Voice over IP (VoIP)

An umbrella term for a set of technologies that allow voice traffic to
be carried over Internet Protocol (IP) networks such as the Internet.

Overview

Voice over IP (VoIP) is one of the driving forces behind convergence
in the networking and telecommunications industry. The term
convergence refers to the goal of combining all forms of enterprise
communications (voice telephony, fax, and data) into a single IP data
stream and managing this through a single integrated system. The goal
of convergence is twofold:

. To reduce costs by eliminating redundant infrastructures.  For
example, the typical corporate office building has two separate wiring
infrastructures: voice telephony wiring and Category 5 (Cat5) or
enhanced Category 5 (Cat5e) twisted-pair network wiring. Similarly,
large enterprises spanning several geographical locations often have
two communication infrastructures as well: Private Branch Exchanges
(PBXs) linked by dedicated tie lines and leased lines for wide area
networking (WAN). VoIP enables an enterprise to consolidate these two
systems into a single communications infrastructure running on an IP
data network, with resultant cost savings.

. To simplify management of all forms of business communication (such
as e-mail, voice mail, and faxes) by consolidating all messages into a
single user interface, typically in the user.s e-mail Inbox.

To achieve these two goals of convergence, VoIP addresses the first
issue and unified messaging (UM) addresses the second. These systems
may be deployed separately or together in the enterprise, depending on
business needs and goals.

Advantages and Disadvantages

Although VoIP promises to save enterprises money by avoiding costly
long-distance toll charges, the expertise needed to implement VoIP and
the cost of new equipment is often seen as a barrier to embracing this
new technology. Other concerns often voiced by network managers
include

. Latency: Because of the best effort nature of IP communications,
voice communications sometimes sound garbled and contain undesirable
pauses due to high jitter and latency. This is probably the number one
issue that companies consider when choosing a VoIP solution. The
problem is especially bad when IP traffic is routed over the public
Internet, where VoIP conversations can sound like old CB radio
sessions in quality. Improvements in IP quality of service (QoS)
discussed below are reducing the importance of this issue, but these
IP QoS solutions are complex and difficult to implement.

. Interoperability: Because there are competing VoIP standards and
vendors have developed their own proprietary solutions, when a company
adopts a VoIP plan it must essentially use equipment from only one
vendor. Even simple IP phones with RJ-45 Ethernet jacks from one
vendor usually do not work with VoIP systems from another vendor. In
contrast, traditional PBXs and digital telephones from different
vendors interoperate seamlessly. If a dominant VoIP standard emerges
over the next few years, however, this interoperability problem might
finally be resolved.

. Cost: Companies that have already invested heavily in legacy PBXs,
digital phone systems, and .dumb.  routers might be reluctant to
abandon them to deploy all-new VoIP switching, routing, telephones,
and call control equipment. The cost of the expertise needed to
implement and maintain VoIP solutions must also be considered.

Despite these concerns, VoIP is encroaching steadily into large
companies and will probably overtake the legacy PBX market in the next
five years if trends continue.

Architecture

Most VoIP solutions developed by different vendors are still
proprietary solutions that require all customer premises equipment
(CPE) to be purchased or leased from a single vendor, but recently
there have been moves toward standards-based VoIP solutions that
should eventually allow equipment from different vendors to
interoperate in a plug and play (PnP) fashion.  The main problem at
this point is that competing standards have evolved in the VoIP arena,
and it is still unclear which standards will dominate in the emerging
VoIP industry. The main standards and protocols of relevance to VoIP
are:

. H.323: The earliest VoIP systems were based on the H.323 and T.120
of protocols developed by the International Telecommunication Union
(ITU) for audio, video, and data communication and conferencing over
IP networks. H.323 works well but is complex to implement.a typical
H.323 VoIP system includes end-user stations known as VoIP terminals,
gateways for translating between IP networks and telephony networks,
gatekeepers for handling calling functions, and multipoint control
units (MCUs) for handling multipoint conferencing.  In addition to its
complexity, H.323 also suffers from considerable protocol overhead
because it was originally designed to support video communications.
Nevertheless, most VoIP vendors have built their systems around H.323,
and it is still considered the primary standard for interoperability
between different VoIP systems.

. Session Initiation Protocol (SIP): This is an application- layer
control protocol defined in RFC 2543 that is designed to allow
different telephony, data networking, and VoIP equipment to
communicate with one another. SIP is a lightweight protocol with much
less overhead than H.323 and is seen by some industry analysts as the
logical successor to H.323.

. Media Gateway Control Protocol (MGCP): This protocol is defined in
RFC 2705 and is designed to translate between the voice traffic
carried on the Public Switched Telephone Network (PSTN) and IP traffic
carried on the Internet. MGCP is a master/ slave protocol that enables
media gateway controllers and media gateways to communicate with each
other and to control IP telephony terminal equipment.  MGCP has been
implemented in some VoIP systems but is being phased out in favor of
the newer Megaco/H.248 standard.

. Megaco/H.248: This protocol is a joint effort of the Megaco working
group of the Internet Engineering Task Force (IETF) and the ITU-T
Study Group 16.  Megaco/H.248 is the successor to MGCP and includes
additional features that support peer-topeer communication and
simplified implementation of media gateways and controllers.

For more information on the different VoIP protocols, see the
individual entries about them elsewhere in this book.

Implementation

VoIP is all about saving cost for companies, and there are a variety
of ways in which it can be implemented using the various architectures
outlined previously. One important scenario is for linking offices to
eliminate long-distance calling costs. The traditional way for
companies to do this has been to connect the PBXs in two offices using
tie lines. A tie line is a dedicated connection used for both
signaling and call transfer between PBXs. This approach is called a
.toll bypass.  solution because it bypasses the tolls levied for
traditional long-distance traffic, in effect emulating a .tollfree.
solution.

Tie lines are usually implemented over T1 lines. For offices
relatively near each other, the monthly cost of leasing T1 lines for
this purpose can be less than the accumulated cost of a month.s
long-distance calls between the offices, but when the distance between
offices is great, the T1 line costs become excessive as they are
billed by the mile for most U.S. telcos. Another problem is that a
separate tie line must be deployed between each pair of offices.calls
cannot be routed across multiple call lines.

VoIP can save costs in this situation by routing voice traffic over
existing data lines between offices, also typically T1 lines for large
companies. This means instead of needing two T1 lines between offices
(one for data and one for tying PBXs together) companies need only one
line (voice and data combined). In addition, companies do not need T1
lines between every pair of offices (full mesh topology), as IP
packets can be routed across the enterprise provided there is at least
one path between each pair of offices (partial mesh).

A more general VoIP solution is to replace existing PBXs with
VoIP-enabled routers that perform gateway and gatekeeper functions to
handle call setup and routing.  One hidden cost in this
.all-IP. approach is that traditional digital phones costing about $50
must be replaced by special IP phones often costing hundreds of
dollars. This approach to VoIP, championed by Cisco Systems, is
sometimes called an IP PBX approach since it completely replaces
existing legacy PBX systems.  Companies that have invested heavily in
legacy PBX equipment may be reluctant to follow this route also and
may instead prefer to use VoIP for replacing PBX tie lines to continue
leveraging their investment in legacy equipment. Alternatively, some
VoIP vendors such as Nortel Networks offer a VoIP gateway approach
that let the legacy PBXs continue to handle call processing while
line-provisioning is performed by specialized VoIP routers and
servers.

Marketplace

Cisco has been the dominant player in the high-end enterprise VoIP
market for some time. In fact, Cisco itself uses VoIP across its
company, and with more than 10,000 IP phones deployed around the
organization, this is still probably one of the largest and most
successful VoIP implementations around. Cisco has developed a special
architecture for VoIP solutions called Architecture for Voice, Video,
and Integrated Data (AVVID) that consisting of three layers: switching
and routing infrastructure for routing IP traffic, applications such
as Cisco Call Manager running Microsoft Windows 2000 servers for
implementing call control, and wired or wireless IP telephones. In the
infrastructure area, for example, Cisco 2600, 3600, and AS5300 routers
and Cisco Catalyst 6009 switches all support the H.323 standards and
can function as H.323 gateways and gatekeepers for building highly
scalable VoIP solutions.  Cisco.s AVVID architecture is
straightforward in concept and can effectively scale to more than
100,000 users to meet the needs of the largest enterprise.  The
approach taken by Avaya, which, together with Nortel, represents the
two dominant players in the legacy PBX market in the United States, is
adapted specifically to its own proprietary line of PBX equipment, and
scales to 10,000 users for large VoIP deployments. The advantage here
is that Avaya offers a clear upgrade path from using its legacy PBXs
to move to an all-IP telephony solution. Nortel.s Call Pilot solution
also supports up to 10,000 users and is popular in Europe but just
emerging in the U.S. market. Another large player is Alcatel, whose
VoIP solution supports legacy PBX connections and whose call control
applications run on UNIX platforms. Alcatel even offers an IP
Telephony Starter Kit to enable small companies to quickly and
painlessly roll out a VoIP solution. This kit includes an Alcatel
OmniStack 6024 Ethernet switch, an OmniPXC 4400 IP PBX, and 10 IP
Reflex telephones.

Some larger telecommunication carriers such as AT&T and WorldCom are
now offering their own VoIP solutions using Cisco 2600 and 3600
routers deployed at the customer premises in a VoIP gateway
approach. AT&T integrates its VoIP solution with Internet access and
virtual private network (VPN) support to provide an allin- one
communications infrastructure solution for its customers.

Notes

Despite the great attention garnered by VoIP in the press, other
technologies also can serve well for various needs in the enterprise,
namely:

. Voice over frame relay (VoFR): Standards for this technology are
more mature than for VoIP, and VoFR is a good option to consider for
replacing tie lines between PBXs in branch offices. Instead of
connecting branch offices in pairs using dedicated point-to-point T1
lines, frame relay assembler/ disassemblers (FRADs) are used to
connect all the offices using the carrier.s frame relay cloud. VoFR
then multiplexes voice traffic and compresses them into packets and
routes them between each pair of offices using permanent virtual
circuits (PVCs) preconfigured by the carrier to simulate point-topoint
links.

. Voice over High-level Data Link Control (VoHDLC): This is another
good solution for pointto- point connections, and, like VoFR, it
supports multiplexing and compression of voice traffic. The downside
here is that HDLC is a Cisco proprietary Layer 2 protocol, so
implementing it means you have to go with only one vendor.

The main advantage VoIP has over both of these alternative
technologies is scalability.both VoFR and VoHDLC require that voice
traffic be encoded and decoded multiple times in transit, and this
adds overhead that limits the scalability of these technologies. In
contrast, VoIP can be used on IP networks of any size, including the
largest of networks, the Internet. The downside of VoIP, of course, is
that IP is a best effort packet delivery service with no intrinsic
quality of service (QoS). As a result, the quality of voice
communications can be poor when using VoIP, especially over a large
network such as the Internet, and can result in unacceptable levels of
latency, jitter, and dropouts. However, by employing special standards
and protocols such as 802.1p, 802.1Q, Resource Reservation Protocol
(RSVP), and Multiprotocol Label Switching (MPLS), QoS can be added to
IP to the point where voice quality approaches that offered by VoFR,
VoHDSL, or even the PSTN.

Another alternative that is used by only a few large companies is
Voice over Asynchronous Transfer Mode (VoATM), which provides
excellent voice quality since ATM technology has built-in support for
QoS. The problem is that to implement VoATM you already need an
underlying ATM network connecting branch offices, and only a limited
number of large enterprises have implemented ATM across the WAN due to
the high cost and complexity of this solution.

An emerging technology gaining a lot of attention is Voice over
Digital Subscriber Line (VoDSL), which enables carriers to provision
multichannel voice and data communications over Symmetric Digital
Subscriber Line (SDSL). Since SDSL is relatively cheap compared to T1
lines, this is an attractive option for certain IP telephony
solutions. A typical VoDSL solution involves placing an integrated
access device (IAD) at the customer premises, which is connected to
both telephone equipment and the computer network. Most VoDSL IADs
currently support only 16 voice channels, so this solution is
currently marketed mainly toward small and mid-sized businesses. One
difficulty is that VoDSL works only with SDSL and does not support the
more commonly available Asymmetric Digital Subscriber Line (ADSL)
technology. This limits its availability in some areas, but this will
still be a technology to watch in the next few years. Vendors of
first-generation VoDSL equipment include Copper Mountain Networks,
Jetstream Communications, and Tollbridge Technologies.

See Also: 802.1p, 802.1Q, Asynchronous Transfer Mode (ATM), Digital
Subscriber Line (DSL), frame relay, Frame Relay Access Device (FRAD),
H.323, High-level Data Link Control (HDLC), Internet Protocol (IP), IP
PBX, IP telephony, jitter, latency, Megaco, Multiprotocol Label
Switching (MPLS), permanent virtual circuit (PVC), Private Branch
Exchange (PBX), Public Switched Telephone Network (PSTN), Resource
Reservation Protocol (RSVP), session initiation protocol (SIP),
Unified Messaging (UM)