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IP Telephony
Technology Evaluation Guide

Copyright © 2001
Mier Communications, Inc.

All rights reserved.

Reproduction in any form,
including dissemination electronically,
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 IP PBXs: You’ve Got Questions.
We’ve Got Answers.
The first commercial deployment of IP telephony –
most commonly known as voice-over-IP (or VoIP) – was in
backbone transmission, via VoIP gateways from vendors
including Cisco Systems and Lucent Technologies
(through Ascend Communications, formerly Lucent). This
began in earnest in 1998, with gateways initially handling
one or two T1 lines of call volume. Today, just a few years
later, high-end VoIP gateways for carriers can handle mul-
tiple T3s (each 28 T1s) of call volume.

This same technical migration is now occurring with

Private Branch Exchange (PBX) systems. With near univer-
sal agreement that telephony will be carried within IP pack-
ets in the future, vendors are preparing their next-genera-
tion PBXs with features and functionality that will position
these products as competitors to “classical” or “tradition-
al” PBX systems.

The rollout of VoIP seems to be following the same

pattern as the last major evolutionary change in the Public
Switched Telephone Network (PSTN) – the analog-to-
digital conversion – which began in the 1970’s. It was
found that by digitizing real-time voice streams, the num-
ber of concurrent calls that could be carried over a copper
transmission facility (multiplexing over a T1) could be
increased tenfold.

VoIP is now also emerging as the basis for telephone

switching equipment. In the PSTN/central office arena, a
dozen vendors are working diligently on VoIP-based
replacements to traditional Class 4 (“toll” switch) and
Class 5 (end office) CO switches. The same technical
migration is now occurring with PBXs.

©2001 Miercom
1
 Traditional PBXs have all relied on a TDM “switching
matrix” (in some cases called a “switching bus”) as the
fabric for establishing “circuit-switched” connections. And
with few exceptions (Cisco Systems and Sphere
Communications are examples), most IP PBXs on the mar-
ket today still employ an integral switching matrix. In the
Avaya and Alcatel architectures, VoIP calls pass through
the switching matrix only if they need to connect with ana-
log or digital stations, or with analog or digital trunks.
IP PBXs have to contend with three marketplace reali-
ties. First, traditional digital PBXs represent more than a
decade of technical maturity, and so, for the most part,
they have earned a reputation for delivering incredibly
high levels of reliability and system up time. This is
ingrained in the minds of many telecommunications man-
agers, who question whether IP PBXs today can deliver the
same high level of non-stop reliability.
The second point is that IP PBXs are still new in the
marketplace, and there has not been sufficient deployment
for long-enough periods of time to measure their relative
up-time or reliability, or to earn them familiar levels of
reliability like “four nines” or “five nines.” (Four nines
means uninterrupted operation for 99.99 percent of the
time; five nines means 99.999 percent up time.) Despite
these challenges, organizations such as Miercom are
beginning the process of lab testing and field reliability
studies which range from six months to two years.
The last point regarding IP PBX reliability is that addi-
tional components are required in the IP-telephony envi-
ronment, such as LAN switches and IP routers, which are
not involved in traditional-PBX networks. This is an added
layer of complexity, which may not even be related to the
©2001 Miercom
2 3
IP PBX equipment, but which nevertheless can result in the
“phone system” going down if a major component at this
layer fails.
IP PBXs have indeed come a long way during the past
eighteen months. Growing from 50- to 100-station systems
in their infancy, IP PBXs can now scale to 10,000 stations
and more according to research we have recently conduct-
ed at Miercom.
During the past two years Miercom has been a pioneer
in developing test methodologies and conducting in-depth
testing of products, such as VoIP gateways and IP PBXs,
which are the powerful new technologies for real-time
voice communications. During this time, Miercom has
also conducted detailed surveys of both markets, which
included “on- and off-the-record interviews” with key
vendors and end users who are actually implementing
VoIP in their networks.
This powerful combination of “hands-on” testing of
voice-over-IP products, including IP PBXs, and in-depth
research of the market, vendors and end users provides
Miercom with a unique perspective on this exciting new
technology.
On the following pages is a “Question and Answer”
summary of IP PBXs: what they are, what they do, and
what to look for when considering acquiring one for your
organization. This information provides the “big picture”
about IP PBXs and their role in the new-technology world
of VoIP.
©2001 Miercom
 IP PBXs: Q and A

Q: What is an “IP PBX?”

A: Basically an IP PBX - also known as an IP telephony sys-

tem, an IP communications server, or by various other
names – is a Private Branch Exchange (PBX) telephone
switching system that supports VoIP.

Q: What benefits does PBX offer over

“classical” PBXs?
A: Having an IP PBX allows the end user to mix voice, data
and video in one system, and is referred to as “conver-
gence.” This provides the potential for a number of appli-
cations, including the ability to make “phone calls” over a
PC. Unified messaging -the ability to store “voice” mails as
emails, answer phone calls with emails, etc. – is another
popular application. Convergence also effects videophone
service in which the caller and called can see one another
during the call.

Q: How do IP PBXs differ from their “classical”

or“traditional” PBX cousins?

A: Classical” or “traditional” PBXs are based on fixed-

bandwidth, time-division multiplexed switch fabrics, while

©2001 Miercom ©2001 Miercom

4 5
 “native” IP PBXs are based on IP-based “distributed”
architectures in which all information is transmitted in
packets. A number of “traditional” PBX vendors have “IP-
enabled” their traditional PBX with gateways that can han-
dle VoIP transmission. However, these products are still
based on the “traditional” circuit-switched architecture,
and VoIP on these PBXs is an “added-on” feature.
Q: What is voice-over-IP (VoIP)?
A: The original voice transmissions were over analog
lines, but in the 1970’s it was found that by digitizing real-
time voice streams (via pulse-code modulation and time-
division multiplexing), the number of concurrent calls that
could be sent over a copper “wire” could increase tenfold.
Using VoIP, voice streams are converted into “packets”
that are then sent over IP links. What this means is that
voice and data (and also video streams) can now all be
sent over the same network – one based on IP. This could
be the Internet or a private IP network.
Q: How many companies are actually using IP PBXs?
A: According to most estimates, about 20 percent of busi-
nesses in the U.S. are currently implementing some type
of VoIP application. International Data Corp. (IDC) reports
that by the end of 2000, there were about 5,800 “LAN-
CBX” systems installed. (“LAN CBX” is IDC’s term for IP
and VoIP-oriented systems.) They project that the number
of such systems will increase tenfold by the end of 2002.
©2001 Miercom
6 7
These projections are in line with others we’ve seen during
the past few months.
Q: Are IP PBX architectures basically the same?
A: No, in fact no two IP PBXs are even very similar.
Products differ in a number of ways, including the
following:
• The way they handle call control (how the system
sets up calls) and whether VoIP support is native or
an ìadd-onî subsystem to a traditional PBX;
• Whether they house their modules in a multi-slot
chassis (or shelf/rack) or whether self-contained
modules are distributed and IP-interconnected;
• Station capacity (ranging from 84 maximum
stations up to 15,000) and scalability;
• Support for IP phones and ìsoftî phones (some han-
dle both; some handle one or the other; a couple of
IP PBX support neither).
On the following page are two illustrations of architecture.
The first is the conceptual topology for Cisco’s AVVID and
CallManager systems; the second is Shorelines’s IP Voice
Communications System. Each Vendor has its own unique
architecture.
©2001 Miercom
 Conceptual Topology: Q: How is the current IP PBX market segmented?
Cisco AVVID and CallManager
Multi-Slot Chassis
A: According to our market analysis (based on primary
Analog
(i.e., Cisco Cat 6509) research conducted by Miercom), the IP PBX market is seg-
Phone mented into low-, midrange and high-end systems. Low-
VoIP-T1 T1 Trunk(s)
Fax Gateway end systems (with from 50 to 300 stations) comprise slight-
ly over half of the market; products in the midrange (sup-
Modem VoIP-Analog Analog Trunks porting up to 5,000 stations) comprise about 19 percent of
Device Gateway
the market; and products at the high end (with over 5,000
IP Win2000 stations) account for about 32 percent of the market. These
Phone CallMgr
statistics are based on a survey conducted by Miercom in
Soft IP November 2000.
Phone
IP Trunk
IP PBX Lines/Systems Shipped
(as reported by vendors)
Conceptual Topology:
Shoreline IP Voice Communications System bb Number Lines,
Extensions, Number of IP-PBX
Vendor Users, Ports, What & When Systems
Up to 50 distributed Stations (etc.) Shipped/Installed
Analog switch modules Shipped/
Phone
Dist. Call Control Installed
Dist. Call Control
Fax “IP phones” 5,400 systems
VoIP-Analog 3Com 110,000 shipped as of shipped as of
Gateway September 2000 September 2000
VoIP-Analog
Modem Gateway
Device “extensions”
Altigen 150,000 6,500 systems
shipped as of 3Q
shipped
2000
Win NT IP =PCM / TDM
“IP stations”
switching 2,200 IP-enabled
installed on
Optional Avaya/Definity 70,000 Definity PBXs as of
Definity PBXs as of
Appl’ns November 30, 2000
VoIP-T1
November 30, 2000
Processor
Gateway T1 Trunk
“IP phones” 700+ systems
Dist. Call Control Cisco 210,000 shipped during installed as of
2000 September 2000

Source: 2001:IP Telephony, Copyright 2001, Mier Communications, Inc. Source: 2001:IP Telephony, Copyright 2001, Mier Communications, Inc.
8 9
 IP PBX Lines/Systems Shipped, Continued Q: What are the important performance criteria
(as reported by vendors) I should evaluate when selecting an IP PBX?
Number Lines,
Extensions, Number of IP-PBX A: Miercom has tested many IP PBX systems during the
Vendor Users, Ports, What & When Systems
Stations (etc.) Shipped/Installed
past year, evaluating products according to key perform-
Shipped/
Installed ance criteria, which are based upon our “real-world” expe-
rience working with end users who are implementing
“IP stations” to 200 systems
Mitel 12,000 ship on Ipera 2000 to ship by these systems in their networks.
by January 2001 January 2001

“stations” 150 IP-enabled All potential buyers of IP PBXs raise two performance-
NEC 3,000 shipped as of NEAX systems, as of
3Q 2000 3Q 2000
related concerns:

About 1,000 IP- (1) Is voice quality good enough?

“IP phones”
Nortel/Meridian 19,000 shipped in 2000 enabled Meridians
(Miercom estimate) (2) Is the system reliable enough to sustain our pro
duction voice communications needs?
“IP phones”
OKI 4,000 100 systems
shipped

“IP end points”

Regarding voice quality, the IP PBXs we tested can all
(IP phones and soft 300 systems shipped deliver voice quality that is equivalent to a traditional PBX.
Siemens 12,000
phones) shipped as as of 3Q 2000
of 3Q 2000 By “equivalent” we mean that users in typical work envi-
“ports” installed as 600 to 700 systems ronments would not know they are using an IP-telephony
Shoreline 40,000
of March 2001 (Miercom estimate) based system, as opposed to a traditional PBX. There are
“stations” some qualifications to this blanket statement, but in gen-
shipped as of 200+ “customers” as
Sphere 65,000
September 2000 of September 2000 eral we find that voice quality on IP PBXs is adequate to
“stations”
very good.
800 systems installed
Vertical 16,000 installed as of
as of September 2000
September 2000 On system reliability, we note that IP PBXs represent more
complex systems than most traditional PBXs. And as a
This data provides a general overview of the IP-telephony installed
base, but variations in inclusive dates, units and products make an exact rule, wherever there is greater complexity, there are more
breakdown difficult, if not impossible. For example, the “native-IP” sys- things that can go wrong and, statistically, a higher proba-
tems ofShoreline and Sphere support only analog ports and stations,
while Cisco and 3Com installations consist almost entirely of IP phones.
bility that something will go wrong, all else being equal.

Source: 2001:IP Telephony, Copyright 2001, Mier Communications, Inc.

©2001 Miercom
10 11
 However, several IP PBXs offer fail-over, back-up and Q: Which features are the “must haves”
redundancy features and options, which can appreciably on these systems?
reduce reliability concerns.
A: Traditional PBX systems typically support hundreds of
features, and while their IP PBX counterparts are still play-
ing “catch up” on this point, they are making strides in this
area. A PBX – whether of the traditional or IP PBX variety –
must include a number of “basic” features including the
following: call hold, call transfer, call waiting, call forward-
ing, and call conferencing. More “advanced” features (typ-
ically supported on both types of products) include sup-
port for analog phones, call-detail recording, fax connec-
tion/transmission, voice mail, modem connection and
transmission, 911 special services and accounting/billing
support. Advanced features supported on IP PBXs only
include PC “soft phone” clients, TAPI messaging and IP
phones.
Q: What about security? Are IP PBXs secure enough?

A: IP-telephony systems are currently more susceptible to Basic PBX Feature Support
security breaches, such as malicious hacker attacks, than
are conventional PBX networks. Indeed, the ability of an IP • Call Waiting • Call Hold
PBX to operate over an IP network is a two-edged sword. • Call Forwarding • Call Transfer
While it can extend the reach of a private phone system to
• Call Conferencing • Call Blocking
remote branches and home-office telecommuters, it is
unquestionably also a vulnerability. An IP PBX network can • Caller ID • Speed Dialing
be rendered inoperative via a denial-of-service attack. And • Hunt Groups • Auto Route
while there are documented cases of traditional PBXs Select / Least-Cost
being broken into by hackers, this is not nearly as preva-
Routing
lent – or as easy to do – as it is with an IP PBX.

©2001 Miercom Source: 2001:IP Telephony, Copyright 2001, Mier Communications, Inc.
12 13
 Advanced/Optional/Add-On the standard available, and not all the vendors who claim
to support H.323v2 have implemented the same pieces, in
Features/Capabilities:
(Partial List) the same way, to promise widespread interoperability.

In January 2001, Miercom and Network World co-spon-

• CDR (Call Detail Records) • Voicemail sored a VoIP interoperability lab based on H.323v2. Results
• Billing/accounting software • TAPI message support of this event showed that while the nine vendors who par-
• PWT Wireless • IVR ticipated could set-up a call and establish a voice path
• Unified Messaging • ACD/Call Center based on H.323v2, it took quite a bit of tweaking to get this
• Voice mail/Email Integration • Find Me/Follow Me done. Furthermore, we found out that the H.323 specifica-
• Multi-Tenant • Exceptional scalability tion allowed for two different models of call-control –
• ATM trunks • CTI direct and routed – and gatekeepers that supported each
• QoS • Auto-attendant one were not compatible.
• Multi-media messaging • Web access w/remote call
• Wireless IP Phones • Distinctive ringing Some IP PBX vendors are now eyeing some newer proto-
• Call queuing • VoIP Gateway col standards, such as Session Initiation Protocol (SIP) and
Media Gateway Control Protocol (MGCP) – both developed
within the Internet Engineering Task Force (IETF) – which
Q: What are some of the impediments to wide- are considered much simpler and easier to implement than
spread implementation of IP PBX systems? the ponderous H.323 standard.

A: Clearly, one of the biggest impediments is the lack of a
uniform standard for VoIP call set-up, signaling, IP trunking
and multi-system networking. Without standardization,
VoIP equipment, including IP PBXs, cannot interoperate,
which limits its ability to co-exist in multi-vendor networks.
A number of VoIP standards currently co-exist. The
International Telecommunications Union’s (ITU’s) H.323
“umbrella” standard is the most widely supported today.
Version 2 of H.323 is the most broadly implemented,
according to our research, but there are currently four ver-
sions (versions 1 and 2 are not backward compatible!) of
The ITU’s H.248/Megaco standard has recently been solid-
ified, as well, and we can except to see more implementa-
tions of that standard before the end of 2001.
H.248/Megaco has an interesting history. It was originally a
specification named “Megaco” in the IETF, introduced as a
fix for the shortcomings of MGCP. But it was handed over
to the ITU and subsequently renamed H.248.
The jury is still out, however, on SIP and MGCP, as well as
H.248/Megaco, as to whether they will bring about more
widespread interoperability in the VoIP market.

Source: 2001:IP Telephony, Copyright 2001, Mier Communications, Inc. ©2001 Miercom
14 15
 Q: So, interoperability is really important?
A: Yes. If devices don’t “interoperate” they cannot work
together in the network. That means that a user has to
deploy equipment from the SAME vendor in the network,
putting themselves at a keen disadvantage (one vendor
solution isn’t the best thing in this environment). It also
means that users night have problems communicating
across networks. Interoperability is a major issue in the
development of communications technologies. Issues are
technical, as well as political.
Q: What about pricing? Can we save money with an
IP PBX?
A: Almost certainly not; not at this time. It will invariable
cost you more today to buy and deploy an IP PBX than to
purchase the equivalent capacity in traditional PBX equip-
ment. Based on U.S. List prices, IP PBX per-port prices
average in the $550 to $650 range, while traditional PBXs
hover closer to $400 per port. That’s roughly a 50-percent
premium for IP Telephony.
This price comparison presumes that all else is equal,
which is seldom the case. PBXs and IP PBXs differ consid-
erably, as far as which features are included as part of the
basic system and which are extra-priced options. For
example, many IP PBXs include voice mail integrally, while
others offer it as an extra-priced option. The same is true
©2001 Miercom
16
for automated attendant and interactive voice response
(IVR) support.
Most enterprises that are testing or deploying IP PBXs
today are not doing so to save money in the short term,
although that is a reasonable prospect for the long term.
Rather, the organizations view IP Telephony as a strategic
technology for the future and are testing the waters to gain
competitive experience with the technology, as well as to
groom their in-house networking staff.
Q: What is an IP phone and why do I need one?
A: You don’t need an IP phone to work with an IP PBX.
Most support analog and/or digital phones, but an IP
phone offers greater features - the biggest being the abili-
ty to handle the “converged” voice and data (which an
analog phone cannot do). So if one wants to realize the full
potential of the IP PBX, it’s best to have an IP phone
Q: How much should I expect to pay for an IP phone?
A: According to our research, prices for an IP phone can
range from about $200 up to around $770, depending on
the extent of features and options supported. . A mid-
range price is about $487, but can be expected to drop
once the technology becomes more widely implemented.
Look for IP phones to become commodities as the market
stabilizes.
©2001 Miercom
17
 Q: Where can I obtain more detailed information
about IP PBXs?
A: Miercom has recently published an in-depth study of
this new technology. This 250-page resource includes an
overview of the technology, an “apples-to-apples” com-
parison of 22 systems (based on Miercom’s most recent
survey of the IP PBX market), as well as “hands-on” test-
ing results of eight leading IP PBXs from the following ven-
dors: Alcatel (OmniPCX 4400); Avaya (IP600 IP
Communications Server); Cisco Systems (AVVID and
CallManager); Mitel (Ipera 2000); NEC (NEAX 2500 IPX);
Shoreline (IP Voice Communications System); Sphere
Communications (Sphericall); and Vertical Networks
(InstantOffice). Each “hands-on” report includes at least 20
pages of information per vendor, based on tests conduct-
ed recently at Miercom’s independent labs in Princeton
Junction, NJ.
©2001 Miercom
18
Glossary of Terms
ADPCM (Adaptive Differential Pulse Code Modulation): A
voice encoding standard, which yields fairly high voice
quality within a digital stream of 24 to 32 kbps; a voice
encoding that’s used in some ITU-T-specified vocodings,
including G.726.
AMIS (Audio Messaging Interchange Specification): An
industry standard for network communications between
different voice messaging systems. Regarded as a
rudimentary interface for connecting telephony (including
IP telephony) and voicemail systems.
ATM (Asynchronous Transfer Mode): A wide-area,
carrer/ISP-oriented, Layer 2 transmission technology that
carries all information – voice, data, video – within 53-byte
cells (48 bytes of payload, 5 bytes overhead), typically over
SONET-based fiber transport. Provides excellent QoS, but
high overhead and fundamentally incompatible with
packet-based IP and VoIP.
Call Agent: A logical node that provides signaling and con-
trol functions similar to those performed by a Class 5 (“end
office”) PSTN central office switch. The call agent, which
is integral to VoIP call-control protocols including MGCP
and H.248/Megaco, provides these signaling and control
functions via IP, rather than the Signaling System 7 (SS7)
network.
Codec (Coder/DeCoder): Any technology that digitizes
information that is not inherently in digital form, such as
audio voice or video, typically for transmission via a data
©2001 Miercom
19
 network. Codecs can be implemented in software, hard-
ware, or a combination of both.
DS-1, DS-3: See T1, T3.
DTMF (Dual Tone Multi Frequency): A combination of two
specific frequencies or tones assigned to each touch-tone
telephone key, to facilitate advanced signaling; replaced
pulse, or rotary, dialing.
Endpoint: An end-user device, whether a PC, phone or
other terminal device, which originates a call to another
endpoint (or endpoints) through the network; same as a
terminal.
Frame packing: The number or duration of digitized voice
samples that are included in the same IP packet; a variable
that’s often accessible and user-settable in VoIP gateways.
The frame-packing value directly affects the amount of
bandwidth that a VoIP stream requires, as well as the
end-to-end latency.
G.711, G.723.1, G.726, G.729, G.729a: See table
under vocoders
Gain: Also known as ‘’level;’’ the ‘’volume’’ control of an
analog or PCM-digitized voice signal, measured in decibels
(dB); a key parameter in tuning VoIP networks, which
greatly influences voice quality; many VoIP gateways offer
access to adjust gain on a per-channel basis, ideally in both
incoming and outgoing directions.
Gatekeeper: As defined in H.323: a network control node (a
standalone platform) that manages all gateways within the
©2001 Miercom
20
gatekeeper’s “zone” of responsibility. At a minimum, a
gatekeeper handles call routing and PSTN-IP address
translation. In practice, the same physical ‘’gatekeeper’’
node will also often handle additional functions, which
may include IVR (interactive voice response), pre-paid
calling card support/caller authentication, and/or CDR (call
detail recording) storage/processing.
Gateway: In VoIP parlance, a hardware and software plat-
form which handles the interface between IP packet-based
networks and circuit-switched networks (such as the PSTN,
or PBX); performs code and protocol conversion required
to enable real-time voice connections between end users
across these two different networks.
Ground Start: Call initiation from a user-based phone sys-
tem, such as a PBX, to the PSTN central office by briefly
grounding one side of a line.
H.248: ITU-T specification for centralized VoIP call control
over media gateways; initiated within an ‘informational’
IETF RFC as MEGACO (MEdia GAteway COntrol).
H.323: ITU-T specification that defines multimedia commu-
nications over packet-based networks. The first version of
H.323 was adopted in 1996; currently there are 4 versions
of the H.323 specification. H.323 is an “umbrella” specifi-
cation, which includes by reference many other specifica-
tions and protocols that define other aspects of multimedia
communications.
H.450: ITU-T specification that defines support of supple-
mentary services for use with H.323-based IP telephony.
©2001 Miercom
21
 Hybrid Switch, or VoIP-based Central Office Switch: A
class of hardware and software switching system designed
to integrate classical telephony switching, such as
PSTN/Central Office, with IP transport (VoIP). Unlike gate-
ways, which augment central-office switches with add-on
VoIP functionality, hybrid switches are designed to sup-
plant existing CO switches (such as 4ESS, 5ESS, DMS 100,
DMS 250, etc.). These are specialized, high-capacity VoIP
systems, typically scaleable to beyond 100 T1s.
IP Phone: A telephone “instrument,” which may be a
standalone desktop device or a software application that
runs on a user’s PC, that delivers real-time voice commu-
nications and operates over an IP network.
ITU-T: The ITU Telecommunication Standardization Sector
(ITU-T); one of the three sectors of the International
Telecommunication Union (ITU). The ITU-T was created in
1993 within the framework of the “new” ITU, replacing the
former International Telegraph and Telephone
Consultative Committee (CCITT). A United Nations organi-
zation, the ITU-T develops international standards cover-
ing all areas of telecommunications, except radio aspects.
Jitter: Variation in latency that occurs over an IP data net-
work that’s exhibiting problems (typically congestion).
Jitter can affect the synchronization and time-sequence
reassembly of a VoIP packet stream. A key requirement of
VoIP equipment is to be able to adjust jitter.
Jitter buffer: A special temporary buffer that’s used by a
VoIP receiver (such as a gateway) to realign and resyn-
chronize voice samples that were sent in separate packets
across an IP network. Expressed in milliseconds of voice-
©2001 Miercom
22
sample duration. The ‘’size’’ of the jitter buffer (say, 60 mil-
liseconds) adds directly to the end-to-end latency that each
voice stream experiences. All VoIP systems apply, at a
minimum, a jitter buffer the size of one packet’s voice sam-
ple (which can vary with vocoder and frame packing). The
jitter buffer should grow in proportion to the amount of jit-
ter that the network applies to VoIP streams, yet stay as
small as possible due to the effect on latency. A “dynam-
ic” jitter buffer, which starts low, monitors jitter automati-
cally, and expands only if required, offers the best solution
to addressing jitter.
LAN/IP-based PBX: Another name for an IP telephony
system - hardware and software - which implements clas-
sical PBX functions over a data LAN (such as Ethernet),
rather than separate star-wired telephone cabling, and
which supports VoIP. These systems may or may not
include “station” equipment, such as a ‘’PC’’ or ‘’IP
phone,” and may or may not also support conventional
analog or digital phone sets (via VoIP adapters/converters,
as applicable).
Latency: The constant delay experienced by information
traversing a network. There are many sources of delay
(routers in an IP network, propagation delay, etc.) and all
contribute to cumulative latency. Latency is the single
most important characteristic affecting VoIP voice quality.
Based on extensive VoIP tests by Miercom, one-way, end-
to-end latency under 100 milliseconds helps assure excel-
lent voice quality; latencies of 100 to 150 milliseconds
result in noticeable, but acceptable delay; and latencies
over 150 milliseconds have a marked degrading effect on
perceived voice quality.
©2001 Miercom
23
 Layers 1, 2, 3 and 4: Referring to the “ISO Reference
Model,” a delineation and separation of network functions
into discrete layers. As applied in VoIP and IP telephony
(and in most other network references), “Layer 1” refers to
the “physical layer” of a network, involving such issues as
cabling, power and connectors. Layer 2, called the “link
layer,” refers to the exchange of data between adjacent
nodes (point to point), and involves such technologies as
Ethernet (the “MAC” layer), ATM, and PPP (the point-to-
point protocol). Layer 3, called the “network layer,”
involves the movement of data end-to-end through a
multi-node network, and includes, for example, all the
functions of IP — packet structure, IP addressing, error
detection and correction. Layer 4, called the “transport
layer,” now generally refers to “higher” network function-
ality above IP (Layer 3). VoIP, for example, employs vari-
ous Layer 4 mechanisms, such as the connection-less UDP
(user datagram protocol) and RTP (the real-time protocol).
Loop Start: A call initiation signal created by a loop across
the two wires of a telephone local-loop wire pair.
MCU (Multipoint Control Unit): Defined in H.323; an MCU
allows connection of multiple units for conferencing.
MOS (Mean Opinion Score): An ITU-T standardized proce-
dure for rating the voice quality of VoIP communications,
typically from recordings of male and female voices played
through a VoIP system. A MOS panel involves quality
ratings of voice recordings by a diversified group of non-
technical laypersons (Miercom MOS panels employ 10 or
more people). A five-point scale is used, where 1 repre-
sents the poorest voice quality and 5 represents perfect
voice quality.
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MPLS (Multi-Protocol Label Switching): an IETF initiative
integrating Layer 2 information about bandwidth, latency,
utilization into Layer 3 (IP) within a particular Internet
Service Provider (ISP) in order to simplify and improve
IP-packet exchange.
Media Gateway Controller (MGC): As defined in VoIP
control protocols including MGCP and H.248/Megaco;
provides the interface between the PSTN (including the
SS7 signaling network) and an IP-based VoIP network.
MEGACO (MEdia GAteway COntrol): The original name
for a VoIP call-control specification that evolved to become
the ITU-T’s H.248 specification for VoIP gateway-gateway
and gateway-control communications.
MGCP (Media Gateway Control Protocol): An IETF-origi-
nated VoIP call-control protocol, version 1.0 of which was
published in October 1999 as RFC 2705 for ‘’informational’’
purposes (and specifically NOT as an Internet standard).
MGCP is based on a centralized ‘’call agent’’ and address-
es the dialog between call agents and gateways. MGCP is
now largely being promulgated and refined under the
aegis of the International Softswitch Consortium.
Multiplexing: Simultaneously transmitting two or more,
otherwise independent information streams over a single
channel. A “multiplexer” (also known as a “mux”) is a net-
work device used to divide a transmission facility into two
or more subchannels. This may be done using frequency-
division, time-division or other technologies. Multiplexers
enable bandwidth savings and more efficient use of trans-
port facilities and channels.
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 PBX (Private Branch Exchange): A telephone system used
within an enterprise or private organization. A PBX
enables the organization’s users to share a number of out-
side lines, which is much less expensive than connecting
an external telephone line to everyone in the organization.
PCM (Pulse Code Modulation): A voice-digitization tech-
nique where an analog signal is sampled 8,000 times per
second. Each sample is then represented by an 8-bit
digital value, for a total digital-stream bandwidth of 64
Kbps. There are two standards for PCM coding: Mu-Law
(used in North America and in Japan) and the A-Law
standard (used in most other countries).
PSTN (the Public Switched Telephone Network), the
current, ubiquitous public phone network, based on
circuit-switching and TDM (time-division multiplexing), as
opposed to VoIP, which is based on IP packet and data-
routing technology.
QoS (Quality of Service): A guaranteed level of network
performance – with regards to latency and packet loss (and
usually also jitter and round-trip delay) for specified types
of data traffic, such as VoIP or Web transactions. ATM
providers with QoS capabilities can guarantee to their cus-
tomers, for example, that end-to-end latency will not
exceed a specified level.
Q.Sig: A signaling system and protocol defined for ISDN
(Integrated Services Digital Network) communications,
which has been adopted by some phone system vendors
(mainly non-IP, traditional PBXs) for system-to-system sig-
naling.
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Signaling Gateway: Similar in many ways to a call agent;
a signaling gateway is a VoIP network entity that provides
the necessary signaling conversion between an MGC
(media gateway controller) and the SS7 network.
Silence suppression: see VAD (Voice Activity Detection).
The ability of some VoIP products and systems to identify
silence periods in a voice stream and to eliminate “empty”
packets (representing silence in the conversation) from the
packet stream. This reduces the VoIP bandwidth require-
ments dramatically.
SIP (Session Initiation Protocol): VoIP call-control protocol
developed within the IETF and published initially in March
1999 as RFC 2543, with several subsequent Internet-Draft
revisions and extensions. SIP is an application-layer
signaling protocol that specifies IP-phone call control.
Unlike other VoIP control protocols, SIP is characterized by
distributed call control, which is embodied in stateful or
stateless proxy servers.
Soft phone: A software application that typically runs on
an end-user’s desktop PC or laptop, and which provides all
the functionality of a standalone IP phone, obviating the
need for a separate desktop phone. Besides soft phones,
many vendors of IP telephony systems offer special
“client” software designed to enhance telephone function-
ality (such as for browsing contact lists), but these are
designed to augment, rather than replace, a regular phone
set.
SONET (Synchronous Optical Network): A standard that
defines high-speed transmission formats, multiplexing
and connectivity for carrier and service provider fiber-optic
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 transmission systems. An ANSI standard, Bellcore first into time slots, each representing a separate subchannel.
proposed SONET in the mid-1980s. The standard defines a Similar to a “TDM bus” switching fabric.
hierarchy of interface rates allowing data streams at differ-
ent rates to be multiplexed. The international equivalent of VAD (Voice-Activity Detection), also Silence Suppression:
SONET, standardized by the ITU, is called SDH The ability of some VoIP equipment, as an added feature of
(Synchronous Digital Hierarchy). their vocoding, to monitor individual voice streams for
periods of silence, and to eliminate these from the packet
SS7 (Signaling System #7): the “control” network of the stream that is transmitted over the IP network. VAD is a
PSTN; considered arcane and overly complex by many. powerful way to reduce the IP-network bandwidth required
SS7 signaling and control information is carried in a sepa- to convey real-time voice conversations, since most voice
rate, out-of-band network (not along with voice traffic). conversations are inherently half duplex (one party speaks
Features different levels of interfaces and functional mes- while the other is silent).
sage sets.
Vocoder (Voice Coder) The implementation of any technol-
T1, T3: T1 (also referred to as DS-1 or DS-3). A T1 is a dig- ogy that converts an analog speech signal into a binary,
ital communications facility that supports a bi-directional digital stream for transmission or storage. A correspon-
1.544-Mbps digital stream. One T1 link can be divided into ding voice decoder receives this bit stream and reconsti-
24 voice channels. A T3 (DS-3) digital facility supports tutes it back into an analog signal form. The below table
44.736 Mbps of traffic, typically subdivided into 28 T1s, or summarizes the key comparative aspects of leading ITU-T-
672 voice channels (each represented by a 64-kbps DS-0). standardized vocoders used in VoIP.
The European equivalents are called E1, which supports
2.04 Mbps of traffic, and E3, which supports 35 Mbps. Prominent Vocoders:
TAPI: (Telephony Application Programming Interface). A Vocoder Bandwidth of Relative processing Relative voice
(ITU-T encoded voice complexity and delay quality (based on
software interface – i.e., a set of standard software “calls” reference) stream (per Miercom Interactive
(latency) associated
– which allows a sophisticated end user or application soft- direction; does with this vocoder and Mean Opinion
not include IP or Score tests)
ware developer to program advanced features and servic- packet overhead)
es and/or communications devices to allow blending (con-
G.711 64 kbps Low Very High
vergence) of PSTN and IP telephony applications.
G.729 8 kbps High High
G.729a 8 kbps Medium to high Medium
TDM (Time-Division-Multiplexing) switching matrix: Most G.723.1 5.3 or 6.3 kbps Medium Low to medium
“traditional” digital PBXs and phone systems support this G.726 32 kbps Low Medium
type of switching fabric, which is based on time-division G.728 16 kbps Low to medium Medium
multiplexing, where transmission bandwidth is divided GSM EFR 13 kbps Low to medium Low to medium
(Enhanced
Full Rate)

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 VoIP (Voice over Internet Protocol): Also referred to as IP
telephony, Voice over the Internet or Internet Telephony. A
technology, industry, marketplace and broad category of
hardware and software that enables real-time voice tele-
phone calls to be carried via an IP data network.
Zone: Defined in H.323 as a collection of gateways and all
endpoints associated with those gateways; each zone is
under the control of a gatekeeper.
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About Miercom:
Miercom, founded in 1988 as Mier Communications,
Inc., is an independent networking consultancy and prod-
uct-test center located in Princeton Junction, NJ. The com-
pany pioneered the comparative assessment of network-
ing hardware and software, having developed methodolo-
gies for testing products. Miercom has developed the
industry’s first comparative test methodologies for testing
VoIP equipment.
In 1995, the company launched its “NetWORKS As
Advertised” program, in which any vendor can submit its
networking-related products for a comprehensive, inde-
pendent assessment. Many companies, including most of
the leaders in networking, have had their products tested
and certified by Miercom.
Miercom is also a member of the Network World
Global Test Alliance and the test lab of record for Business
Communications Review (BCR). Miercom staffers have
published numerous articles about VoIP technologies and
products, including IP PBXs, in these publications.
Miercom also publishes special reports on important
networking technologies. Call 609-490-0200 for more infor-
mation on the latest report, “2001: IP Telephony: The
Definitive Special Report on IP PBXs.” Visit www.mier.com
for more information on Miercom’s full line of products
and services.
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