Voice over IP (VoIP)
Hashir Zariwala – Mohammad Adil Khan – Saghir Malik
Students of Engineering (Telecommunication),
Bahria University Karachi Campus
Abstract
This report consists of information regarding VoIP that
how it is used to establish multimedia session (audio,
video) traffic to be transmitted over Internet Protocol.
VoIP uses communication services for transmission of
voice, fax, SMS and other voice messaging applications
via the internet rather than PSTN (Public Switched
Telephone Network). Signaling and media channel
establishment is done by two Protocols (H.323) and
SIP.
Key Words- Teleconferencing, H.323, SIP, gate
keepers, gateway, and Unified communications
1. Introduction
Voice over Internet Protocol (Voice over IP, VoIP) is a
general term used for the transmission of voice and
multimedia sessions over (IP) Internet Protocol
networks, such as the Internet. The other terms that can
be used with VoIP are IP telephony, VoBB (voice over
broadband), Internet telephony and broadband phone.
VoIP uses communication services for transmission of
voice, fax, SMS and other voice messaging applications
via the internet rather than PSTN (Public Switched
Telephone Network). The steps involved in originating
a VoIP telephone call are signaling and media channel
setup, digitization (and under some conditions
compression) of the analog voice signal, packetization
and transmission as Internet Protocol (IP) packets over
a packet-switched network. On the receiving side
similar steps reproduce the original voice stream.
Following are the steps used in establishing a VOIP
call:
 Signaling
 Media Channel Setup [1]
2. Signaling
The VoIP signaling protocols use TCP to set up,
manage and tear down the VoIP phone call. Signaling
protocols are not concerned with the actual media
stream of voice or video. Their basic functions are to
first initiate a session, then to find common ground for
communication between
the parties involved and to terminate the session when
the calls end. On a closed IP network, real time two
way communications between nodes can be managed
relatively easily. Signaling protocols take additional
responsibilities such as address translation, bandwidth
management, and authorization and in some cases make
routing decisions, while many sets of protocols have
been developed over the years to handle above
described function.
There are two types of Signaling protocol in VoIP.
 H.323
 SIP (Session initiation Protocol)
2.1 H.323
H.323 was developed in mid nineties and was originally
established for the videoconferencing over a packet
based network, but was quickly adopted for Voice over
IP. As a session layer protocol, its main function is to
perform call control and management on an IP network.
The following H.323 suite of protocols is used for
signaling purpose.
 DVB - Digital Video Broadcasting
 H.225 - Call Setup Termination
 H.235 - Security and Authentication
 RTP - Real time Transport Protocol
 RTCP - Real time Control Protocol
 H.245 - Flow control, Port Number
 Q931 - Management of Call Setup
The H323 specification relies on two additional
signaling protocols, H.225 and H.245 for call setup and
management. When a session is initiated between two
H.323 devices, the H.225 standard uses the Q931 ISDN
protocol to perform setup and teardown functions using
TCP for a reliable connection. H.245 then opens
another TCP connection to establish the capabilities of
the devices, negotiate the codec’s, and determine which
ports will be used for the session. A channel is then
opened on which the actual media will travel using
UDP for transport because of its speed and relying on
the upper layer Real Time Protocol (RTP) for
sequencing and timing information. It is important to
note that for session set up, negotiation, and
management, TCP is used for its reliability. For time
sensitive media such as voice and video, UDP is
utilized as the transport mechanism because of its speed
and low overhead. The packet size can be further
reduced by using RTP header compression, reducing
the combined IP/UDP/RTP header from over 65% to as
low as 10% of the entire packet size. Smaller means
faster, and in IP Telephony, the name of the game is
speed.
2.2 Components of H.323
An H.323 network is comprised of four logical
components, not all of which will be needed on every
network and that can reside on a wide variety of
devices. [2]
2.2.1 Terminal
A terminal is an endpoint device such as an IP
telephone, a computer running an H.323 software
application or a dedicated conferencing device. An
H.323 compliant terminal must support H.245 for
channel and capabilities negotiation, RAS (Registration,
Admission, Status), Q931 for signaling and setup and
support for RTP and RTCP on which to stream the
media. Terminals must support audio with video and
T.120 data communications being optional. [3]
2.2.4 Gatekeepers
As an optional component that is usually found in larger
Enterprise networks, the gatekeeper is the most
important component in the H.323 configuration. The
gatekeeper manages all the registered terminals,
gateways, and MCUs in a single H.323 zone, which can
span multiple LAN/WAN segments. Services such as
addressing, authorization and authentication of H.323
components, bandwidth management, accounting and
billing can all be configured on the gatekeeper. But
once they are, all endpoints must obey! The king of the
zone can make routing decisions, and can also simplify
management of multiple gateways by handling their call
control functions in a centralized manner. While a
gatekeeper can be implemented on a gateway or MCU,
in larger organizations you will usually find them on a
dedicated server (such as Microsoft’s ISA server) or on
a Cisco IOS router. [2]

2.2.2 Multiple point Control Units

An MCU provides services to allow three or more

terminals to participate in a conference. The MCU
consists of a Multipoint Controller (MC) and an
optional Multipoint Processor (MP). The MC is
responsible for H.245 functions (negotiating common
ground) while the optional MP handles the actual
mixing of media streams, and manages the streams to
avoid bandwidth contention. H.323 supports
Centralized, Decentralized, and a Hybrid concept of
multipoint conferencing. When terminals participating
in a conference reside in both a centralized and
decentralized environment (mixed) the MCU acts as a
bridge between the two.
MC and MP functions can reside on a dedicated
component, terminal, a gateway or a gatekeeper, but
when endpoints exist off network (i.e. PSTN), it is Figure1 Implementing Gatekeepers
recommended that the MC be utilized on a gateway. [2]
One big advantage that the H.323 standard seems to
have over its nearest competitor is in the area of address
2.2.3 Gateways
resolution. The gatekeeper has the ability to use a
Gateways provide a variety of services, not the least of number of methods and protocols to resolve a
which is protocol conversion between H.323 networks destination address. First, it can ask another gatekeeper.
and non-H.232, e.g. switched circuit networks (SCN). A If that doesn’t work, it can use Annex G/H.225.0, TRIP,
gateway performs call setup and teardown, translates ENUM, or DNS protocols for address resolution.
audio, video and data formats, and can perform RAS for Security enhancements to H.323 are provided by H.235,
registration with the gatekeeper. On the H.323 side, the adding authentication, encryption and integrity to the
gateway uses H.225 and H.245 for call setup and mix. Optional password based and PKI security profiles
management, and on the circuit switched side, it utilizes can be used to authenticate the person and the call
the protocols specific to SCNs such as ISDN and SS7. signaling channel can be encrypted using TLS or IPSec.
A gateway can be implemented on a gatekeeper, an [2]
MCU, or on a voice enabled router or switch.
3. SIP (Session Initiation Protocol)
SIP (Session Initiation Protocol) is another signaling
protocol used to control multimedia communication
such as voice and video calls over Internet Protocol
(IP). The protocol can be used for creating, modifying
and terminating two-party (unicast) or multiparty
(multicast) sessions consisting of one or several media
streams. The modification can involve changing
addresses or ports, inviting more participants and
adding or deleting media streams. The SIP protocol is
an Application Layer protocol designed to be
independent of the underlying transport layer; it can run
on Transmission Control Protocol (TCP), User
Datagram Protocol (UDP) or Stream Control
Transmission Protocol (SCTP). It is a text-based
protocol, incorporating many elements of the Hypertext
Transfer Protocol (HTTP) and the Simple Mail Transfer
Protocol (SMTP). Other feasible application examples
include video conferencing, streaming multimedia
distribution, instant messaging, presence
information, file transfer and online games.[4]
3.1 SIP Network Elements
SIP employs design elements similar to the HTTP
request/response transaction model. Each transaction
consists of a client request that invokes a particular
method or function on the server and at least one
response. SIP reuses most of the header fields, encoding
rules and status codes of HTTP, providing a readable
text-based format.
A SIP user agent (UA) is a logical network end-point
used to create or receive SIP messages and thereby
manage a SIP session. A SIP UA can perform the role
of a User Agent Client (UAC), which sends SIP
requests, and the User Agent Server (UAS), which
receives the requests and returns a SIP response. These
roles of UAC and UAS only last for the duration of a
SIP transaction. A SIP phone is a SIP user agent that
provides the traditional call functions of a telephone,
such as dial, answer, reject, hold/unhold, and call
transfer. SIP phones may be implemented by dedicated
hardware controlled by the phone application directly or
through an embedded operating system (hardware SIP
phone) or as a soft phone, a software application that is
installed on a personal computer or a mobile device.
E.g. A personal digital assistant (PDA) or cell phone
with IP connectivity. Each resource of a SIP network,
such as a User Agent or a voicemail box, is identified
by a Uniform Resource Identifier (URI), based on the
general standard syntax also used in Web services and
e-mail. A typical SIP URI is of the form,
sip:username:password@host:port. The URI scheme
used for SIP is sip. If secure transmission is required,
the scheme sips: is used and SIP messages must be
transported over Transport Layer Security (TLS).
In SIP the user agent may identify itself using a
message header field 'User-Agent', containing a text
description of the software/hardware/product involved.
The User-Agent field is sent in request messages, which
means that the receiving SIP server can see this
information. SIP network elements sometimes store this
information and it can be useful in diagnosing SIP
compatibility problems.SIP also defines server network
elements. Although two SIP endpoints can
communicate without any intervening SIP
infrastructure, which is why the protocol is described as
peer-to-peer, this approach is often impractical for a
public service. [4]
3.2 RFC 3261 Server Elements
A proxy server "is an intermediary entity that acts as
both a server and a client for the purpose of making
requests on behalf of other clients. A proxy server
primarily plays the role of routing, which means its job
is to ensure that a request is sent to another entity
"closer" to the targeted user. Proxies are also useful for
enforcing policy (for example, making sure a user is
allowed to make a call). A proxy interprets, and, if
necessary, rewrites specific parts of a request message
before forwarding it."
"A registrar is a server that accepts REGISTER requests
and places the information it receives in those requests
into the location service for the domain it handles."
"A redirect server is a user agent server that generates
3xx responses to requests it receives, directing the client
to contact an alternate set of URIs. The redirect server
allows SIP Proxy Servers to direct SIP session
invitations to external domains."
The RFC specifies: "It is an important concept that the
distinction between types of SIP servers is logical, not
physical."
Other SIP related network elements are
Session border controllers (SBC), they serve as middle
boxes between UA and SIP server for various types of
functions, including network topology hiding, and
assistance in NAT traversal.
Various types of gateways at the edge between a SIP
network and other networks (as a phone network). [4]
Figure2 RFC 3261 Server Element [4]
3.3 SIP Messages
SIP is a text-based protocol with syntax similar to that
of HTTP. There are two different types of SIP
messages: requests and responses. The first line of a
request has a method, defining the nature of the request,
and a Request-URI, indicating where the request should
be sent. The first line of a response has a response code.
For SIP requests, RFC 326s1 defines the following
methods:
 REGISTER: Used by a UA to indicate its
current IP address and the URLs for which it
would like to receive calls.
 INVITE: Used to establish a media session
between user agents.
 ACK: Confirms reliable message exchanges.
 CANCEL: Terminates a pending request.
 BYE: Terminates a session between two users
in a conference.
 OPTIONS: Requests information about the
capabilities of a caller, without setting up a
call.
The SIP response types defined in RFC 3261 fall in one
of the following categories.
 Provisional (1xx): Request received and being
processed.
 Success (2xx): The action was successfully
received, understood, and accepted.
 Redirection (3xx): Further action needs to be
taken (typically by sender) to complete the
request.
 Client Error (4xx): The request contains bad
syntax or cannot be fulfilled at the server.
 Server Error (5xx): The server failed to fulfill
an apparently valid request.
 Global Failure (6xx): The request cannot be
fulfilled at any server. [4]
3.4 SIP-ISUP Networking
SIP-I, or the Session Initiation Protocol with
encapsulated ISUP, is a protocol used to create, modify,
and terminate communication sessions based on ISUP
using SIP and IP networks. Services using SIP-I include
voice, video telephony, fax and data. SIP-I and SIP-
T are two protocols with similar features, notably to
allow ISUP messages to be transported over SIP
networks. [4]
4. Advantages of VoIP
4.1 Cost Saving and Free Calls
This is perhaps the most obvious. True enough, the very
nature of VoIP technology means that everyone can
make significant cost savings for their business,
especially if you have multiple branches nationwide or
overseas. Cheap calls and free calls may be the
attraction for VoIP right now, but the future will be
about value-added VoIP services, and cost will take the
backseat. [5]
4.2 Portability
One important concept to understand about VoIP is that
unlike it’s forefathers (let’s call them PSTN for now), it
is not distance or location dependent. As far as VoIP is
concerned, you could be calling your supplier 1,000
miles away in Indonesia or calling your business partner
on the other end of town, and it doesn’t make any
difference at all, in terms of connectivity and cost.
Many of our recommended VoIP service providers have
this feature. A VoIP phone number, unlike your regular
phone number, is completely portable. Most commonly
referred to as a virtual number, you can take it with you
anywhere you go. Even if you change your office
address to another state, your phone number can go
with you. You can even take your whole business with
you wherever you travel.
Integrated Communications
The best business phone systems can stand the test of
time and grow with your business needs. Businesses
simply send all of their information over their
Broadband Internet connection whether it be Internet
data from PCs or voice calls from their employees.
 Making cheap local and international phone
calls
 Audio conferencing & Video conferencing
 Have Voice messages sent to your email
 Call forwarding, call waiting
 Fax thru e-mail
 Send and receive multimedia files
 Sharing photos while talking
 And much, much more. [5]
5. Disadvantages of VoIP
Although most experts agree that the minus points of
VoIP are just a “temporary” problem that will be
eliminated as the technology goes from strength to
strength, let have a look at it anyway, and so does Voice
over IP. While the pros may be overwhelmingly
attractive, as a small business owner, you should know
the disadvantages as well. Here are some of the main
disadvantages of VoIP. [5]
5.1 Power Supply Dependency
VoIP service depends on power supply. No power, no
phone calls. This is mainly because the equipment is
hosted on your side, and not in some
telecommunications exchange like the usual PSTN
networks. However, in the future when most countries
make a complete shift to IP based networks, and most
Telco’s go 100% VoIP, this problems should by all
means cease to exist. Most PSTN lines are power-
independent simply because they have back-up power
in the exchanges, and even during blackouts you’ll be
able to make calls. This is one thing that VoIP does not
have at the moment. This disadvantage would mean that
you will need a back-up PSTN line in case of
emergencies; In fact, most businesses we have
consulted do indeed have at least one backup PSTN
phone line. [5]
5.2 Security Issues
Experts agree that security is a major concern when
choosing a VoIP solution. One of the major concerns is
“packet sniffing” means call can be “spy” without
effecting the conversation at all. However, there are
things in your control, and you can take simple steps to
make sure you’re getting the best protection that money
can buy. [5]
5.3 Less Quality Control
The basics of VoIP are very different from regular
PSTN, which uses “C7 signals” for controlling quality
of service. Due to the nature of VoIP, your calls are
streamed by packets to the destination, and any
inconsistency would mean issues like jitter, packet loss
and echo. These problems, while posing some
considerable inconvenience a few years back, are being
eliminated even as you’re reading this. [5]
5.4 Bandwidth Dependent
In any small business or home office setting, you’ll
typically have one broadband line which is shared by
multiple users, for downloading data, sending emails,
and viewing web sites and multimedia applications.
Add a VoIP system to that and your bandwidth will
soon be sucked dry. The only way to solve this problem
is to have a dedicated E1 (larger user base) or at least a
dedicated broadband connection. Home offices may not
face this problem if there are less than 3 simultaneous
users. [5]
6. Telepresence
Tele presence refers to a set of technologies which
allow a person to feel as if they were present, to give
the appearance that they were present, or to have an
effect, via telerobotics, at a place other than their true
location.
Tele presence requires that the users' senses be provided
with such stimuli as to give the feeling of being in that
other location. Additionally, users may be given the
ability to affect the remote location. In this case, the
user's position, movements, actions, voice, etc. may be
sensed, transmitted and duplicated in the remote
location to bring about this effect.
Therefore information may be traveling in both
directions between the user and the remote location.
A popular application is found in Tele presence
videoconferencing, a higher level of video telephony
which deploys greater technical sophistication and
improved fidelity of both video and audio than
in traditional videoconferencing. [6]
6.1 Implementation
Telepresence has been described as the human
experience of being fully present at a live real-world
location remote from one's own physical location.
Someone experiencing video Telepresence would
therefore be able to behave, and receive stimuli, as
though part of a meeting at the remote site. The
aforementioned would result in interactive participation
of group activities that would bring benefits to a wide
range of users
To provide a Telepresence experience, technologies are
required that implement the human sensory elements of
vision, sound, and manipulation. [6]
6.1.1 Vision
A minimum system usually includes visual feedback.
Ideally, the entire field of view of the user is filled with
a view of the remote location, and the viewpoint
corresponds to the movement and orientation of the
user's head. In this way, it differs from
television or cinema, where the viewpoint is out of the
control of the viewer.
In order to achieve this, the user may be provided with
either a very large (or wraparound) screen, or small
displays mounted directly in front of the eyes. The latter
provides a particularly convincing 3D sensation. The
movements of the user's head must be sensed, and
the camera must mimic those movements accurately
and in real time. This is important to prevent unintended
motion sickness. Another source of future improvement
to Telepresence displays, compared by some
to holograms, is a projected display
technology featuring life-sized imagery. [6]
6.1.2 Sound
Sound is generally the easiest sensation to implement
with high fidelity, based on the
foundational telephone technology dating back more
than 130 years. Very high-fidelity sound equipment has
also been available for a considerable period of time,
with stereophonic sound being more convincing
than monaural sound. [6]
6.1.3 Manipulation
The ability to manipulate a remote object or
environment is an important aspect of real Telepresence
systems, and can be implemented in large number of
ways depending on the needs of the user. Typically, the
movements of the user's hands (position in space, and
posture of the fingers) are sensed by wired
gloves, inertial sensors, or absolute spatial position
sensors. A robot in the remote location then copies
those movements as closely as possible. This ability is
also known as teleportation.
The more closely the robot re-creates the form factor of
the human hand, the greater the sense of Telepresence.
Complexity of robotic effectors varies greatly, from
simple one axis grippers, to fully anthropomorphic
robot hands.
Hap tic teleportation refers to a system that provides
some sort of tactile force feedback to the user, so the
user feels some approximation of the weight, firmness,
size, and/or texture of the remote objects manipulated
by the robot. [6]
6.2 Transparency of Implementation
A good Telepresence strategy puts the human factors
first, focusing on visual collaboration solutions that
closely replicate the brain's innate preferences for
interpersonal communications, separating from the
unnatural "talking heads" experience of traditional
videoconferencing. These cues include life–size
participants, fluid motion, accurate flesh tones and the
appearance of true eye contact. This is already a well-
established technology, used by many businesses today.
Telepresence to teleporting from Star Trek, and said
that he saw the technology as a potential
billion dollar market for Cisco.
Rarely will a Telepresence system provide such a
transparent implementation with such comprehensive
and convincing stimuli that the user perceives no
differences from actual presence. But the user may set
aside such differences, depending on the application.
The fairly simple telephone achieves a limited form of
Telepresence using just the human sensory element of
hearing, in that users consider themselves to be talking
to each other rather than talking to the telephone itself.
Watching television, for example, although it stimulates
our primary senses of vision and hearing, rarely gives
the impression that the watcher is no longer at home.
However, television sometimes engages the senses
sufficiently to trigger emotional responses from viewers
somewhat like those experienced by people who
directly witness or experience events. Televised
depictions of sports events, or disasters such as
the September 11 terrorist attacks, can elicit strong
emotions from viewers.
As the screen size increases, so does the sense of
immersion, as well as the range of subjective mental
experiences available to viewers. Some viewers
have reported a sensation of genuine vertigo or motion
sickness while watching IMAX movies of flying or
outdoor sequences.
Because most currently feasible Telepresence gear
leaves something to be desired; the user must suspend
disbelief to some degree, and choose to act in a natural
way, appropriate to the remote location, perhaps using
some skill to operate the equipment. In contrast, a
telephone user does not see herself as "operating" the
telephone, but merely talking to another person with it.
[6]
7. Advantages
An industry expert described some benefits of
Telepresence: "There were four drivers for our decision
to do more business over video and Telepresence. We
wanted to reduce our travel spend, reduce our carbon
footprint and environmental impact, improve our
employees' work/life balance, and improve employee
productivity."
Rather than traveling great distances in order to have a
face-face meeting, it is now commonplace to instead
use a Telepresence system, which uses a multiple codec
video system (which is what the word "Telepresence"
most currently represents). Each member/party of the
meeting uses a Telepresence room to "dial in" and can
see/talk to every other member on a screen as if they
were in the same room. This brings enormous time and
cost benefits. It is also superior to phone conferencing
(except in cost), as the visual aspect greatly enhances
communications, allowing for perceptions of facial
expressions and other body language. [6]
8. Unified Communication
Unified communications (UC) -- also called unified
messaging or UM -- is the new buzzword in the IT
industry, but what does it really mean? In some cases, it
depends on whom you ask; vendors tend to put their
own spin on the definition depending on what they're
trying to sell you. But by most definitions, UC refers to
the ability to integrate different types of
communications -- including voice mail, e-mail, faxes,
instant messages, and video conferencing -- into one
common interface and/or repository.
Unified Communications is NOT necessarily a product;
Unified Communications is really a STRATEGY!
A strategy to dramatically reduce the effort needed to
establish effective communications between people,
whether they are colleagues, partners or customers.
Effective communications is accomplished over the
most appropriate medium to reach the right person, with
the right device, the very first time!
That said, there are many ways to implement a unified
communications solution in an organization. In its
simplest form, it provides a way for users to access their
faxes and voice mail messages via their e-mail clients.
More sophisticated implementations provide advanced
features such as the ability to hear e-mail messages read
to you over the phone as well as the ability to dictate a
reply and send it as an e-mail, instant message, fax, or
audio message. [7]
You don't necessarily need VoIP to implement UC; you
can use the regular phone system. But VoIP does make
it easier: VoIP services already include mechanisms for
forwarding voice mail to e-mail, Find Me Follow Me
(FMFM) functionality, and other features used in a UC
system. In addition, you get more scalability and better
integration with VoIP than with UC-type products that
rely on traditional phone services.
Combining an asynchronous communication type such
as e-mail with a real-time communication type such as
telecommunications presents some challenges.
However, it gives users far more flexibility and allows
each of them to receive, process, and send messages in
the way that works best for that individual.
VoIP and Unified Messaging Challenges
Enterprises are moving towards Voice over Internet
Protocol (VoIP) and Unified Communications (UC) in
corporate networks because of its many benefits,
including
 Substantial cost savings by using the internet
to bypass long distance tolls
 Implementing advanced applications such as
unified messaging and presence.
 Improving employee collaboration and
productivity.
9. References
[1] http://en.wikipedia.org/wiki/Voice_over_IP.htm
[2]www.protocols.com/pbook/h323.htm
[3] www.protocols.com/pbook/terminal.htm
[4]http://en.wikipedia.org/wiki/Session_Initiation_Proto
col.htm
[5]http://www.cisco.com/en/US/products/sw/voicesw/a
dvantage.html
[6]http://en.wikipedia.org/wiki/Telepresence.htm
[7]McGowan_VoIP_June17th2010.pdf (page # 4-20)
[8]http://www.cisco.com/en/US/tech/tk652/tk701/techn
ologies_white_paper09186a00800d6b75.shtml

VoIP and Unified Communications Applications are

poised to become the dominant form of
communications within enterprises, replacing
traditional circuit switched telephony technology.
Enterprises must ensure there networks are secure,
scalable and reliable; dial tone reliability is paramount.
VoIP and Unified Communications applications
provide numerous challenges for enterprise networks.
All VoIP handsets require IP addresses to send and
receive data over the network; in most cases, deploying
VoIP will double IP allocation requirements overnight.
Traditional DHCP services offered on a switch or router
do not address the issues of failover, centralized
management, firmware management or management of
custom vendor options. Furthermore, services offered
on call managers and media servers are rudimentary at
best, lacking adequate service level guarantees.
Adoption of VoIP brings with it numerous potential
risks that must be addressed to maximize benefit and
minimize risk exposure. Today, compromising your
legacy phone system means someone has to physically
cut the line to your PBX or gain access to the physical
phone network. This is no longer the case with VoIP.
Modern messaging protocols such as SIP remove call
functions from the circuit switch (PBX) and place call
management (setup, teardown) at the VoIP endpoint;
greatly increasing an organization’s exposure to risk.
Like your data network, your phone system is now
vulnerable to Viruses, Trojans, Hijacking, Spoofing,
and Denial of Service Attacks. A defense-in-depth
strategy is required, yet this strategy cannot come at the
cost of serviceability and QoS. Legacy solutions are not
a survivable solution. [8]