IP over Voice: Transmitting Data Over an POTS

Network and VOIP

Masters Project Report

May 2003

Advisor
Prof. Ausif Mahmood
Associate Professor
Dept. of Computer Science & Engineering
University of Bridgeport
 Submitted by
Parth Dave
S.Id 459802
Contents

Abstract & Introduction 3

1) Overview of the PSTN

A) The Beginning of the PSTN 4
B) Drawbacks to the PSTN 4

2) VOIP: Transmitting Voice Over an IP Network

A) What is VOIP? 5
B) Why do we need Voice over IP? 5
C) A Basic VOIP System Architecture 6

3) Identification of major system components 6

4) VOIP product development issues 8

5) Protocols for VOIP 12

6) Four ways to do VOIP 13

A) Gateways: popular, but lack features
B) IP PBXs: great features, scalability lacking
C) Converged appliances: simplified management
D) Other options: mix and match

7) VOIP SERVICES 15
A) PC to Phone Services
B) PC to PC services
C) Phone to Phone Services
D) Network Services

8) Data over POTS Case Study: Hotel Internet Access 17

9) Frequently Asked Questions (FAQ) 22

Conclusion 29

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References / Books 29

WebPages 30
ACKNOWLEDGEMENTS

I sincerely thank Prof. Ausif Mahmood for his invaluable guidance and support
through the course of my project.

Abstract
With massive advances in technology the computational power available to us
has increased manifold. As a result of this it has become possible for us to
transmit data over Plain Old Telephone System (POTS), which is a circuit-
switched network, and voice over an IP network, which is a packet switched
network. Since voice traffic maps directly to a circuit switched network, and not,
onto a packet switched network therefore to be able to transmit voice data over a
packet switched network entails several complications, which must be dealt with
before we can use the Internet for transmitting real time audio traffic.
This project covers the basics of a VOIP network and the current market
products and trends. The purpose of this paper is to study some of these
complications namely delay, packet loss, jitter, encoding and to present some of
the solutions to these problems. This is a broad overview of VOIP and is meant
to give the reader an idea of the issues involved in trying to cater to voice traffic
in an IP network.

Introduction:

In this paper I shall present a broad overview of the different complications that
arise in trying to provide Data over POTS and Voice traffic in the current Internet.
The reason why these complications arise is due to the fact that voice traffic has
certain differing characteristics and requires much more stringent ‘Quality of
Service Guarantees’.
We begin this paper by explaining what we mean by VOIP, and then we go on to
discuss whether it is wise to cater for voice traffic in a packet switched network in
the first place. We then describe a basic VOIP architecture and describe its
working. Then we go on to analyze the various characteristics of voice traffic,
which need to be taken into account when we discuss its transmission in a

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packet switched network. In this section we discuss it’s delay properties,
temporal properties, response to packet loss, bandwidth management, encoding,
jitter, silence suppression and echo cancellation. We shall present how all these
characteristics present a problem in voice transmission in an IP network and how
they may be provided for by employing certain mechanisms.

The Beginning of the PSTN

The Public Switched Telephone Network (PSTN) has been evolving ever since
Alexander Graham Bell made the first voice transmission over wire in 1876. The
existing PSTN does not fit all the needs of its builders or users. After you
understand where today’s PSTN is lacking, you will know where to look to find a
solution. This section sets the stage for why the voice and data networks are
merging into a single network.

Drawbacks to the PSTN

Although the PSTN is effective and does a good job at what it was built to do
(that is, switch voice calls), many business drivers are striving to change it to a
new network, whereby voice is an application on top of a data network. This is
happening for several reasons:
• Data has overtaken voice as the primary traffic on many networks built for
voice. Data is now running on top of networks that were built to carry voice
efficiently. Data has different characteristics, however, such as a variable use
of bandwidth and a need for higher bandwidth. Soon, voice networks will run on
top of networks built with a data-centric approach. Traffic will then be
differentiated based upon application instead of physical circuits. New
technologies (such as Fast Ethernet, Gigabit Ethernet, and Optical Networking)
will be used to deploy the high-speed networks that needed to carry all this
additional data.
• The PSTN cannot create and deploy features quickly enough.
• Data/Voice/Video (D/V/V) cannot converge on the PSTN as currently built.
With only an analog line to most homes, you cannot have data access
(Internet access), phone access, and video access across one 56-kbps
modem. High-speed broadband access, such as digital subscriber line (DSL),
cable, or wireless, is needed to enable this convergence
• The architecture built for voice is not flexible enough to carry data.

It is also important to note that circuit-switched calls require a permanent 64-kbps

dedicated circuit between the two telephones. Whether the caller or the person
called is talking, the 64-kbps connection cannot be used by any other party. This
means that the telephone company cannot use this bandwidth for any other
purpose and must bill the parties for consuming its resources.
Data networking, on the other hand, has the capability to use bandwidth only
when it is required. This difference, although seemingly small, is a major benefit
of packet-based voice networking.

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What is VOIP?

We must begin by discussing what is Voice over IP? By voice over IP we mean
the transmission of real time voice signals over an IP network namely the
Internet. The question then arises that what is the big deal in transmitting voice
signals over a packet switched network? Well if we look closely at the underlying
architecture of the Internet, it becomes apparent that to be able to provide voice
service over an IP network needs a lot of different mechanisms to be built into
the IP networks and also in the end hosts. The reason for this is that the service
requirements of VOIP don’t map onto the Internet in an inherent way. The current
for this is that current Internet provide only best effort service, whereas Voice
transmission over the Internet require much more stringent guarantees from the
Internet. As a result of which we need to build mechanisms over the current
architecture to provide these services to the voice applications. The specific
requirements and characteristics of Voice traffic are discussed later in the paper.
We must mention here the difference between multimedia voice transmission
and voice transmission over IP. By VOIP we mean real time voice transmission.
The voice signal at the sender is encoded and sent to some receiver where the
receiver immediately replays the packets as soon as it receives it. All this
happens in real time. Whereas in a multimedia system, the receiver may receive
the whole voice segment and store it locally and then play it back whenever the
user requests it. The analogy would be something like a voice telephone call with
a voice mail.

Why do we need Voice over IP?

We mentioned that voice traffic has characteristics which are different from
traditional traffic carried by the Internet and to provide for voice traffic is not a
trivial issue. Then the natural question is whether we should carry voice traffic in
the Internet in the first place? After all we already have a telephone system in
place, why do we need to transmit voice over the Internet? The answer to this
lies in the massive advantages which can be achieved from multiplexing voice
and the traditional data traffic. The inherent nature of both this type of traffic is
such that their presence on a single wire is both complementary. One can just
imagine the massive cut in spending, we could have long distance calls at the
price of local ones. In the current telephone network, currently when we place a
call, the required bandwidth is reserved for the entire path for the duration of the
call and even if the speaker is not saying anything, the bandwidth is still locked
with that person and no body else can use it. As a result of this, for a given

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bandwidth, without statistical multiplexing, the total number of users that can be
supported is much less than of we could use statistical multiplexing .

The advance in technology, both in the network and in the end host, allows us to
empty encoding techniques and time stamping and sequencing without incurring
too much delay so that voice can be transmitted to the other side with acceptable
delays. There are various value added services that can be provided if we use
the Internet for voice transmission. Thus weighing the pros and cons it clearly
stands out that there are massive advantages to be had from integrating voice in
an IP network .

A Basic VOIP System Architecture:

A basic VOIP architecture would have a host where the voice signal must be
compressed, coded and inserted into packets, have a sequence number and
time stamps and sent to the receiver where they must be received and stored in
a payoff buffer and then the signal recreated based on the time stamps and
relative positions of things. VOIP services architecture either be PC-PC or PC-
phone or vice versa. A basic of scenarios of voice over IP in the PC-PC
architecture may be that voice signals at some host are encoded as soon as they
are produced and are sent to the remote machines IP where the remote machine
on receiving the packet decodes the packet and sends it immediately to the
process. An alternative scenario might be a PC calling a remoter telephone. We
specify the telephone number and the remote telephone numbers are mapped to
IP address of the gateway closest to the receiver. The sender address the IP
packets to remote gateway’s IP and remote gateway on receiving the packets
decodes them and sends an analogue signal to the remote telephone. Similarly
in the telephone-PC architecture the gateway would do the packetization and
encoding of data in packets.

IDENTIFICATION OF MAJOR SYSTEM COMPONENTS

1 Gateway

The gateways are the devices that communicate between the telephone signals
and the IP endpoint. The IP endpoint usually speaks H.323 for media stream and
more recently Session Initiation protocol (SIP). The gateways usually perform
the following 6 functions

• Search function

When an IP gateway is used to place a call across an IP network, it

receives a called party phone number. It converts it into the IP address of
the far end gateway, possibly through a table lookup in the originating
gateway or in a centralized directory server.

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 • Connection Function

The originating gateway establishes a connection to the destination

gateway, exchanges call setup, compatibility information and performs
any option negotiation and security handshake.

• Digitizing function

Analog telephone signals coming into a trunk on the gateway are digitized
by the gateway into a format useful to the gateway, usually 64 kbps PCM.
This requires the gateway to interface to a variety of Telephone-signaling
conventions.

• Demodulation functions

With some gateways the gateway trunk can accept only a voice signal or
a fax signal but not both. But sophisticated gateways handle both. When
the signal is a fax, it is demodulated by the DSP back into the original 2.4-
14.4 kbps digital format. This is then put into the IP packets for
transmission. The demodulated information is remodulated back to the
original analog fax signal by the remote gateway, for delivery to the
remote fax machine.

• Compression functions

When the signal is determined to be voice, it is usually compressed by a

DSP from 64K PCM to a 5.3 Kbps signal.

• Decompression and Remodulation functions

At the same time that the gateway performs steps 1-5, it is also receiving
packets. Hence this function is required

2 Gatekeepers

Terminals are the LAN client endpoints that provide real time two-way
communications. When an endpoint is switched on, it performs a multicast
discovery for a gatekeeper and registers with it. Thus the gatekeeper knows how
many users are connected and where they are located. The collection of a

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gatekeeper and its registered endpoints is called as a zone. A gatekeeper is
required to perform the following functions:

• Address translation

Translation of an alias address to a Transport Address using a table

updated via Registration messages.

• Admissions control

Authorization of LAN access, using Admissions Requests or Confirm and

Reject (ARQ/ARC/ARJ) messages. Access is based on call authorization,
bandwidth or some other criteria.

• Bandwidth management

Support for Bandwidth Request, Confirm and Reject messages, or a null

function that accepts all requests for bandwidth changes.

• Zone management

The Gatekeeper provides the above functions for terminals, MCUs, and
Gateways, which are registered in its Zone of control.

3 IP Telephones

These are devices, which replace the existing telephones by providing enhanced
services suited to VOIP. At the same time they should retain the capabilities of
the original phones to keep the user comfortable.

4 PC Software phones

This arrangement consists of a microphone connected to a PC interfaced by a

card and running a software, which permits voice and multimedia transfer over
the Internet. Microsoft NetMeeting is an example.

VOIP PRODUCT DEVELOPMENT ISSUES

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In this section we discuss the points that manufacturers have to take note of
while developing their products.

1 Voice Quality

The voice quality should be comparable to what is available using the PSTN,
even over networks of varying levels of QoS. If a company thinks that reducing
the bills is the criteria and adopts a poor quality VOIP service, then the only
people using that service would be the Managing Director and the Accounting
Officer. The employees will not compromise quality to reduce the company's
bills.

The following factors decide the VOIP quality:

• Use of a Quality CODEC

Codec stands for Coder Decoder. It should give good voice quality and
low delay. The International Telecommunication Union's (ITU's) officially
recommended CODEC for all wide area networking applications is G.729

• Echo cancellation
When a two-wire telephone cable connects to a four-wire PBX interface or
a Telco central office interface, a special electric circuit called a hybrid is
used to do the conversion. But in them a small percentage of telephone
energy is not converted but instead reflected back to the caller creating an
echo. If the delay is more than 10mS the caller hears the echo and this
has to be avoided.

• Delay
o Total Transmission Delay
Total transmission delay is the sum of the compression,
decompression delays, processing delay, the buffering/Queuing
delay, the transmission delay and the network delay. The network
delay is variable while the others can be fixed pre hand to less than
130ms. When this total delay exceeds 200ms, the two speakers
have to make sure that when one speaks the other listens and
pauses to make sure that the speaker is done. Bad timing may
result in stepping on the other's message.

o Delay Jitter
Delay jitter is the variability in arrival time of a packet. When a
packet does not arrive in time to fit into the voice stream going out

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 of the far end gateway, it has to be discarded. It cannot be re
transmitted, as it would delay proceedings too much. If this
happens too often, then the listener will perceive reduced voice
quality.

o Delay management
 VOIP Packet Prioritization
The reason VOIP works well over a corporate IP network is
due more to the corporate network's low jitter than low delay.
Corporate routers usually prioritize voice/fax packets either
by explicit programming of the router or by using a
prioritization protocol like Resource Reservation Protocol
(RSVP).

 IP Packet Segmentation
This is an important step required to ensure that a very long
data packet does not delay the voice packet from exiting the
router in a timely manner. This is achieved by programming
the router to segment all out bound data packets according
to the WAN access link.

 Packet replay technique

To allow for variable packet arrival time and still produce a
steady outgoing stream of speech, at the far end the speech
is not played as soon as the first packet arrives. Instead it is
held in the jitter buffer for some time and then played. This
adds to the overall delay. The lesser the jitter, smaller the
jitter buffer time and lower the delay.

The combination of the above three techniques produces a

VOIP friendly IP network. Such IP networks are called as
Managed IP networks.

• VOIP Forward Error Correction (FEC)

The public Internet has substantial packet corruption and loss. Packet
replay may not suffice. For this FEC can compensate for the corrupted or
missing packet.

o Intra Packet FEC

Here extra bits are added, thus allowing the receiving end to
determine which of the bits were corrupted, yielding a packet ready
for play out.

o Extra packet FEC

Here extra information is added to each packet that allows the

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 receiving gateway to extrapolate from the previously received good
packet and reconstruct the missing or severely corrupted packet

2 High Bandwidth Consumption

A telephone quality call or a toll quality call requires at least 64 kbps/call. This
bandwidth is impossible to dedicate on a data network for voice.

Speech compression techniques as the G.729 reduce this to around 8kbps. The
IP router overhead is around 7 kbps. Thus it is 15 kbps. But modern
compressors make use of an important technique called as silence
suppression. In a typical full duplex phone conversation, only 35-40% is active.
There are significant pauses between words, phrases etc. The bandwidth
consumption is thus reduced by silence suppression. Ultimately voice requires
only 5-6 kbps.

Silence suppression renders the line absolutely silent to the listener so much so
that it sounds absolutely dead. But by inserting Comfort Noise or even better, by
periodically sampling the background noise and regenerating it for the listener,
the line sounds active.

3 Transparency to the user

The user need not know what technology is being used for the call. He should be
able to use the telephone as he does right now.

• Ease of configuration

An easy to use management interface is needed to configure the

equipment. A variety of parameters and options such as telephony
protocols, compressing algorithm selections, dialing plans, access
controls, PSTN fall back features, port arrangement etc. are to be taken
care of.

• Addressing / Directories

Telephone numbers and IP addresses need to be managed in a way that

it is transparent to the user. PCs that are used for voice calls, may need
telephone numbers. IP enabled telephones IP addresses or an access to
one via DHCP protocols and Internet directory services will need to be
extended to include mappings between the two types of addresses.

4 The TCP/UDP issue

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The voice packet is constructed as a UDP/IP packet, to avoid TCP/IP's attempt
to retransmit the corrupted packet. However TCP could be a better alternative for
Fax transmission simply because if lost packets occur during the negotiation of a
page, the fax could be terminated. When TCP/IP is used and the host software
hides the retransmission from the fax machine, there will be no impact.

5 Deployment of the Gateway: Trunk Contentions

At a remote site there are normally 2 to 4 VOIP connections (or trunks) from the
VOIP gateway to the PBX allowing 2 - 4 simultaneous phone/fax connections
between the remote site and other corporate locations. The actual number of
trunks depends upon the number of calls made per day and the total amount
they consume. The number of the head quarter’s trunks is decided by the total
number of phone calls between head quarters and the remote sites and the total
number of simultaneously active calls. Usually, head quarters have a fraction of
the total trunk count. The trunk contention ratio is the ratio of total remote site
trunks to head quarter trunks.

6 Security

• Authentication/ Encryption

VOIP offers the potential for secure telephony by making use of the
services available in TCP/IP environments. Access controls can be
implemented using authentication and calls can be made private using
encryption of the links.

• Security implementation

Security features are usually implemented using four primary components:

Packet Filtering Router, Connection gateway, Address Translating firewall
and Application proxy.

Achieving security is a complex issue. An H.323 call is made up of many

different connections. In addition addresses and port numbers are
exchanged within the data stream of the next higher connection. This
makes it particularly difficult for address translating firewalls, which must
modify the addresses inside those data streams.

The firewall must be able to stand under a large number of simultaneous

connections also. Detection of intruders should be possible on the inside
and the outside of the firewall.

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7 Accounting / Billing

VOIP gateways must keep track of successful and unsuccessful calls. Call detail
records should be produced. But the major issue is the suitable billing model
selection. A number of billing models have been suggested

• Time-based - Metered by flow duration, time-of-day, time-of week

• Destination, distance, carrier-based IP - Rated by called and calling
station IDs associated with the sequence of stages used to support the
call
• QoS-based Voice over IP - reflecting established service parameters such
as priority, selected QoS, and latency.

Protocols for VOIP H.323:

H.323 is a set of protocols for voice, video and data conferencing over packet-
based networks such as the Internet. It is designed to operate above the
transport layer of the underlying network. This protocol assumes that no quality
of service is provided by LANs. H.323 defines four logical components which are:
Terminals, Gateways, Gatekeepers and Multipoint Control Units(MCU). The
terminal, gateways and MCUs are known as endpoints. H.323 terminals are the
LAN client endpoints that provide real time, two way communications. A H.323
terminal can communicate with either another H.323 terminal, a H.323 gateway
or a MCU. An H.323 gateway provides for real-time, two-way communication
between H.323 terminal on the IP network and other ITU terminals on a switched
based network, or to another H.323 gateway. Their basic function is that of a
translator i.e. they perform the translation between different transmission
formats. The gateway is the interface between the PSTN and the Internet. They
take voice from PSTN and put it on the public Internet and vice versa.
Gatekeepers are the most vital part of H.323. A gatekeeper plays the role of a
manager. It acts as the central point for all calls within its zone and provides
services to the registered endpoints. Gatekeepers also do bandwidth
management and controls admission of end points. The MCU is an endpoint on
the network that provides the capability for three or more terminals and gateways
to participate in a multipoint conference. The MCU determines the capabilities of
each terminal and sends each a mixed media stream. In the decentralized model
of multipoint conferencing, a MC ensures communication compatibility but the
media streams are multicast and the mixing is performed at each terminal.

Four ways to do VOIP

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How to install voice over IP on your enterprise network. If you want to get started
today on a transition to voice over IP, there are four options: gateways, IP
PBXs, converged appliances, and other.

1 Gateways: popular, but lack features

A voice-over-IP gateway can be loosely defined as a mechanism that takes

circuit-switched voice from a traditional PBX, converts it to IP and transfers it
across a LAN or WAN to another gateway where it is reconstituted back into a
format that is understood by the receiving phone system.
Gateway functionality can be obtained through stand-alone boxes, modules or
chassis cards for proprietary boxes; also expandable routers or software and
expansion cards for Windows NT servers.
For example, Cisco is taking a modular approach with a voice-over-IP card that
fits its 1750, 2600 and 3600 series routers. All Cisco products can easily be
equipped for voice. Cisco says voice packets can be guaranteed via quality-of-
service policy implementation on a Cisco-switched network.
Lucent, Nortel Networks and Siemens offer similar strategies for providing voice-
over-IP gateway capabilities in some form or another.
While gateways are the most popular voice-over-IP products on the market --
available from at least 30 vendors -- the key point here is that you have voice
packets running over IP. However, the packets are not running on the Internet,
and you're not gaining any of the features and capabilities you get by converging
voice and data networks.

2 IP PBXs: great features, scalability lacking

IP PBXs, such as Altigen's AltiServe and Artisoft's TeleVantage, are great if you
have the luxury of designing your system from the ground up. IP PBXs are
complete phone systems, usually with IP phone options that include many of the
IP telephony applications, such as managing your phone from your desktop PC,
multiline call control and automatic call distribution.
IP PBXs are usually NT servers with telephony software and voice cards.
Disadvantages often include scalability and a dial tone that's dependent on NT,
which doesn't offer the same uptime as a switched phone network.
Until recently, IP PBXs have mainly been targeted at small or branch offices with
100 users or less, but Alcatel recently announced OmniPCX, a voice-over-IP
system that incorporates gateway and call processing in a single device and can
accommodate up to 50,000 users. Additionally, 3Com, Lucent and Cisco have all
announced plans to provide the same type of product.
Cisco's Selsius products and 3Com's NBX series fit in this category because the
goal of both is to provide the same services as OmniPCX on a large scale.
However, while initial versions of these products are in trial stages, they have not
been proven for high numbers of users. Alcatel is the first to stake that claim, and
Cisco and 3Com will have products in the future that compete. 3Com now says
its product is only for midsize businesses with less than 500 users.

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The beauty of an IP PBX is being able to create a distributed system. For
example, Nokia's IP Telephony Gateway and Shoreline Teleworks' ShoreGear
IPBX allow you to distribute your phone system throughout an IP network, so
geographically separated phones -- with features such as direct dial, call
forwarding, conferencing and voice mail -- provide the appearance of being
connected directly to the local PBX. Alcatel, 3Com's, Lukens’s and Cisco's IP
PBXs do not offer these features.

3 Converged appliances: simplified management

Converged appliances that join phone and data networks provide the simplified
management that fulfills the promise of voice over IP. Several vendors offer such
appliances. For example, Vertical Networks' Instant Office offers call services,
voice mail, routing and LAN connect for voice and telecom, for a small to midsize
office, all included in the same box and managed together. Also, Praxon's PDX,
a modular communications platform, combines voice PBX features with a full
complement of data networking, messaging and Internet functions.

4 Other options: mix and match

Aside from stand-alone gateways, IP PBXs and converged appliances, there's

the other category we've defined. Interestingly, this other category is not
necessarily the smallest. In fact, given the number of insertion points for voice
over IP and the options available, it's probably the largest.
Lucent, for example, has a line of products that allow the convergence of your
existing voice with IP system at different points along the chain. Want to IP-
enable your existing Definity PBX? PacketStar ITS-E is the option you need.
Want IP-based trunking? You can use the Definity IP Trunk for Enterprise. Want
just IP phones? You can buy Definity IP Ethernet Telephones. Want to leave
analog phones and fax machines in place, but have an IP PBX? IP
ExchangeComm answers the call.
VocalTec also provides piece parts such as the VocalTec Gatekeeper for IP
address-to-phone number mapping. There's also VocalTec's InternetPhone, an
audio/video/PC phone and the VocalTec Telephony Gateway Series 120 and
2000, two versions of IP public switched telephone network gateways.
Additionally, Cisco, Nortel and 3Com have also outlined convergence strategies
with products that do everything from single device IP integration to complete
infrastructure replacement.

VOIP SERVICES

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With a whole range of products being launched in this field, there are a variety of
services being provided to the end user. The service basically involves
transferring voice from one end to the other. There are different ways though.

1 PC to Phone Services

These Services require a gateway on the receiving side to convert the IP packets
back to Telephone signals.

• VocalTec Surf&Call

A good example would be the VocalTec Surf&Call. It enables Web to

Phone Call center applications, promoting e-commerce. The web user
sees an icon of Surf&Call and when he clicks on that he is connected to
the phone on the other side through the Internet via VocalTec gateway
bypassing the PSTN.

• Dialpad.com

Dialpad.com has started an online VOIP service at www.dialpad.com. This

offers free of cost long distance calling service without any installation of
software through the Internet. Its revenue comes from online advertising.

2 PC-to-PC services

These can be provided without a gateway on either side.

This service is obtained by a variety of software products such as

• Microsoft NetMeeting
• VocalTec Iphone
• TaoTalk.com

It promotes video conferencing applications, Application share, White board etc.

3 Phone-to-Phone Services

A large number of Companies are providing long distance phone call services by
means of VOIP at reduced rates. Examples are:

• AT&T’s 7cents per minute any day any time offer for long distance calls in
the United States. It also offers discounted international calls on purchase
of the above offer.
• AOL offers 9cents/minute service.

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 • IDT Corporation introduced a service, which costs 8cents/minute in US.
UK-18cents, Australia 20cents, Japan 29cents/minute. These rates are
95% less than before.

A variety of calling card services to talk over long distances from anywhere,
including different countries. However in many of these services which offer low
rates, the quality is poor. But there are some, which use good gateways and
reliable billing mechanisms.

Examples:

• AcculinQ:
This offers local Access in 5 Major US Cities including: Austin, Dallas, Fort
Worth, Houston Texas & Denver Colorado at an extraordinarily low long
distance rate of 5.9 cent per minute.
Calls to France and Germany are 11.9 cents per minute.
• USATEL VIA ONE PREPAID calling Card:
This card does not charge the FCC pay phone access fee. It charges 14
cents per minute in Continental USA.

4 Network Services

Here we talk about services being offered to improve the quality of transfer of IP
packets. VOIP in a company Intranet is currently much better than that over the
public Internet. While talking about issues, we talked about the Managed IP
Network. It is believed that fiber networks will improve the quality of transfer.

A Level 3’s IP Crossroad Service

It is a nation wide IP network. This service is intended to give better multimedia

transfer across the network at reduced rates. The customer is charged
depending on the origination and the destination of traffic.

B QWest

QWest Virtual Network Service enables building a virtual private network system
for call networks to meet individual business needs. It is built with Qwest Macro
Capacity Fiber network as a backbone and advanced architecture and includes
features desired by most private users.

Case Study: Hotel Internet Access

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Network Installation Request
Determine Appropriate Internet Access Solution For Guest Usage

Requirements For Network

1. Deliver “Always-On” Internet Access At A High Speed
2. Installation Must Provide Simple Interface For:
a. Management Provisioning
b. Guest Interaction & Usage

Installations Considered

• CAT5 Installs Into Each Room For Use With Ethernet-Based System
o Install Multiple CAT 5 Lines Throughout Building Without
Exceeding 100m Limit
o Bring All Cables to Central Patch Panel For Connection To LAN
System
o Install Centralized Ethernet System (Switches, Routers, Etc.) &
Subscriber Management

• Wireless Access Solution With Omni-Directional Transceivers

o Install Multiple Access Points Inside Hotel
o Install CAT5 Links To Access Points
o Setup Sophisticated/Powerful WEP Security Policies
o Setup Effective Administration/Network Usage For Roamers
o Setup Policies To Disable Un-Authorized Nodes/Users Access
o Setup Subscriber Management System

• Data-Over-POTS (Plain Old Telephone Service) Solution

o No New Wiring, Use Existing Pairs To Deliver Signal/Access
o Allows Long-Distance Runs (Up To 1000ft.1/333m)2 For Easy
Central MDF
o Install Subscriber Management & Basic Services
(Routing/DHCP/DNS)
o Install CPE Access Modules In Each Room

Cause For Service

In the increasingly competitive hospitality market, property/hotel management

companies are continually looking for creative ways to attract and retain
guests/clients while at the same time increase the value of their properties with
heightened service offerings. Property/hotel management companies regularly
offer new services to improve the satisfaction that their guests experience during
their stay, including such amenities as fitness facilities, concierge services, meal
delivery options, and babysitting to name just a few. Today, that list is being

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expanded to include advanced High Speed Internet Access services. The
convenience of accessing an "always-on" Internet service at a high speed from
your own hotel room is greatly influencing prospective guests and clients from
choosing one hotel over another. Many business clients REQUIRE a high-speed
Internet connection for access to their corporate VPN or for Internet based
research. The average guest finds convenience and entertainment in having a
fast on-line connection at their disposal. Business travelers often are indifferent
to an additional charge for service or rates, as they are not paying for their stay
anyway.

Using Data-Over-POTS

City-Net offers hardware solutions that assist property/hotel management

companies in improving guest satisfaction, staff utilization, asset performance,
and vendor management, resulting in a seamless service delivery process from
initial customer contact through service fulfillment and supplier/vendor
management, thereby adding value to all participants in the process. Using a
DOP (Data-Over-Pots) solution enables property managers to not only provide a
value added service, but to provide a profitable one as well! With expenses being
kept to a minimal, the system can be expanded to support hundreds of users,
without substantial changes to the current infrastructure: translating to less
expense, and more money in the property manager’s pocket.
1 Field Testing Has Shown Connectivity And Flawless Transmission Up To
2000ft. Your Results May Vary.
2 City-Net’s HPNA Solutions Have Been Tested & Certified To Provide
Connectivity Up To 1000ft. Other HPNA Devices Connectivity Distance May
Vary.

Installation Site Overview

The property selected for this installation was located within close proximity to a
business community in southern New Jersey. Also very close by were
conference facilities and business centers.
Many nearby hotels began offering High-Speed access at low rates to compete
with local copy shops and other notable hotspots. This particular hotel noticed
that a large amount of their clientele was frequenting the local copy shop to
access the Internet and to perform basic document printing and copying. The
hotel also noticed an alarming amount of their clientele choosing to stay at the
surrounding properties that were offering Internet access. New clientele stopped
walking through the door, and existing repeat guests were no longer making
reservations. It became very apparent to the Hotel Management that they
needed to rectify this problem quickly, and effectively.
The solution was clearer than ever: PROVIDE INTERNET ACCESS!
Immediately the Hotel Management began submitting RFP’s for dedicated
Internet access systems. The demand was simple: provide each room with
affordable high-speed access at a minimal installation and maintenance cost. If

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possible, the hotel requested to have the ability to integrate an inexpensive yet
effective business center for their guests’ usage. The idea was to not only
compete with surrounding hotels, but with business service shops as well. The
hotel management wanted a turnkey solution to provide their guests with a
turnkey service, a one-stop shop for all of your business data and
printing/copying needs. The problem at hand was how to accomplish these
objectives without going over budget, or causing complications in future
upgrades or after-installation maintenance. The hotel currently had an MITEL
digital PBX system with 100 live extensions and 24 CO lines inbound. For the
existing equipment infrastructure, proposals were submitted to complement and
build upon the existing architecture of the wiring, as well as some proposals
specifying the deployment of new wiring.
The core of the system that varied from proposal to proposal was the
transmission system. Getting the data signal to each room affordably and quickly
was the main variable from all the submissions. Few things differed from each
proposal; all contained a T1 line card for the PBX, as well as some routing and
switching equipment. What differed greatly were the types of switching and
routing/services equipment selected, as well as the transmission mediums
(Wireless, CAT5, CAT3, Fiber).

Proposed Switching/Medium

o Cisco Systems – Ethernet Based System Using CAT 5/Fiber

o Cisco Systems – Long Reach Ethernet Over POTS3
o Cisco Systems – Aironet Solution Using 802.11B Wireless
o Bay Networks – Ethernet Based System Using CAT 5
o Tut Systems – POTS Solution Using HPNA Type Technology4
o City-Net – POTS Solution Using HPNA 1.1 Technology

The City-Net solution met all the desired results quite effectively.
However, City-Net was able to deliver a price point more attractive than any
other system. Moreover, City-Net Technology’s switches have a higher
compatibility rate as well as heightened switching and system reliability. Another
reason why City-Net was preferred over other switching solutions is the ability to
scale your installation using management concentrators and stacking
technology. City-Net switches offer convenient interfaces for the management
and control of up to 15 twelve-port switches from a single IP address.

Selected Solution

Data-Over-POTS Solution With City-Net’s HPNA Technology

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Installing a Data-Over-POTS solution is an excellent way to keep installation
times to a minimum and material costs low. Data-Over-POTS enables you to add
a Value Added Service to guests without interrupting their existing voice service.
By utilizing the existing wiring within the facility, you can offer your users high-
speed Internet access at a fraction of the cost of using an Ethernet-Based
delivery solution.
City-Net Technology delivers not only a powerful and economic way to distribute
access to the Internet, City-Net provides a foundation for a scalable, robust
managed network for hundreds of subscribers. City-Net offers one of the only
hardware platforms that enables full SNMP connectivity with a reliable and
administration free switching structure.
Now that the switching and transmission mediums were decided, the rest of the
system just fell into place. Using a relatively inexpensive subscriber management
solution from a 3rd party, the hotel was able to provide the necessary basic
network services to their guests, with a few perks. The 3rd party solution enabled
an IP PnP type connection system to ease the connection of new nodes or
guests computers onto the network without any configuration what so ever. This
technology enables anybody to plug in their PC to the hotel network, and gain
access immediately without any type of configuration settings or alterations on
the user end. Although access is limited when first allowed access to the
network, once authenticated through Radius or an internal authentication
process the node connected to the appropriate room can surf the internet freely,
and at blazingly fast speeds, all with little or no intervention on the hotel staff’s
part.
The 3rd party solution also offered many avenues for revenue generation and
free services to be offered to and by the hotel. Matching these two platforms
together, the subscriber gateway (3rd party equipment) and City-Net switches,
the hotel was able to create a powerful subscriber network, with almost no
installation time and little configuration, and the best part is that it was all done
with no new wiring!

Installation layout and Description

1) CN 2000 Intelligent Central Management Switch Hub

CN-2000 Intelligent Central Management Switch Hub enables your network to

have the ultimate in flexibility and management. The CN-2000 unit offers up to 16
Ethernet ports and 17 Console ports to expand your networking capacity by
interconnecting City-Net HomePNA switching units.
One of the main features of the CN-2000 is the ability to manage other switches
within the CN family. The CN-2000 serves as a master central intelligence
management switch hub. It can manage up to 15 sets of City-Net HPNA switches
such as the CN-1412. The slave switches should contain a DB-9 console port in
its front panel to connect to the CN-2000. The CN-2000 Console port then
connects to the slave switch via PS/2-DB-9 cable.
Port-P can serve as an Up-link or a regular Ethernet port on CN-2000, when

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the Uplink button is pressed down it serve as an Up-link port, when the Uplink
button is released it serve as a regular Ethernet port. Hence when connecting the
CN-2000 to WAN via regular Cat. 5 cable on Port-P press down the Uplink
button.
In order to provide management options and Internet access to slave units,
Console and Ethernet connections must be established from CN-2000 to the
slave units using appropriate cables. Once the two units are connected, both
units are ready to be access through Telnet for maintenances tools and ICD
commands or through Web interface for simple monitoring.

2) CN-1412 HPNA Switch

The CN-1412 HomePNA Switch has plenty of Ethernet ports, to offer you many
paths for upgrading your network, as well as 12 VLAN HomePNA ports ready to
plug in to your existing wiring system using current CAT3/RJ-11 Installations.

Some highlights of the HomePNA Switch included:

• Supports security with port based VLAN function.
• Supports Virtual LAN (VLAN) Grouping.
• Auto Noise Leveling (Automatic & Manual).
• Twelve 1Mbps HomePNA Ports
• Four 10/100 Mbps Base-TX Ethernet Ports
• One Console Port for HomePNA Switch.
• Easy installation – no new wire required inside the building.
• Easy To Use Menu System and Command Interface.
• HomePNA and Ethernet ports status Monitoring.
• Frequency division multiplexing for uninterrupted simultaneous
voice/data transmission.
• Supports Full and half duplex modes.
• HomePNA port transmission speed up to 1Mbps
• Supports 8K MAC addresses entries.

3) HomePNA to Ethernet Converter

HomePNA to Ethernet Converter consists of one HomePNA port and two
Ethernet port, and is a HomePNA to Ethernet converter. It does not require any
software, nor does it require opening the PC’s case cover. You only need plug in
two cables (RJ-11 and RJ-45), and it is very easy to use. Enjoy uninterrupted
voice service (telephone) while surf the Internet with HomePNA.

Product Specifications:
• Speed:1Mbps (phone line)
• Ports:RJ-11, two connectors
• Transmission Distance: Up to 500 ft (150 m)
• LED Indicators: Link / Activity and Collision
• Cabling: Standard Copper RJ-11

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• Communication Speed:10/100 Mbps

FREQUENTLY ASKED QUESTIONS (FAQ)

1) HOMEPNA

Q: What does HomePNA stand for?

A: HomePNA Stands for Home Phone line Networking Alliance

Q: With HomePNA, Can I use my phone or fax and my Internet

at the same time?
A: Yes, you can use them at the same time.

Q: What are the primary objectives of HomePNA?

A: The primary objectives of HomePNA are to:
" Ensure mass deployment of a consumer-friendly, low-cost, high-speed "no-
new-wires" solution for in-home, MDU & MTU, phone line-based networking.
" Develop certification standards to ensure interoperability among HomePNA
member company products from the broadest possible range of technology and
equipment vendors.

Q: What are the benefits of Do/POTS networking?

A: The benefits of Do/POTS networking include: simultaneous, shared Internet
access, printer/peripheral sharing, file and application sharing and networked
gaming. In addition, consumers can enjoy the use of each of these home
entertainment and information applications using existing wiring in the home.

2) SWITCH

Q: How many users can log into the switch through telnet at once?
A: Telnet accepts only 1 connection at a time, however with the Http interface
you can have unlimited administrators logged in. The console port supports only
one user per session.

Q: Can I turn off ports individually?

A: Yes, Use the Console or Http Interface to do so.

Q: Can I adjust bandwidth per port?

A: No, you cannot change the throughput thresholds.

Q: Do I have to set the speeds on the Ethernet ports?

A: No, the ports are fully auto-negotiating for duplex and 10/100Mbps speed.

Q: Will the switch ever get “too busy” to handle large volumes of data?

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A: No, with the flow control mechanisms and Back Pressure features in place,
the unit will always be able to operate at maximum throughput.

Q: If I turn off my HomePNA Switch, will the settings be saved?

A: YES, as long as you used the “S” save command in the ICD command
interface, all manual changes will be saved in the event that you turn off, or lose
power to your switch.

Q: Can I use one port to support multiple users?

A: Yes, the HomePNA standard supports up to 25 subscribers per port, for
shared 1Mbps access on that port, however VLAN functionality is based per port
on the switch itself. If multiple users are connected to the same port, then they
will be able to communicate with each other regardless of the VLAN setting on
the switch.

Q: I am using my HomePNA Switch for my office network; I cannot see the file-
sharing computer. Is there something wrong with my switch?
A: Not at all, just set the VLAN function to OFF and all ports will be able to
communicate with one another.

Q: Is the 1Mb of speed per port, or for the whole backplane?

A: The ports are capable of 1Mb full duplex max throughput per port; the
backplane can handle vast amounts of traffic.

3) LAN

Q: Can I connect computers using HomePNA at distances longer than 1000ft?

A: Under certain conditions it may be possible, our tests have proved 1000ft. as
acceptable.

Q: How many pairs (cable pairs) does HomePNA transmission require?

A: One pair.

Q: Which pair runs the data in a 2 pair RJ-11?

A: It will run on either pair, just be sure that both ends of the cable have the same
wiring configuration.

Q: Can I use Cat.5 Cable to make 4 RJ-11 connections?

A: Yes.

Q: Will other types of HomePNA CPE (Customer Premises Equipment) work with
the HomePNA Switch?
A: Yes, however all Manufacturers do not guarantee the performance of their
products when used with other manufacturer products.

4) WAN

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Q: What types of WAN connections are compatible with the HomePNA Switch?
A: Virtually any connection that has an Ethernet interface can be used.

Q: What router should I use for best results?

A: Any router should perform well with the HomePNA switch.

Q: My router has a firewall, how do I get through remotely to my switch to

monitor and change settings?
A: In your firewall you should have a private connection-tunneling feature, to
allow direct connections between addresses on your network and out of band
workstations.

5) PBX

Q: Will the switch work with all PBXs?

A: Your Switch will work with all analog PBXs and most Digital systems, however
not all PBX is compatible with this product, you may require a low pass filter,
contact your vendor for availability.

Q: What PBX should I use for the best results?

A: Any analog PBX should work well with our system. Because HomePNA is
based on FDM (Frequency Division Multiplexing) it sometimes conflicts with
digital PBX carrier signals

6) TROUBLESHOOTING

This section covers some common problem areas, also known fixes and
solutions. Although the solutions offered in this section should solve your
problem, occasionally a problem might arise that takes on a symptom of an
issue, hence cannot be solved in the same fashion.

A) Cross Talk Noise: Can Include Collisions and Link On/Off

Cross Talk Noise can be generated by HomePNA signals of bundled pairs of
telephone wire. When two pairs are adjacent to each other, or twisted around
each other they can create cross talk noise. A significant amount of cross talk
noise can be generated on the HomePNA switch due to the high power output of
the switch. Therefore, when telephone pairs are close together and twisted at the
switch, the adjacent port may suffer from above problem.
Available solutions for this issue are:
1. Use Cat.5 certified cables between the MDF and switches, including Cat.5
punch down blocks and shielded/booted RJ-11 connectors going into the
switch.
2. Turn on the Auto Noise Leveling Feature
3. Make manual adjustments if previous solutions failed.

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B) Reflection Noise: Caused by Un-Terminated Phone Jacks
HomePNA uses Frequency Division Multiplexing to allow simultaneous flow of
data and voice services on same pair of wires. For this reason, your phone lines
also act as data transmission lines for the frequency range of 5.5 MHz to 9.5
MHz. Therefore, if there is any open jack at the end of the circuit in HomePNA
port, the frequency information (data) will have nowhere to go, and reflect back
into the system, causing noise and non-function.
Solution:
Terminate those jacks using a Terminator or one of its specifications. Otherwise,
remove the excess jacks from the circuit.

C) HomePNA Switch Causes Inoperable or Malfunctioning PBX

There are many models of PBX worldwide today. Some PBX does not
correspond to certain standards when it comes to data transmission and FDM
(Frequency Division Multiplexing). Occasionally, when a HomePNA switch is
installed on a system with a PBX, the PBX will cease to operate or the
HomePNA switch stops functioning. The problem occurs due to the two devices
use FDM to
allow the sharing of the telephone wire. When one device is attached to the
other, the impedance values begin to change the expected frequency responses,
rendering both devices making one of them unusable.

Solution:
Install an impedance matching filter to correct the frequency domain shift.
Depending on the frequency domain, type of PBX, and amount of impedance
shift a filter may be needed between the PBX and the MDF or the CPE and
Telephone set equipment.

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Masters Project 27 Parth Dave
Installation layout Diagram

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Masters Project 29 Parth Dave
Conclusion

This broad overview of the various issues involved in the transmission of voice
over an IP network give us an insight into the kinds of mechanisms that need to
be in place in the internet if we expect to provide services of voice transmission
over the internet.
The differentiating characteristics of voice traffic that have been highlighted make
is necessary that we employ mechanisms in the internet that are able to
recognize different type of traffic and are able to provide them different services
based on their different service requirements. On thing is clear, real-time voice
traffic does not map naturally onto the packet switched network. The different
mechanisms mentioned, to take into account the characteristics of real time
voice traffic are necessary to facilitate its service in the current ‘best effort packet
switched network.’

VOIP is growing fast. The very knowledge of the applications of this technology
is enough for users and manufacturers to flock towards it. It is ideal for computer
based communications and at the same time bringing down the cost of
multimedia transfer. Hence VOIP products and services have flooded the market.
The above paper presented the features of the products of a few major game
players in the field of VOIP.

REFERENCES

Technical Papers

"Voice Over IP" .http://www.techguide.com/

Article on fundamentals of Voice over IP and issues related to quality of transfer

Readings on Voice Over IP: http://www.jeffseaman.com/reading05.html

Books B&N

Marcus Gonclaves, "Voice over IP Networks"

Uyless Black, "Voice over IP"

Jonathan Davidson, Jim Peters, "Voice Over IP Fundamentals,"

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Web Pages (For Product Information)

1. CISCO, http://www.cisco.com/

2. MICOM, http://www.micom.com

3. Lucent Technologies, http://www.lucent.com/

4. Nortel Networks, http://www.nortelnetworks.com/

5. VocalTec,http://www.vocaltec.com/

6. Nuera, http://www.nuera.com/

7. Ericsson, http://www.ericsson.com/

8. Qwest, http://www.qwest.com/

9. ITXC, http://www.itxc.com

10. Motorola, http://www.motorola.com/

11. Delta Three, http://www.telephonyworld.com/service/delta3/

12. City Netek Inc. http://www.citynetek.com/

For new products and news

http://www.VOIP-news.com/

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