Assignment 2
Computer Network
067/BEX/445
Tej Prasad Adhikari
Peer-to-Peer vs.
Client Server Networks
In computer
networking, the architecture or way the network entities are arranged, plays a
key role in determining the communication and privilege flow. Two main models
of computer networks, are client server and peer-to-peer.
Peer-to-Peer vs.
Client Server Networks Comparison
Peer-to-Peer
(P2P) Model
|
Client Server
Model
|
Decentralized
form of networking architecture
|
Centralized
form of networking architecture
|
The network
access, tasks and workload are divided and shared amongst the various
members. It is an "everyone pulls their own weight" sort of
relationship
|
Working is
based on a resource provider or storehouse (server) and the entities that
require the resources (clients). The clients make requests to the server to
access the resources. It is a "make a request and it will be
granted" sort of service
|
Supply and
consumption of resources is carried out by the peers, there is no higher body
or "boss" and no separate entity exists to dole out resources. All
peers in a network can request for resources as well as grant them
|
Two members
of such a system are servers and clients. Servers contain the resources in
the form of information or data. Certain resources like say printers, can be
connected to the server and the client has to request access to the server,
to use the printer
|
Members are
called peers, they have the same privileges and rights and enjoy the same
access to various data sources and devices. There is no difference between
them in any manner. Peers communicate with each other directly, no need for a
median in the middle
|
Clients are
the respective workstations or computers that do not share their resources
but work on their own and makes requests to the server for data or resources
or functions
|
The
Peer-to-Peer network paradigm is commonly used in P2P file sharing programs
like Napster and Bitorrent
|
Email,
banking services, even the HTTP protocol are all examples of client server
model
|
Computers A,
B, C and D are connected in a P2P network. Comp A wants a file from Comp C,
it sends a request to C. C decides to accept the request, finds the file and
sends it to A. B and D are ignorant to what is going on but function normally.
There is a network printer to which all computers are connected to. A sends a
request to print and B sends one too. A's request reached first, so it is
granted. Then the printer will grant B's request
|
Computer A is
the server. Computers B, C and D are the clients. B wants to print a page.
The printer is attached to Comp A. B will send a request to A, asking to
print a page. A will print the page and respond to B. C wants to access a
file, it will send a request to A, asking for the file. A will check C's
credentials, C is not authorized to access the data, A will reject the
request and respond to C by turning down its request
|
2 types of
peer-to-peer networks exist. Structured P2P arranges peers in a order or
manner based on certain rules and algorithms. There is no change in
privilege, just in the way the members communicate. Unstructured P2P have no
such order or manner and consists of 3 models - pure, hybrid and centralized
|
There is no
specific model or type of client-server networks. It's more like a mixed bag
of different styles. For example, having two servers, one just for data and
one for devices and clients have to make requests to 2 different entities for
accessing such resources
|
The physical
structure is independent from the underlying network structure of behavior.
Peers can be arranged in any network topology but in small networks, are
located near to each other physically. The computers are similar in software
content and protocols used for networking
|
Physical
structure is divided. Servers are powerful machines, designed for a dedicated
purpose and should be robust to handle multiple transactions. Their hardware
makeup is more powerful with more storage space or RAM and powerful
processors. The server machine is normally contained in a different room with
increased security and better environmental conditions. Clients are ordinary
workstations, accessed by different users. They have their own data
|
Used by small
businesses and home users
|
Big
corporations or organizations with high security data
|
Comparison
between OSI reference model and TCP/IP reference model
OSI
1. It has 7 layers
2. Transport layer guarantees delivery of
packets
3. Horizontal approach
4. Separate presentation layer
5. Separate session layer
6. Network layer provides both
connectionless and connection oriented services
7. It defines the services, interfaces and
protocols very clearly and makes a clear distinction between them
8. The protocol are better hidden and can
be easily replaced as the technology changes
9. OSI truly is a general model
10. It has a problem of protocol filtering
into a model
TCP/IP
1.
Has only 4 layers
2.
Transport layer does not guarantees delivery of packets
3.
Vertical approach
4.
No session layer, characteristics are provided by
transport layer
5.
No presentation layer, characteristics are provided by
application layer
6.
Network layer provides only connection less services
7.
It does not clearly distinguishes between service
interface and protocols
8.
It is not easy to replace the protocols
9.
TCP/IP cannot be used for any other application
10. The
model does not fit any protocol stack.
Short
Notes
X.25
X.25 is an ITU-T
standard protocol suite for packet switched wide area network (WAN)
communication. An X.25 WAN consists of packet-switching exchange (PSE) nodes as
the networking hardware, and leased lines, plain old telephone service
connections or ISDN connections as physical links. X.25 is a family of
protocols that was popular during the 1980s with telecommunications companies
and in financial transaction systems such as automated teller machines. X.25
was originally defined by the International Telegraph and Telephone
Consultative Committee (CCITT, now ITU-T) in a series of drafts and
finalized in a publication known as The Orange Book in 1976. The general
concept of X.25 was to create a universal and global packet-switched network.
Much of the X.25 system is a description of the
rigorous error correction needed to achieve this, as well as more efficient
sharing of capital-intensive physical resources. The X.25 specification defines
only the interface between a subscriber (DTE) and an X.25 network (DCE). X.75,
a very similar protocol to X.25, defines the interface between two X.25
networks to allow connections to traverse two or more networks. X.25 does not
specify how the network operates internally—many X.25 network implementations
used something very similar to X.25 or X.75 internally, but others used quite
different protocols internally. The ISO equivalent protocol to X.25, ISO 8208,
is compatible with X.25, but additionally includes provision for two X.25 DTEs
to be directly connected to each other with no network in between. By
separating the Packet-Layer Protocol, ISO 8208 permits operation over
additional networks such as ISO 8802 LLC2 (ISO LAN) and the OSI data link
layer.
Although X.25 predates the OSI Reference Model (OSIRM), the physical Layer
of the OSI model corresponds to the X.25 physical layer, the data link layer to
the X.25 data link layer, and the network layer to the X.25 packet layer. The
X.25 data link layer, LAPB, provides a reliable data path across a data link
(or multiple parallel data links, multilink) which may not be reliable itself.
The X.25 packet layer, provides the virtual call mechanisms, running over X.25
LAPB. The packet layer includes mechanisms to maintain virtual calls and to
signal data errors in the event that the data link layer cannot recover from
data transmission errors. All but the earliest versions of X.25 include
facilities which provide for OSI network layer
X.25 was developed in the era of computer terminals connecting to host
computers, although it also can be used for communications between computers.
Instead of dialing directly “into” the host computer – which would require the
host to have its own pool of modems and phone lines, and require non-local
callers to make long-distance calls – the host could have an X.25 connection to
a network service provider. Now dumb-terminal users could dial into the
network's local “PAD” (Packet Assembly/Disassembly facility), a gateway device
connecting modems and serial lines to the X.25 link as defined by the X.29 and
X.3 standards.
Frame Relay
Frame Relay is a standardized wide area network
technology that specifies the physical and logical link layers of digital
telecommunications channels using a packet switching methodology. Originally
designed for transport across Integrated Services Digital Network (ISDN)
infrastructure, it may be used today in the context of many other network
interfaces.
Network providers commonly implement Frame Relay for
voice (VoFR) and data as an encapsulation technique, used between local area
networks (LANs) over a wide area network (WAN). Each end-user gets a private
line (or leased line) to a Frame Relay node. The Frame Relay network handles
the transmission over a frequently-changing path transparent to all end-user
extensively-used WAN protocols. It is less expensive than leased lines and that
is one reason for its popularity. The extreme simplicity of configuring user
equipment in a Frame Relay network offers another reason for Frame Relay's
popularity.
With the advent of Ethernet over fiber optics, MPLS,
VPN and dedicated broadband services such as cable modem and DSL, the end may
loom for the Frame Relay protocol and encapsulation. However many rural areas
remain lacking DSL and cable modem services. In such cases the least expensive
type of non-dial-up connection remains a 64-kbit/s frame-relay line. Thus a
retail chain, for instance, may use Frame Relay for connecting rural stores
into their corporate WAN.
Voice over IP
Voice over IP (voice over Internet Protocol, VoIP)
is a methodology and group of technologies for the delivery of voice
communications and multimedia sessions over Internet Protocol (IP) networks,
such as the Internet.
Early providers of voice over IP services offered
business models and technical solutions that mirrored the architecture of the
legacy telephone network. Second generation providers, such as Skype, have
built closed networks for private user bases, offering the benefit of free
calls and convenience, while potentially charging for access to other
communication networks, such as the PSTN. This has limited the freedom of users
to mix-and-match third-party hardware and software. Third generation providers,
such as Google Talk have adopted the concept of federated VoIP – which is a
departure from the architecture of the legacy networks. These solutions
typically allow dynamic interconnection between users on any two domains on the
Internet when a user wishes to place a call.
VoIP systems employ session control and signaling
protocols to control the signaling, set-up, and tear-down of calls. They
transport audio streams over IP networks using special media delivery protocols
that encode voice, audio, video with audio codecs and video codecs as Digital
audio by streaming media. Various codecs exist that optimize the media stream
based on application requirements and network bandwidth; some implementations
rely on narrowband and compressed speech, while others support high fidelity
stereo codecs. Some popular codecs include μ-law and a-law versions of G.711,
G.722 which is a high-fidelity codec marketed as HD Voice by Polycom, a popular
open source voice codec known as iLBC, a codec that only uses 8 Kbit/s each way
called G.729, and many others.
VoIP is available on many smartphones, personal
computers, and on Internet access devices. Calls and SMS text messages may be
sent over 3G or Wi-Fi.
Next-generation network
The next-generation network (NGN) is body of key
architectural changes in telecommunication core and access networks. The
general idea behind the NGN is that one network transports all information and
services (voice, data, and all sorts of media such as video) by encapsulating
these into packets, similar to those used on the Internet. NGNs are commonly
built around the Internet Protocol, and therefore the term all IP is also
sometimes used to describe the transformation toward NGN.
According to ITU-T, the definition is:
A
next-generation network (NGN) is a packet-based network which can provide
services including Telecommunication Services and able to make use of multiple
broadband, quality of Service-enabled transport technologies and in which
service-related functions are independent from underlying transport-related
technologies. It offers unrestricted access by users to different service
providers. It supports generalized mobility which will allow consistent and
ubiquitous provision of services to users.
From a practical perspective, NGN involves three
main architectural changes that need to be looked at separately:
·
In the core network, NGN implies a
consolidation of several (dedicated or overlay) transport networks each
historically built for a different service into one core transport network (often
based on IP and Ethernet). It implies amongst others the migration of voice
from a circuit-switched architecture (PSTN) to VoIP, and also migration of
legacy services such as X.25, frame relay (either commercial migration of the
customer to a new service like IP VPN, or technical emigration by emulation of
the "legacy service" on the NGN).
·
In the wired access network, NGN implies
the migration from the dual system of legacy voice next to xDSL setup in local
exchanges to a converged setup in which the DSLAMs integrate voice ports or
VoIP, making it possible to remove the voice switching infrastructure from the
exchange.
·
In the cable access network, NGN
convergence implies migration of constant bit rate voice to Cable Labs Packet
Cable standards that provide VoIP and SIP services. Both services ride over
DOCSIS as the cable data layer standard.
In an NGN, there is a more defined separation
between the transport (connectivity) portion of the network and the services
that run on top of that transport. This means that whenever a provider wants to
enable a new service, they can do so by defining it directly at the service
layer without considering the transport layer – i.e. services are independent
of transport details. Increasingly applications, including voice, tend to be
independent of the access network (de-layering of network and applications) and
will reside more on end-user devices (phone, PC, set-top box).
Multiprotocol Label Switching
Multiprotocol Label Switching (MPLS) is a mechanism
in high-performance telecommunications networks that directs data from one
network node to the next based on short path labels rather than long network
addresses, avoiding complex lookups in a routing table. The labels identify
virtual links (paths) between distant nodes rather than endpoints. MPLS can
encapsulate packets of various network protocols. MPLS supports a range of
access technologies, including T1/E1, ATM, Frame Relay, and DSL.
MPLS is a highly scalable, protocol agnostic,
data-carrying mechanism. In an MPLS network, data packets are assigned labels.
Packet-forwarding decisions are made solely on the contents of this label,
without the need to examine the packet itself. This allows one to create
end-to-end circuits across any type of transport medium, using any protocol.
The primary benefit is to eliminate dependence on a particular OSI model data
link layer technology, such as Asynchronous Transfer Mode (ATM), Frame Relay,
Synchronous Optical Networking (SONET) or Ethernet, and eliminate the need for
multiple layer-2 networks to satisfy different types of traffic. MPLS belongs
to the family of packet-switched networks.
MPLS operates at a layer that is generally
considered to lie between traditional definitions of layer 2 (data link layer)
and layer 3 (network layer), and thus is often referred to as a "layer
2.5" protocol. It was designed to provide a unified data-carrying service
for both circuit-based clients and packet-switching clients which provide a
datagram service model. It can be used to carry many different kinds of
traffic, including IP packets, as well as native ATM, SONET, and Ethernet
frames.
A number of different technologies were previously
deployed with essentially identical goals, such as Frame Relay and ATM. MPLS
technologies have evolved with the strengths and weaknesses of ATM in mind.
Many network engineers agree that ATM should be replaced with a protocol that
requires less overhead, while providing connection-oriented services for
variable-length frames. MPLS is currently replacing some of these technologies
in the marketplace. It is highly possible that MPLS will completely replace
these technologies in the future, thus aligning these technologies with current
and future technology needs.
In particular, MPLS dispenses with the
cell-switching and signaling-protocol baggage of ATM. MPLS recognizes that
small ATM cells are not needed in the core of modern networks, since modern
optical networks (as of 2008) are so fast (at 40 Gbit/s and beyond) that even
full-length 1500 byte packets do not incur significant real-time queuing delays
(the need to reduce such delays — e.g., to support voice traffic — was the
motivation for the cell nature of ATM).
At the same time, MPLS attempts to preserve the
traffic engineering and out-of-band control that made Frame Relay and ATM
attractive for deploying large-scale networks.
While the traffic management benefits of migrating
to MPLS are quite valuable (better reliability, increased performance), there
is a significant loss of visibility and access into the MPLS cloud for IT
departments
Digital subscriber line (xDSL)
Digital subscriber line (DSL, originally digital
subscriber loop) is a family of technologies that provide Internet access by
transmitting digital data over the wires of a local telephone network. In
telecommunications marketing, the term DSL is widely understood to mean
asymmetric digital subscriber line (ADSL), the most commonly installed DSL
technology. DSL service is delivered simultaneously with wired telephone
service on the same telephone line. This is possible because DSL uses higher
frequency bands for data. On the customer premises, a DSL filter on each
non-DSL outlet blocks any high frequency interference, to enable simultaneous
use of the voice and DSL services.
The bit rate of consumer DSL services typically
ranges from 256 Kbit/s to 40 Mbit/s in the direction to the customer
(downstream), depending on DSL technology, line conditions, and service-level
implementation. In ADSL, the data throughput in the upstream direction, (the
direction to the service provider) is lower, hence the designation of
asymmetric service. In symmetric digital subscriber line (SDSL) services, the
downstream and upstream data rates are equal.
okay , this seems good
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