calls at a time, connecting users, campuses/workgroups, and customer
premises sites.
It is clear that a total conversion to a VoP infrastructure will
ultimately prevail because of the imperatives of the inherent economic
advantages. However, some of the factors that contribute to inconsistent and
incremental deployment must be overcome. These include variations in network
service capabilities, access and capacity limitations, and limited voice
quality due to disruptive network characteristics such as latency and packet
loss. Finally, interoperability between manufacturers, and the need for
consistent standards for product development and deployment, must continue
to be addressed. That cooperation has accelerated as service providers and
operators perceive the opportunity for new revenues. Nevertheless, when
infrastructure, common access, and performance issues are resolved, industry
experts believe rapid and widespread deployment and adoption of integrated
networks based on common standards will prevail.
A VoP call is defined as any call in which any part of the communication
is carried over a packet network. Although many PSTN or circuit-switched
calls are digitized and converted from analog to digital, the signal is sent
as a continuous stream of bits at a rate of 64 kilobits per second. For a
packetized network call the key criterion requires fixed or variable length
packets of bits sent over the network, not the continuous stream.
So, what happens when someone makes a VoP call? The process of sending
voice over a network (or sending voice packets over a network) is compatible
with using an Internet service provider (ISP) to transmit voice packets. The
packet flow begins after the call is placed. There are many variations, but
one scenario starts with a VoIP call through a local-area network (LAN) or
IP PBX originating at the end-users premises. The call is routed via a local
packet network, often DSL or frame relay, as a series of packets to the
local exchange carrier's central office. The central office is the user's
access point to the PSTN. At this point the voice packets may be converted
via a trunking gateway into continuous pulse code modulation (PCM) bit
stream for connection into the PSTN.
The call then travels through the digital trunk lines to the Internet
service provider's (ISP) point of presence (POP). Digital trunk lines are
the physical (wire) and logical connections between circuit switches across
which telephone network travels in PCM form. The trunk lines terminate in
the trunking gateway, or Internet gateway processor, that connects to an
ISP's packet network. The trunking gateway compresses the voice data and
creates network packets that are transmitted into the packet network or
Internet.
Once on the packet network, the call is routed to the end destination
premise gateway through a virtual circuit. A virtual circuit is a
communications link that appears to the user as a dedicated point-to-point
circuit. In reality, the packetized data can be sent over different physical
paths through a network to its destination. The virtual circuit determines
where the packetized speech for the specific phone call should be routed and
enables the receiving gateway to verify receipt, enable packet re-order,
identify, and correct discrepancies. In fact, virtual circuits illustrate
the key benefits of packet-based networks, and they are cheaper and faster.
In the receiving gateway, packets are converted back to a continuous stream
of data and connection with the local PSTN for transmission to the receiver.
There are variations to this theme depending upon the nature of both the
sending and receiving networks. In fact, even part of a virtual circuit may
require interim conversion to a circuit-switched system where the packets
are converted to a continuous stream and subsequently re-packetized. Again,
the system employs gateways to enable the conversions.
An interesting variation is with calls through the use of wireless
networks. Most wireless networks use digitized systems similar to speech
packets in a VoP call. However, the wireless networks use a variety of
protocols to compress speech, one of the essential steps in the packet
creation process. As a result, the call first has to route through a
transcoder and rate adaption unit (TRAU) that converts speech from the user
of a wireless system into the digital code required by the IP-based network
compatible with standards such as H.323. This adds both additional
complexity of equipment and can increase latency, because speech compression
decoding and encoding incurs additional latency.
Latency is a problem inherent in VoP systems and is primarily caused by
network delay or by the functions that occur within the gateway. However,
any additional latency due to transcoding is a problem. Consequently efforts
are underway to enable networks to enable a common IP-based protocol across
wireless and wireline systems.
Finally, the packetization process in which real-time speech and
real-time control protocols convert, sequence, and then transmit packets to
the network must also enable the signal to be reconstructed before it
reaches the receiving user. These packets constantly enter and exit the
processing components, and arrive delayed, distorted, and out of sequence,
far different from PSTN continuous bit-stream systems. Hardware and software
at the users end gateway detects and processes the voice signals,
compensates for distortion and provides the call routing. The end result is
these hybrid elements combine to work with the Internet protocol (or other
network-based protocol) and the PSTN to deliver the phone call transparently
to the user.
Scott Robertson is product line manager for Remote Access Products
at Analog Devices. Analog Devices
is a semiconductor company that develops, manufactures, and markets
high-performance integrated circuits (ICs) used in signal-processing
applications.
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