Beruflich Dokumente
Kultur Dokumente
Voice over IP
(VoIP)
Table of Contents
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Applications and Benefits of VoIP . . . . . . . . . . . . 4
IP Network Support for Voice . . . . . . . . . . . . . . 14
Software Support for VoIP . . . . . . . . . . . . . . . . . 19
Implementing VoIP in Systems. . . . . . . . . . . . . . 26
Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
CASE STUDY: How Entrata’s Partnership
With Telogy Networks Filled a Niche
in the Voice over Packet Market . . . . . . . . . . 31
Glossary of Terms . . . . . . . . . . . . . . . . . . . . . . . 34
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Telogy Text 10/14/98 2:38 PM Page 2
possibility of introducing usage-based pricing and the very least, will stimulate improvements in cost and
increasing their traffic volumes is very attractive. Users function throughout the industry.
are seeking new types of integrated voice/data applica- Figure 1 illustrates one scenario for how telephony
tions as well as cost benefits. and facsimile can be implemented using an IP
Successfully delivering voice over packet networks network. This design would also apply if other types of
presents a tremendous opportunity; however, imple- packet networks (such as frame relay) were being used.
menting the products is not as straightforward a task as
it may first appear. This Technology Guide examines
PC with PC with
the technologies, infrastructures, software, and systems Fax Software Voice Software
drive the market and benefit the most from the conver- Voice
Switch
Voice
Switch
gence of voice and data networks will be identified. Private Voice
Phone Network Phone
PSTN
Some examples of VoIP applications that are likely f) Internet call center access: Access to call
to be useful would be: center facilities via the Internet is emerging as a
a) PSTN gateways: Interconnection of the Internet valuable adjunct to electronic commerce applica-
to the PSTN can be accomplished using a gateway, tions. Internet call center access would enable a
either integrated into a PBX (the iPBX) or customer who has questions about a product being
provided as a separate device. A PC-based offered over the Internet to access customer service
telephone, for example, would have access to the agents online. Another VoIP application for call
public network by calling a gateway at a point centers is the interconnection of multiple call
close to the destination (thereby minimizing long centers.
distance charges).
One of the immediate applications for IP
b) Internet-aware telephones: Ordinary telephony is real-time facsimile transmission. Facsimile
telephones (wired or wireless) can be enhanced to services normally use dial-up PSTN connections, at
serve as an Internet access device as well as speeds up to 14.4 Kbps, between pairs of compatible
providing normal telephony. Directory services, for fax machines. Transmission quality is affected by
example, could be accessed over the Internet by network delays, machine compatibility, and analog
submitting a name and receiving a voice (or text) signal quality. To operate over packet networks, a fax
reply. interface unit must convert the data to packet form,
c) Inter-office trunking over the corporate handle the conversion of signaling and control proto-
intranet: Replacement of tie trunks between cols (the T.30 and T.4 standards), and ensure complete
company-owned PBXs using an Intranet link delivery of the scan data in the correct order. For this
would provide economies of scale and help to application, packet loss and end-to-end delay are more
consolidate network facilities. critical than in voice applications.
Most VoIP applications that have been defined are
d) Remote access from a branch (or home)
considered to be real-time activities. Store-and-forward
office: A small office (or a home office) could gain
voice services will also be implemented using VoIP. For
access to corporate voice, data, and facsimile
example, voice messages could be prepared locally
services using the company’s Intranet (emulating a
using a telephone and delivered to an integrated
remote extension for a PBX, for example). This
voice/data mailbox using Internet or intranet services.
may be useful for home-based agents working in a
Voice annotated documents, multimedia files, etc. will
call center, for example.
also become standard within office suites in the near
e) Voice calls from a mobile PC via the future. The real-time and store-and-forward modes of
Internet: Calls to the office can be achieved using operation will need to be compatible and
a multimedia PC that is connected via the interoperable.
Internet. One example would be using the Widespread deployment of a new technology
Internet to call from a hotel instead of using seldom occurs without a clear and sustainable justifica-
expensive hotel telephones. This could be ideal for tion, and this is also the case with VoIP. Demonstrable
submitting or retrieving voice messages. benefits to end users are also needed if VoIP products
(and services) are to be a long-term success. Generally, SNMP-based management can be provided for
the benefits of technology can be divided into the both voice and data services using VoIP. Universal
following four categories: use of the IP protocols for all applications holds
• Cost Reduction. Although reducing long out the promise of both reduced complexity and
distance telephone costs is always a popular topic more flexibility. Related facilities such as directory
and would provide a good reason for introducing services and security services may be more easily
VoIP, the actual savings over the long term are still shared.
a subject of debate in the industry. Flat rate • Advanced Applications. Even though basic
pricing is available with the Internet and can telephony and facsimile are the initial applications
result in considerable savings for both voice and for VoIP, the longer term benefits are expected to
facsimile (at least currently). It has been estimated be derived from multimedia and multiservice
that up to 70% of all calls to Asia are to send applications. For example, Internet commerce
faxes, most of which could be replaced by FoIP. solutions can combine WWW access to informa-
These lower prices, however, are based on tion with a voice call button that allows immediate
avoiding telephony access charges and settlement access to a call center agent from the PC. Needless
fees rather than being a fundamental reduction in to say, voice is an integral part of conferencing
resource costs. The sharing of equipment and systems that may also include shared screens,
operations costs across both data and voice users whiteboarding, etc. Combining voice and data
can also improve network efficiency since excess features into new applications will provide the
bandwidth on one network can be used by the greatest returns over the longer term.
other, thereby creating economies of scale for
voice (especially given the rapid growth in data Although the use of voice over packet networks is
traffic). relatively limited at present, there is considerable user
• Simplification. An integrated infrastructure that interest and trials are beginning. End user demand is
supports all forms of communication allows more expected to grow rapidly over the next five years. Frost
standardization and reduces the total equipment & Sullivan and other research firms have estimated
complement. This combined infrastructure can that the compound annual growth rate for IP-enabled
support dynamic bandwidth optimization and a telephone equipment will be 132% over the period
fault tolerant design. The differences between the from 1997 to 2002 (from $47.3M in 1997 to $3.16B by
traffic patterns of voice and data offer further 2002). It is expected that VoIP will be deployed by
opportunities for significant efficiency 70% of the Fortune 1000 companies by the year 2000.
improvements. Industry analysts have also estimated that the annual
revenues for the IP fax gateway market will increase
• Consolidation. Since people are among the from less than $20M in 1996 to over $100M by the
most significant cost elements in a network, any year 2000. It is clear that a market has already been
opportunity to combine operations, to eliminate established and there exists a window of opportunity
points of failure, and to consolidate accounting for developers to bring their products to market.
systems would be beneficial. In the enterprise,
Delay: Two problems that result from high end- Phone Phone
Voice Packet Voice
to-end delay in a voice network are echo and Processing
R Network R
Processing
talker overlap. Echo becomes a problem when the Compression: 20-45 Queuing: Transmission: .25 - 7 Queuing: Decompression: 10
Inter-Process: 10 Up to 20 Network: Variable Up to 20 Inter-Process: 10
round-trip delay is more than 50 milliseconds. up to 400+ Buffer: 50
(configurable)
Since echo is perceived as a significant quality
problem, VoIP systems must address the need for
Figure 3: Delay and Jitter
echo control and implement some means of echo
cancellation. Talker overlap (the problem of one
Maintenance of acceptable voice quality levels
caller stepping on the other talker’s speech)
despite inevitable variations in network performance
becomes significant if the one-way delay becomes
(such as congestion or link failures) is achieved using
greater than 250 milliseconds. The end-to-end
such techniques as compression, silence suppression,
delay budget is therefore the major constraint and
and QoS-enabled transport networks. Several develop-
driving requirement for reducing delay through a
ments in the 1990s, most notably advances in digital
packet network.
signal processor technology, high-powered network
Jitter (Delay Variability): Jitter is the variation switches, and QoS-based protocols, have combined to
in inter-packet arrival time as introduced by the enable and encourage the implementation of voice
variable transmission delay over the network. over data networks. Low-cost, high-performance DSPs
Removing jitter requires collecting packets and can process the compression and echo cancellation
holding them long enough to allow the slowest algorithms efficiently.
packets to arrive in time to be played in the Software pre-processing of voice conversations can
correct sequence, which causes additional delay. also be used to further optimize voice quality. One
The jitter buffers add delay, which is used to technique, called silence suppression, detects whenever
remove the packet delay variation that each packet there is a gap in the speech and suppresses the transfer
is subjected to as it transits the packet network. of things like pauses, breaths, and other periods of
Packet Loss: IP networks cannot provide a guar- silence. This can amount to 50-60% of the time of a
antee that packets will be delivered at all, much call, resulting in considerable bandwidth conservation.
less in order. Packets will be dropped under peak Since the lack of packets is interpreted as complete
loads and during periods of congestion (caused, silence at the output, another function is needed at the
for example, by link failures or inadequate receiving end to add “comfort noise” to the output.
capacity). Due to the time sensitivity of voice Another software function that improves speech
transmissions, however, the normal TCP-based re- quality is echo cancellation. As was noted earlier, echo
transmission schemes are not suitable. Approaches becomes a problem whenever the end-to-end delay for a
used to compensate for packet loss include inter- call is greater than 50 milliseconds. Sources of delay in a
polation of speech by re-playing the last packet, packet voice call include the collection of voice samples
and sending of redundant information. Packet (called accumulation delay), encoding/decoding and
losses greater than 10% are generally not packetizing time, jitter buffer delays, and network transit
tolerable. delay. The ITU recommendation G.168 defines the
performance requirements that are currently required most of the time). This would generally be the
for echo cancellers. case with a private IP network (an Intranet) that is
Engineering a VoIP network (and the equipment owned and operated by a single organization.
used to build it) involves trade-offs among the quality of • Using management tools to configure the
the delivered speech, the reliability of the system, and network nodes, monitor performance, and
the delays inherent in the system. Minimizing the end- manage capacity and flow on a dynamic basis.
to-end delay budget is one of the key challenges in Most internetworking devices (routers, switches,
VoIP systems. Ensuring reliability in a “best effort” etc.) include a variety of mechanisms that can be
environment is another. Equipment producers that useful in supporting voice. For example, traffic can
offer the flexibility to configure their systems to fit the be prioritized by location, by protocol, or by appli-
environment and thereby optimize the quality of the cation type, thereby allowing real-time traffic to be
voice produced will have a competitive advantage. given precedence over non-critical traffic.
Queuing mechanisms can also be manipulated to
minimize delays for real-time data flows. More
recent developments, such as tag switching and
IP Network Support for Voice flow switching, can also improve overall
performance and reduce delays.
A key requirement for successful VoIP deployment
• Adding control protocols and mechanisms
is the availability of an underlying IP-based network
that help avoid or alleviate the problems inherent
that is capable of supporting real-time telephone and
in IP networks. Protocols such as RTP (real-time
facsimile. As was noted above, voice quality is affected
protocol) and RSVP (Resources Reservation
by delay, jitter, and unreliable packet delivery - all of
Protocol) are also being used to provide greater
which are typical characteristics of the basic IP
assurances of controlled QoS within the network.
network service.
Other mechanisms such as admission controls and
Most of today’s data network equipment - routers,
traffic shaping may also be used to avoid
LAN switches, ATM switches, network interface cards,
overloading a network (this would be comparable
PBXs, etc. - will need to be able to support voice
to getting a network busy signal on the telephone
traffic. Furthermore, VoIP-specific equipment will
at peak periods such as Christmas).
either have to be integrated into these devices or work
compatibly with them. VoIP equipment must also
VoIP equipment, which can be categorized into
accommodate environments ranging from private, well-
client, access/gateway, and carrier class/infrastructure
planned corporate Intranets to the less predictable
segments, should be configurable to capitalize on these
Internet. Three different techniques are used
different techniques but must also be sufficiently flex-
(separately or in combination) to improve network
ible to add new techniques as they become available.
quality of service.
Producers that make use of embedded software should
• Providing a controlled networking environ- focus on how to best utilize the functions instead of
ment in which capacity can be pre-planned and focusing on the problems associated with implementing
adequate performance can be assumed (at least and testing the objects themselves.
Real-time voice traffic can be carried over IP Future VoIP networks will include IP-based PBXs
networks in three different ways: (iPBXs), which will emulate the functions of a
• Voice trunks can replace the analog or digital traditional PBX. These will allow both standard tele-
circuits that are serving as voice trunks (such as phones and multimedia PCs to connect to either the
private links between company-owned PBXs) or PSTN or the Internet, providing a seamless migration
PSTN-access trunks (links between a PBX and the path to VoIP. An iPBX can also combine the features
carrier). Voice packets are transferred between of today’s switches and routers and could become the
pre-defined IP addresses, thereby eliminating the gateway into a variety of value-added services such as
need for phone number to IP address conversions. directories, message stores, firewalls and other network-
Fallback to the PSTN (or other private voice based servers. Such a VoIP system would also combine
circuits) is always an option in this scenario. real-time and non real-time communications. Voice
and facsimile messaging, for example, use functions
• PC-to-PC voice can be provided for multimedia that are very similar to a telephone call but do not
PCs (i.e., PCs with a microphone and sound need the same levels of QoS in the underlying
system) operating over an IP-based network network.
without connecting to the PSTN. PC applications Figure 4 illustrates the IP network protocols that
and IP-enabled telephones can communicate are currently being used to implement VoIP.
using point-to-point or multipoint sessions (a form
of Internet ham radio). This type of system may
emulate a CB radio or an Internet chat group and H.323
could be combined with shared data systems such RTP, RTCP, RSVP
as whiteboards (i.e., multimedia solutions). UDP/ TCP
Network Layer (IPv4, IPv6, IPM)
• Telephony (any phone-to- any other phone) Data Link Layer
communications (as is illustrated in Figure 1) Physical Layer
appears like a normal telephone to the caller but
may actually consist of various forms of voice Figure 4: VoIP Protocol Structure
over packet network, all interconnected to the
PSTN. Gateway functionality is required when
interconnecting to the PSTN or when interfacing
the standard telephones to a data network. In the
future, IP-enabled telephones will connect directly.
For true universality, standards for VoIP (and voice
over frame relay or ATM) must be adopted and
applied.
IA 1.0 VoIP Forum Implementation Voice and telephone calling can be viewed as one
Agreement 1.0 selecting protocol of many applications for an IP network, with software
options for interoperable VoIP
being used to support the application and interface to
TCP, UDP Internet standard Transport Layer the network. The emergence of VoIP is a direct result
protocols
of the advances that have been made in hardware and
IPv4, IPv6, IP multicast and Internet standard Network Layer software technologies in the early 1990s.
various routing protocols protocols (currently IPv4 is in The software functionality required for voice-to-
widespread use) both for data
transfer and routing
packet conversion in a VoIP terminal or gateway are:
the packets to the IP network (or other packet “comfort noise” can be inserted into the call (so
networks). Signaling information is converted from that the listener does not get dead air on their tele-
telephony protocols to the packet signaling phone).
protocol. d) The Tone Detector, which detects the reception
• The Network Management module, which of DTMF tones and discriminates between voice
provides management agent functionality, allowing and facsimile signals. These can be used to invoke
remote fault, accounting, and configuration the appropriate voice processing functions (i.e., the
management to be performed from standard decoding and packetizing of facsimile information
management systems (see the next section). The or the compression of voice).
Network Management module could include e) The Tone Generator, which generates DTMF
ancillary services such as support for security tones and call progress tones under command of
features, access to dialing directories, and remote the operating system.
access support.
f) The Facsimile Processing module, which
The Voice Processing module must include soft- provides a facsimile relay function by
ware to perform the following functions: demodulating the PCM data, extracting the rele-
vant information, and packing the scan data into
a) The PCM Interface, which receives samples packets.
from the telephony (PCM) interface and forwards
them to the appropriate VoIP software module for g) The Packet Voice Protocol module, which
processing (and vice versa). The PCM interface encapsulates the compressed voice and fax data for
performs continuous phase re-sampling of output transmission over the data network. Each packet
samples to the analog interface. includes a sequence number that allows the
received packets to be delivered in the correct
b) The Echo Cancellation Unit, which performs order. This also allows silence intervals to be repro-
echo cancellation on sampled, full-duplex voice duced properly and lost packets to be detected.
port signals in accordance with the ITU G.165 or
G.168 standard. Since round-trip delay for VoIP is h) The Voice Playout module at the destination,
always greater than 50 milliseconds (the point at which buffers the packets that are received and
which echo becomes intolerable), echo cancellation forwards them to the voice codec for playout. This
is a requirement. Operational parameters may be module provides an adaptive jitter buffer and a
programmable. measurement mechanism that allows buffer sizes to
be adapted to the performance of the network.
c) The Voice Activity/Idle Noise Detector,
which suppresses packet transmission when voice The Call Processing (signaling) subsystem detects
signals are not present (and hence saves additional the presence of a new call and collects addressing
bandwidth). If no activity is detected for a period information. Various telephony signaling standards
of time, the voice encoder output will not be trans- must be supported. A number of functions must be
ported across the network. Idle noise levels are also performed if full telephone calling is to be supported.
measured and reported to the destination so that
• The interface to the telephone network must be The ability to digitize and process voice streams
monitored to collect incoming commands and using self-contained software building blocks is the key
responses. to success with VoIP implementation. VoIP equipment
• The signaling protocols (e.g., E&M) must be should comply with the H.323 standard which has
terminated and the information must be been defined by the ITU to describe terminals, equip-
extracted. ment, and services for multimedia communication over
networks (such as LANs or the Internet) that do not
• The signaling information must be mapped into a provide a guaranteed QoS. H.323 is a family of soft-
format that can be used to establish a session ware-based standards that define various options for
across the packet network. compression and call control. Figure 5 illustrates the
• Telephone numbers (E.164 dial addresses) must be functional components of terminals that use the H.323
converted into IP addresses (with the potential standards.
need for an external reference to a directory
service). Two approaches to dialing are being
Voice Network SNMP
used: single stage (dial the destination number and Speech
Call Processing
Processing Management Messages
use automatic route selection functions), and two
stage (dial the VoIP gateway number, then dial the
Packet IP
Signaling
real destination). Processing Packets
Needless to say, the software used in VoIP devices Figure 5: Voice Gateway/Terminal Functions
must also be supported by a real-time operating envi-
ronment and provided with the ability to communicate The following table lists the various standards that
among the modules and with the external world. have been adopted as part of the H.323 family. These
Implementation of protocols is another area where are implemented as part of the software described
development time, testing, and risk can be minimized above and need to be “open,” that is, implementations
through the use of embedded software. The objective from multiple vendors must be compatible.
should always be to develop new ways to optimize the
use of standard protocol software, not to re-invent
basic functions that require extensive testing for stan-
dards compliance and product interoperability.
H.323 and Related Recommendations Although H.323 is the recognized standard for
Recommendation Brief Description VoIP terminals, there are additional standards that are
more appropriately suited for client applications, such
H.323 Document called “Visual telephone
systems and equipment for local as IP phones. As H.323 was originally designed for the
area networks which provide a desktop, a higher priority was given to rich function-
non-guaranteed quality of service”
(November, 1996)
ality, rather than resource allocation. This has given
rise to alternative protocols that can interoperate with
H.225 Call control messages including H.323, and whose orientation are to be more “light-
signaling, registration, and admis-
sions, and for the packetization and weight” in nature. These are listed in the table below.
synchronization of media streams
including both point-to-point and
multipoint calls
Other VoIP Protocols
H.245 Messages for opening and closing Protocol Brief Description
channels for media streams, and
other commands, requests, and SGCP (Simple Gateway Control Simple UDP-based protocol for
indications Protocol) managing endpoints and connec-
tions between endpoints.
H.261 Video codec for audio visual
services at multiples of 64 Kbps SAP (Session Announcement Protocol used by multicast session
Protocol) managers to distribute a multicast
H.263 Specifies a codec for video over session description to a large
the PSTN group of recipients
G.711 Audio codec for 3.1 Kbps band- SIP (Session Initiation Protocol) Protocol used to invite an indi-
width over 48,56, and 64 K bps vidual user to take part in a point-
channels (normal telephony) to-point or unicast session
G.722 Audio codec for 7 Kbps bandwidth RTSP (Real-Time Streaming Protocol used to interface to a
over 48,56, and 64 Kbps channels Protocol server that will provide real-time
data
G.728 Audio codec for 3.1 Kbps band-
width over 16 Kbps channels SDP (Session Description Describes the session for SAP, SIP
Protocol) and RTSP
G.723, G.723.1 Audio codec for 3.1 Kbps band-
width over 5.3 and 6.3 Kbps chan-
nels (G.723.1 has been selected by
the VoIP Forum for use with VoIP) An embedded VoIP software solution should be
designed with well-defined interfaces between the
G.729, G.729a Audio codec for 3.1 Kbps band-
width over 8 Kbps channels modules (for example, the interface between the Voice
(adopted by the Frame Relay Forum Processing performed on a DSP and the rest of the
for voice over Frame Relay)
system must be clearly defined). This also allows the
T.120 Data and conference control same device to be configured to work with IP, Frame
Relay, or ATM without a complete re-design.
Implementing VoIP in Systems Some of the functions that are required for a VoIP
system include:
The deployment of a VoIP infrastructure for public a) Fault Management: One of the most critical
use involves much more than simply adding compression tasks of any telecommunications management
functions to an IP network. Anyone must be able to call system is to assist with the identification and reso-
anyone else, regardless of location and form of network lution of problems and failures. Full SNMP
attachment (telephone, wireless phone, PC, or other management capabilities using MIBs should be
device). Everyone must believe the service is as good as provided for enterprise-level equipment.
the traditional telephone network. Long-term costs (as Integrating the management facilities of the tele-
opposed to simply avoiding regulatory costs) must make phone and data systems using TMN-based stan-
the investments in the infrastructure worthwhile. Any dards is essential for carrier-class systems.
new approach to telephony will naturally be compared to b) Accounting/Billing: VoIP gateways must keep
the incumbent and must be seen as being no worse (i.e., track of successful and unsuccessful calls. Call
the telephone still has to work if the power goes off), detail records that include such information as call
implying that all necessary management, security, and start/stop times, dialed number, source/destination
reliability functions are included. IP address, packets sent and received, etc. should
Figure 6 is a refinement of Figure 1 that includes be produced. This information would preferably
the placement of the VoIP gateway and the system be processed by the external accounting packages
level support functions that are integral to a high- that are also used for the PSTN calls. The end
quality VoIP system. The VoIP Gateway is shown here user should not need to receive multiple bills.
as a separate component, but it could also be c) Configuration: An easy-to-use management
integrated into the voice switch (a PBX or CO Switch) interface is needed to configure the equipment
or into an IP Switch. (even while the service is running). A variety of
parameters and options are involved. Examples
PC with Fax include: telephony protocols, compression
Software
IP algorithm selection, dialing plans, access controls,
PC with Voice Switch/
Software Router PSTN fallback features, port arrangements,
Directory IP
IP Enabled Security
Network Internet timers, etc.
Phone
Network d) Addressing/Directories: Telephone numbers
Management
VO/P and IP addresses need to be managed in a way
Gateway
that is transparent to the user. PCs that are used
Operations
Support for voice calls may need telephone numbers, IP-
Systems
enabled telephones will need IP addresses (or at
Phone
Voice Telephone least access to one via DHCP protocols) and
Switch Network
Fax Internet Directory services will need to be
extended to include mappings between the two
Figure 6: A Combined PSTN/VoIP System types of address.
e) Authentication/Encryption: VoIP offers the VoIP systems must be flexible enough to grow to
potential for secure telephony by making use of very large user populations, to allow a mix of
the security services available in TCP/IP environ- public and private services and to adapt to local
ments. Access controls can be implemented using regulations. The need for large numbers of
authentication and calls can be made private using addressable points may force the use of improved
encryption of the links. Internet protocols such as IPv6.
• Accessibility: Telephone systems assume that
Implementations of full-scale VoIP systems must any telephone to call any other telephone and to
provide all the “-abilities” that are usually taken for allow conferencing of multiple telephones across
granted in open systems (including the PSTN). These wide areas. This will be driven by functions that
include: map between telephone numbers and other types
• Interoperability: In a public networking envi- of packet network address, specifically IP
ronment different products will need to interwork addresses. There must, of course, exist gateways
if any-to-any communications is to be possible. that allow every device to be reachable.
Using common software that has been tested for • Viability: Many are claiming significant
conformance to all applicable standards (such as economic advantages to the implementation of
for compression) can significantly reduce the cost VoIP. These are often based on flat rate prices for
of product development. Interconnection of VoIP Internet service, the fact that services such as the
to the PSTN also involves meeting the specific “Internet 911” are not required and that there is
standards for telephone network access. no regulatory prohibition against interconnection
• Reliability: The VoIP network, whether by of telephone systems with IP systems. Also
design or through management, should be fault assumed is that higher performance compression
tolerant with only a very small likelihood of will not be used in the telephone network to
complete failure. In particular, the gateway reduce costs. If circumstances change, the motiva-
between the Telephone and VoIP systems needs to tion for VoIP purely for cost avoidance reasons
be highly reliable. may change also.
• Availability: Sufficient capacity must be avail-
able in the VoIP system and its gateways to mini-
mize the likelihood of call blocking and mid-call
disconnects. This will be especially important
when the network is shared with data traffic that
may cause congestion. Mechanisms for admission
control should be available for both the voice and
data traffic, with prioritization policies set.
• Scalability: There is potential for extremely high
growth rates in VoIP systems, especially if they
prove the equal of the PSTN at much lower cost.
“Our primary philosophy was that the issue what customers are seeking. We view Telogy as an
shouldn’t be whether information travels across IP, extension of our development arm. Moving forward,
Frame Relay, or ATM networks,” recalled Compagoni. we feel we can do some unique things together.”
“We really didn’t want the focus of our product to be Entrata has incorporated Telogy’s Golden
on the network transport technology.” Gateway™ Voice over IP (VoIP) embedded communi-
Compagnoni related Entrata’s opportunity to the cations software into the company’s debut multiservice
role of a service provider. Where a service provider can access line. The product is designed for use on copper
offer a variety of services to users and enable them to or fiber-optic lines for circuit-switched, packet based, or
connect to their network of choice, Entrata purported ATM networks, to facilitate high bandwidth connec-
to deliver those same kinds of options to the market tivity between major user sites and backbone digital
with its proposed integrated access device. networks utilized worldwide. Entrata’s products provide
Analysis of the highly competitive market revealed a single access point to a diverse variety of services and
that most of Entrata’s competitors were focusing on applications over a single integrated circuit.
individual aspects of product development. For This partnership gives Entrata’s customers the
example, some companies were developing Voice over benefits of Telogy’s market leading Golden Gateway
IP applications while others were producing ATM-only VoIP embedded software, which boasts the highest
products. By pledging to deliver megabit access to quality sound and reliability available in the market
anyone and everyone, Entrata’s mission was to develop today.
a single access product designed to allow users to Golden Gateway software, embedded on a digital
connect to any network. signal processor (DSP) and microprocessor, provides
“Partnering with a software company like Telogy communications equipment developers scalable Voice
Networks was key to our success,” Compagnoni and Fax over IP, ATM, or Frame Relay solutions.
explained. “It enabled us to go off and focus on our Telogy’s multi-function, multi-channel software capa-
own core competencies, and most importantly, it saved bilities support up to four simultaneous channels and
us critical time to market in developing our product.” allow multiple voice and fax calls to be dynamically
One of Entrata’s major challenges in developing a allocated and processed in a single DSP.
Voice over Packet-enabled networking product was Telogy’s software combined with Entrata’s high
finding the key resources to work on the project. By performance architecture and multiple transport
working with an outside vendor like Telogy and not protocol support provides a “top notch solution,”
having to assemble an engineering team with the right remarked Campganonni. “Since we’re representing
experience and qualifications, Compagnoni estimated Telogy’s software in our system, it has to work and it
that it saved the company at least twelve months. has to be good. That’s why the decision to incorporate
“The key things we were looking for were a high Golden Gateway into our product was so critical.”
quality partner, good support capabilities, and a soft-
ware solution equal to or better than what we would
have developed in-house,” said Compagnoni. “Market
data has shown us that in our industry, delivering high-
performance hardware with top quality software is
one host to transfer files to and from another host over International Telecommunications Union-
a network. Telecommunications Standards Sector (ITU-
TSS)—The new name for CCITT. An international
Foreign Exchange Office (FXO)—A remote
standards body which is a committee of the ITU, a
Telephone Company Central Office used to provide
UN treaty organization.
local telephone service over dedicated circuits from that
office to the user’s local central office and premises. Internet—(note the capital “I”) The largest internet
in the world consisting of large national backbone nets
Foreign Exchange Station (FXS)—That user
(such as MILNET, NSFNET, and CREN) and a
premises to which a foreign exchange circuit is
myriad of regional and local campus networks all over
connected.
the world. The Internet uses the Internet protocol
Frame Relay—High-performance interface for suite. To be on the Internet you must have IP connec-
packet-switching networks. Considered more efficient tivity, i.e., be able to Telnet to or ping other systems.
than X.25 which it is expected to replace. Frame relay Networks with only e-mail connectivity are not actually
technology can handle “bursty” communications that classified as being on the Internet.
have rapidly changing bandwidth requirements.
Internet Protocol (IP)—A Layer 3 (network layer)
H.323—A standard approved by the International protocol that contains addressing information and
Telecommunication Union (ITU) that defines how some control information that allows packets to be
audiovisual conferencing data is transmitted across routed. Documented in RFC 791.
networks. In theory, H.323 should enable users to
Internet Service Provider (ISP)—(1) Any of a
participate in the same conference even though they
number of companies that sell Internet access to indi-
are using different videoconferencing applications.
viduals or organizations at speeds ranging from 300
Although most videoconferencing vendors have
bps to OC-3. (2) A business that enables individuals
announced that their products will conform to H.323,
and companies to connect to the Internet by providing
it’s too early to say whether such adherence will actu-
the interface to the Internet backbone.
ally result in interoperability.
Internet Telephony—Generic term used to describe
Implementation Agreement—The formal vendor
various approaches to running voice telephony over IP.
agreement specifying the details of a system deploy-
ment. Internetwork—A collection of networks intercon-
nected by routers that function (generally) as a single
Interexchange Carrier (IXC) or Interexchange
network. Sometimes called an internet, which is not to
Common Carrier—(1) Any individual, partnership,
be confused with the Internet.
association, joint-stock company, trust, governmental
entity, or corporation engaged for hire in interstate or Intranet—A private network inside a company or
foreign communication by wire or radio, between two organization that uses the same kinds of software that
or more exchanges. (2) A long-distance telephone you would find on the public Internet, but that is only
company offering circuit-switched, leased-line or for internal use. As the Internet has become more
packet-switched service or some combination. popular, many of the tools used on the Internet are
being used in private networks; for example, many
companies have Web servers that are available only to MUX— A multiplexing device. A mux combines
employees. multiple signals for transmission over a single line. The
signals are demultiplexed, or separated, at the receiving
ISDN BRI—A digital access line that is divided into
end.
three channels. Two of the channels, called B channels,
operate at 64 Kbps and are always used for data or Off-Hook—The active condition of Switched Access
voice. The third D channel is used for signaling at 16 or a Telephone Exchange Service line.
Kbps.
On-Hook—The idle condition of Switched Access or
ISDN PRI—Based physically and electrically on an a Telephone Exchange Service line.
E1 circuit, but channelized so that two channels are
Operations Support System (OSS)—The
used for signaling and 30 channels are allocated for
computerized platform and related software used to
user traffic. ISDN PRI is available in E1 and T1 frame
support the operations of a network.
formats, depending on country.
Overhead (OH)—Bits in frame or cell required for
Latency—The delay between the time a device
framing, CRC, routing, etc.
receives a frame and the frame is forwarded out of the
destination port. Packet—(1) A logical grouping of information that
includes a header and (usually) user data. (2)
Local Area Network (LAN)—A network covering a
Continuous sequence of binary digits of information is
relatively small geographic area (usually not larger than
switched through the network and an integral unit.
a floor or small building). Compared to WANs, LANs
Consists of up to 1024 bits (128 octets) of customer
are usually characterized by relatively high data rates.
data plus additional transmission and error control
(2) Network permitting transmission and communica-
information.
tion between hardware devices, usually in one building
or complex. Packet Loss Rate—The measure loss, over time, of
data packets as a percentage of the total traffic trans-
Management Information Base (MIB)—A data-
mitted.
base of information on managed objects that can be
accessed via network management protocols such as Permanent Virtual Circuit (PVC)—Virtual circuit
SNMP and CMIP. that is permanently established. PVCs save bandwidth
associated with circuit establishment and tear down in
Mean Opinion Scores (MOS)—A system of
situations where certain virtual circuits must exist all
grading the voice quality of telephone connections.
the time.
The MOS is a statistical measurement of voice quality,
derived from a large number of subscribers judging the Plain Old Telephone System (POTS)—What we
quality of the connection. consider to be the “normal” phone system, used with
modems. Does not include leased lines or digital lines.
Million Instructions Per Second (MIPS)—
A measure of a computer’s speed or power. Private Branch Exchange (PBX)—A small tele-
phone network for customer premises. Provides local
connectivity and switching and connections to the wide RTP Control Proctocol (RTCP)—A protocol
area voice network. providing support for applications with real-time prop-
erties, including timing reconstruction, loss detection,
Protocol—(1) A formal description of a set of rules
security, and content identification. RTCP provides
and conventions that govern how devices on a network
support for real-time conferencing for large groups
exchange information. (2) Set of rules conducting
within an Internet, including source identification and
interactions between two or more parties. These rules
support for gateways (like audio and video bridges) and
consist of syntax (header structure) semantics (actions
multicast-to-unicast translators.
and reactions that are supposed to occur) and timing
(relative ordering and direction of states and events). Switched Virtual Circuit (SVC)—Virtual circuit
(3) A formal set of rules. that is dynamically established on demand and is torn
down when transmission is complete. SVCs are used in
Protocol Stack—Related layers of protocol software
situations where data transmission is sporadic.
that function together to implement a particular
communications architecture. Examples include Time-Division Multiplexing (TDM)—Technique
AppleTalk and DECnet. in which information from multiple channels can be
allocated bandwidth on a single wire-based on preas-
Public Switched Telephone Network (PSTN)—
signed time slots. Bandwidth is allocated to each
General term referring to the variety of telephone
channel regardless of whether the station has data to
networks and services in place worldwide.
transmit.
Pulse Code Modulation (PCM)—Transmission of
Transmission Control Protocol/Internet
analog information in digital form through sampling
Protocol (TCP/IP)—(1) The common name for the
and encoding the samples with a fixed number of bits.
suite of protocols developed by the U.S. Department of
QSIG—Signaling system between a PBX and CO, or Defense in the 1970s to support the construction of
between PBXs uses to support enhanced features such world-wide internetworks. TCP and IP are the two
as forwarding and follow me. best-known protocols in the suite. TCP corresponds to
Quality of Service (QoS)—Measure of Layer 4 (the transport layer) of the OSI reference
performance for a transmission system that reflects its model. It provides reliable transmission of data. IP
transmission quality and service availability. corresponds to Layer 3 (the network layer) of the OSI
reference model and provides connectionless datagram
Real-Time Transport Protocol (RTP)—The stan- service. (2) The collection of transport and application
dard protocol for streaming applications developed protocols used to communicate on the Internet and
within the IETF. other networks
Resource Reservation Protocol (RSVP)—A Unspecified Bit Rate (UBR)—QoS class defined
protocol that supports the reservation of resources by the ATM Forum for ATM networks. UBR allows
across an IP network. Applications running on IP end any amount of data up to a specified maximum to be
systems can use RSVP to indicate to other nodes the sent across the network, but there are no guarantees in
nature (bandwidth, jitter, maximum burst, and so on) terms of cell loss rate and delay.
of the packet streams they wish to receive.
NOTES NOTES
46 • Notes Notes • 47
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