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Many
of the documents below require Adobe Acrobat, if
you would like the most recent version of the application.
Click the Acrobat logo and follow the download instructions. |
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WHITE
PAPERS
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What’s
SIP Got To Do With It?
Enterprises are
rapidly recognizing the value of world-wide IP communications
integrated with simple, secure, standards-based, applications-rich—IP
messaging, IP conferencing, IP contact centers, and IP mobility
solutions—implementations and services. This paper offers
five compelling reasons that these organizations are looking
to Session Initiation Protocol as the standard on which to
build their productivity enhancing and cost reducing converged
networks. |
Download
White Paper - (1.25 Mbytes)  |
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Taking
the Guesswork Out of Deploying IP Telephony
Enterprises are
rapidly recognizing the value of world-wide IP communications
integrated with simple, secure, standards-based, applications-rich—IP
messaging, IP conferencing, IP contact centers, and IP mobility
solutions—implementations and services. This paper offers
five compelling reasons that these organizations are looking
to Session Initiation Protocol as the standard on which to
build their productivity enhancing and cost reducing converged
networks. |
Download
White Paper - (124 Mbytes)  |
 |
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Architecture
for Convergence
This white paper first appeared in the "Transforming
Telephony" supplement to the Business Communications
Review, October 2004, and is offered here with permission.
Its author, Gary Audin, President of Delphi, Inc. consultancy
and an independent communications and security consultant
for 25 years, discusses three architectures for IP telephony
from the perspective of eight specific attributes: flexibility,
longevity, availability, disaster recovery, common services,
management, load balancing, and expansion. His analysis emphasizes
the need to implement an applications and user interface architecture
that enhances the business mission, requiring more than a
focus only on the network. The paper closes with these words:
"Choose the next architecture for convergence wisely.
You will have it for most of your career." |
Download
White Paper - (338 Kbytes)  |
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A
Roadmap for Convergence
This white paper first appeared in the "Transforming
Telephony" supplement to the Business Communications
Review, October 2004, and is offered here with permission.
Its author, Gary Audin, President of Delphi, Inc. consultancy
and an independent communications and security consultant
for 25 years, highlights the flexibility of convergence applications
architecture as he reviews related technical issues, standards,
and opportunities for interoperability. |
Download
White Paper - (332 Kbytes)  |
 |
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SIP
Market Overview (Data Connections)
Session Initiation Protocol (SIP)
is continuing to develop rapidly and it is difficult to keep
up with all of its innovations and uses. This white paper
is aimed at people who want to understand the concepts and
drivers behind SIP adoption, and how it is evolving to face
new challenges. This paper summarizes where SIP has come from,
how it works, and what makes it such a useful protocol. It
then describes how SIP is used in applications including telephony,
conferencing and messaging, and how it is being extended to
provide innovative services and accommodate the requirements
of real-world deployment, where NATs, service level agreements
and regulators exist. In covering this broad range of SIP-related
topics, it provides a summary of the state of this increasingly
important protocol. |
Download
White Paper - (464 Kbytes)  |
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DEFENITIONS |
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Network
Devices and Components Overview |
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Network
components and devices are the physical entities connected
to a network. There are many types of network devices and
increasing daily. The basic network devices are: Computers
either a PC or a Server, Hubs, Switches, Bridges, Routers,
Gateways, Network interface cards (NICs), Wireless access
points (WAPs), Printers and Modems. The following is a overview
of the main network components and devices:
Individual
Computers: The personal computer is typically a desktop
computer, a workstation or a notebook for individual users.
The individual computers are the most common type of microcomputer
and is found in the majority of organizations.
Server:
A computer on a network or other network device that stores
all necessary information and is dedicated to provide a particular
service. For example, a database server would store all data
and software related to a certain database and allows other
network devices to access and process database queries. A
file server is a computer and storage device dedicated to
storing files for any user on the network to store files on
the server. A print server is a device that manages one or
more printers, and a network server is a computer that manages
network traffic.
Network
Interface Card: Network Interface Cards (NIC) are
adaptors attached with a computer or other network device
to provide the connection between the computer with the network.
Each NIC is design for a specific type of network such as
Ethernet, Token Ring, FDDI or wireless LAN. The NIC operates
using the physical layer (layer 1) and data link layer (layer
2) specifications. NIC basically defines the physical connection
methods with the cable and the framing methods used to transmit
bit streams over the network. It also defines the control
signals that provide the timing of data transfers across network.
Hubs:
Hubs are the simplest network devices. Computers connect to
a hub via a length of twisted-pair cabling. On a hub, data
is forwarded to all ports, regardless of whether the data
is intended for the system connected to the port. In addition
to ports for connecting computers, even a very inexpensive
hub generally has a port designated as an uplink port that
enables the hub to be connected to another hub to create larger
networks.
Switches:
Switch is a layer 2 and multi-port device. Switch provides
similar functions as a hub or a bridge but has more advanced
features that can temporarily connect any two ports together.
It contains a switch matrix or switch fabric that can rapidly
connect and disconnect ports. Unlike Hub, a switch only forward
frame from one port to the other port where the destination
node is connected without broadcast to all other ports.
Routers:
Routers route data around the network from data senders to
receivers. A router is able to determine the destination address
for the data and determines the best way for the data to continue
its journey. Unlike bridges and switches, which use the hardware-configured
MAC address to determine the destination of the data, routers
use the logic network address such as IP address to make decisions.
Gateway:
The term gateway is applied to any device, system, or software
application that can perform the function of translating data
from one format to another. Gateway will not change the data
itself. For example, a router that can route data from an
IPX network to an IP network is, technically, a gateway. The
same can be said of a translational switch that converts from
an Ethernet network to a Token Ring network and back again.
Modems:
Modems are access devices that translate digital signals from
a computer into analog signals that can travel across conventional
phone lines. The modem modulates the signal at the sending
end and demodulates at the receiving end. Modems are required
for many access methods such as 56k data modern, ISDN, DSL
etc. They can be as internal devices that plug into expansion
slots in a system; external devices that plug into serial
or USB ports; PCMCIA cards designed for use in laptops; and
specialized devices designed for use in systems such as handheld
computers. In addition, many laptops now come with integrated
modems. For large-scale modem implementations, such as at
an ISP, rack-mounted modems are also available. |
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Proxy
Server |
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A
proxy server, also called proxy, is a computer network service
that allows clients to make indirect network connections to
other network services. A client connects to the proxy server,
then requests a connection, file, or other resource available
on a different server. The proxy provides the resource either
by connecting to the specified server or by serving it from
a cache. In some cases, the proxy may alter the client's request
or the server's response for various purposes. |
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Proxy
Gateway |
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Proxy
gateway is a system which passes on a request for a URL from
a World-Wide Web browser such as Mosaic to an outside server
and return the results. This provides clients that are sealed
off from the Internet a trusted agent that can access the
Internet on their behalf. Once the client is properly configured,
its user should not be aware of the proxy gateway. A proxy
gateway often runs on a firewall machine. Its main purpose
is to act as a barrier to the threat of crackers. It may also
be used to hide the IP addresses of the computers inside the
firewall from the Internet if they do not use official registered
network numbers. |
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Server |
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Server
is a computer or other network device that stores all necessary
information and is dedicated to provide a particular service.
For example, a database server would store all data and software
related to a certain database and allows other network devices
to access and process database queries. A file server is a
computer and storage device dedicated to storing files for
any user on the network to store files on the server. A print
server is a device that manages one or more printers, and
a networkserver is a computer that manages network traffic.
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Router |
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A
router is a device or a piece of software in a computer that
forwards and routes data packets along networks. A router
connects at least two networks, commonly two LANs or WANs
or a LAN and its ISP network. A router is often included as
part of a network switch. A router is located at any gateway
where one network meets another, including each point-of-presence
on the Internet. |
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Hub |
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The
Hub, also called repeater, is a device that accepts Ethernet
connections from network devices and cross connects them.
Data arriving via the receive pair of one connection is regenerated
and sent out on the transmit pair to all connected devices
except for the device who originated the transmission. |
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Switch |
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A
switch is a networking device that connects network segments.
Technically, network switches operate at layer two (Data Link
Layer) of the OSI model. They were
developed from the electronic hub, where the hub provided
a central nodal device for a star-configured network. In a
shared hub, all star network connections receive a broadcast
frame. A switch is similar to a hub in that it provides a
single broadcast domain, but differs in that each port on
a switch is its own collision domain. Generally, a switch
contains more "intelligence" than a hub. Network
switches are capable of inspecting the data packets as they
are received, determining the source and destination device
of that packet, and forwarding that packet appropriately.
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Switch
Types |
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Layer
2 Switch
Layer
2 switch is a local area network switch that forwards traffic
based on MAC layer (Ethernet or Token Ring) addresses.
Layer
3 Switch
Layer
3 switch is a network device that forwards traffic based on
layer 3 information at very high speeds. Layer 3 switch uses
the same routing algorithms as traditional routers do. However,
Layer 3 switch performs its operations using application specific
integrated circuit (ASIC) hardware, while a router does it
using software in a microprocessor. A Layer 3 switch goes
beyond the Layer 2 MAC addressing and routing. The Layer 3
switch looks at the incoming packets networking protocol.
Layer 3 switching is more effectively used to segment a LAN
than to provide a WAN connection. Traditionally, routers,
which inspect layer 3, were considerably slower than layer
2 switches.
Layer
4 Switch
Layer
4 switch, based on the OSI "transport" layer, allows
for policy-based switching such as limiting different types
of traffic on specific end-user switch ports, or for prioritizing
certain packet types, such as database or application server
traffic. Layer 4 switches also offer a powerful combination
of Network Address Translation (NAT) with higher-layer address
screening. Actually, layer 4 switch may make forwarding decisions
based upon information at any OSI layer from 4 through 7,
depending upon the particular product. In fact, some of the
so-called "Layer 4 Switches" even monitor the state
of individual sessions from beginning to end, just as firewalls
do, in which case they're referred to as "session switches."
Therefore, it is called Layer 4 - 7 switch.
Layer
7 Switch
A
Layer 7 Switch performs wire-speed processing of packet header
content, not only at Layer 2 or Layer 3, but also at the transport
layer (Layer 4) up through the application layer (Layer 7).
Layer 7 switch integrates routing and switching by forwarding
traffic at layer 2 speed using layer 7 information. For example,
an XML switch can analyze the XML tags at the application
level and make forwarding decisions. |
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Networking
Standards and Protocols |
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1000Base-CX
Gigabit over 150 ohm coaxial cable up to 200 meter for Ethernet
1000Base-LX
Gigabit over fiber with long wave laser up to 3 kilometers
for Ethernet
1000Base-SX
Gigabit over fiber with short wave laser up to 550 meters
for Ethernet
1000BaseT
Gigabit over twisted pair for Ethernet
1000BaseX
Gigabit over multiple media for Ethernet
100BaseT
100 Mbps over twisted pair for Ethernet
100BaseX
100 Mbps for Ethernet for multiple media: FX: Fiber
10Base2 Thin
10 Mbps over thin coaxial cable
10Base5 Thick
10 Mbps over 50 ohm thick coaxial cable for Ethernet
10BaseF
10 Mbps over Fiber for Ethernet
10BaseT
10 Mbps over twisted pair for Ethernet
10Broad36
10 Mbps over coaxial cable up to 3600 meters with Frequency
Division Multiplexing
1Base5
1 Mbps over unshielded twisted pair for Ethernet
802.1
IEEE protocols suite for internetworking of LAN, MAN and WAN;
LAN security, and management.
802.12
100 VG-Any LAN IEEE standard
802.1ad
This standard, also referred to as ?Q-in-Q? tag stacking,
builds on the IEEE?s 802.1Q (Virtual LANs) to enable stacked
VLANs IEEE
802.1D
Spanning Tree Protocol IEEE
802.1P
LAN Layer 2 traffic prioritization (QoS) specification IEEE
802.1Q
Virtual LAN (VLAN) Switching IEEE protocol
802.1s
Multiple Spanning Tree Protocol IEEE
802.1w
Rapid Spanning Tree Protocol, an evolution of the Spanning
Tree Protocol, provides for faster spanning tree convergence
after a topology change. IEEE
802.1X
LAN/WLAN Authentication and Key Management (EAPOL) IEEE
802.2
Logical Link Control IEEE protocol
802.3
Ethernet LAN IEEE protocol suite
802.5
IEEE Token-passing access on ring topology using unshielded
twisted pair
802.6
Metropolitan Area Network (MAN) layer 2 IEEE standard (DQDB)
802.3ab
Gigabit Ethernet over twisted pair (1000BaseT) IEEE
802.3ad
Ethernet link aggregation IEEE
802.3ae
10 Gigabit Ethernet IEEE standard
803.3ah
Ethernet OAM: link monitoring, fault signaling, and remote
loopback IEEE
802.3u
Fast Ethernet - 100 Mbps LAN IEEE
802.3z
Gigabit Ethernet over fiber IEEE standard (1000BaseX) |
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LAN
Architecture and Topologies: Bus, Star, Ring and Tree |
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The
components in a Local Area Network can be connected in a few
ways, which is call LAN topologies. There exit 4 basic LAN
topologies:
Star:
All stations are connected by cable (or wireless) to a central
point, such as hub or a switch. If the central node is operating
in a broadcast fashion such as a Hub, transmission of a frame
from one station to the node is retransmitted on all of the
outgoing links. In this case, although the arrangement is
physically a star, it is logically a bus. In the case of the
central node acting as switch, an incoming frame is processed
in the node and then retransmitted on an outgoing link to
the destination station. Ethernet protocols (IEEE 802.3) are
often used in the Star topology LAN.
Ring:
All nodes on the LAN are connected in a loop and their Network
Interface Cards (NIC) are working as repeaters. There is no
starting or ending point. Each node will repeat any signal
that is on the network regardless its destination. The destination
station recognizes its address and copies the frame into a
local buffer as it goes by. The frame continues to circulate
until it returns to the source station, where it is removed.
Token Ring (IEEE 802.5) is the most popular Ring topology
protocol. FDDI (IEEE 802.6) is another protocol used in the
Ring topology, which is based on the Token Ring.
Bus:
All nodes on the LAN are connected by one linear cable, which
is called the shared medium. Every node on this cable segment
sees transmissions from every other station on the same segment.
At each end of the bus is a terminator, which absorbs any
signal, removing it from the bus. This medium cable apparently
is the single point of failure. Ethernet (IEEE 802.3) is the
protocols used for this type of LAN.
Tree:
The tree topology is a logical extension of the bus topology.
The transmission medium is a branching cable with no closed
loops. The tree layout begins at a point called the head-end,
where one or more cables start, and each of these may have
branches. The branches in turn may have additional branches
to allow quite complex layouts. |
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Ethernet:
IEEE 802.3 Local Area Network (LAN) protocols
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Ethernet
protocols refer to the family of local-area network (LAN)
covered by the IEEE 802.3. In the Ethernet standard, there
are two modes of operation: half-duplex and full-duplex modes.
In the half duplex mode, data are transmitted using the popular
Carrier-Sense Multiple Access/Collision Detection (CSMA/CD)
protocol on a shared medium. The main disadvantages of the
half-duplex are the efficiency and distance limitation, in
which the link distance is limited by the minimum MAC frame
size. This restriction reduces the efficiency drastically
for high-rate transmission. Therefore, the carrier extension
technique is used to ensure the minimum frame size of 512
bytes in Gigabit Ethernet to achieve a reasonable link distance.
Four
data rates are currently defined for operation over optical
fiber and twisted-pair cables:
- 10
Mbps - 10Base-T Ethernet (IEEE 802.3)
- 100
Mbps - Fast Ethernet (IEEE 802.3u)
- 1000
Mbps - Gigabit Ethernet (IEEE 802.3z)
- 10-Gigabit
- 10 Gbps Ethernet (IEEE 802.3ae)
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Fast
Ethernet: 100Mbps Ethernet (IEEE 802.3u)
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Fast
Ethernet (100BASE-T) offers a speed increase ten times that
of the 10BaseT Ethernet specification, while preserving such
qualities as frame format, MAC mechanisms, and MTU. Such similarities
allow the use of existing 10BaseT applications and network
management tools on Fast Ethernet networks. Officially, the
100BASE-T standard is IEEE 802.3u. Like
Ethernet, 100BASE-T is based on the CSMA/CD LAN access method.
There are several different cabling schemes that can be used
with 100BASE-T, including:
- 100BASE-TX:
two pairs of high-quality twisted-pair wires
- 100BASE-T4:
four pairs of normal-quality twisted-pair wires
- 100BASE-FX:
fiber optic cables
The
Fast Ethernet specifications include mechanisms for Auto-Negotiation
of the media speed. This makes it possible for vendors to
provide dual-speed Ethernet interfaces that can be installed
and run at either 10-Mbps or 100-Mbps automatically.
Fast
Ethernet standard is defined by IEEE (http://www.ieee.org
) in 802.3 & 802.3u. |
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Gigabit
(1000 Mbps) Ethernet: IEEE 802.3z (1000Base-X), 802.3ab (1000Base-T)
and GBIC |
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Ethernet
protocols refer to the family of local-area network (LAN)
covered by the IEEE 802.3 standard. The Gigabit Ethernet is
based on the Ethernet protocol, but increased speed tenfold
over the fast Ethernet, using shorter frames with carrier
Extension. It is published as the IEEE 802.3z and 802.3ab,
supplement to the IEEE 802.3 base standards.
Carrier
Extension is a simple solution, but it wastes bandwidth. Packet
Bursting is "Carrier Extension plus a burst of packets".
Burst mode is a feature that allows a MAC to send a short
sequence (a burst) of frames equal to approximately 5.4 maximum-length
frames without having to relinquish control of the medium.
The
Gigabit Ethernet standards are fully compatible with Ethernet
and Fast Ethernet installations. It retains Carrier Sense
Multiple Access/ Collision Detection (CSMA/CD) as the access
method. It supports full-duplex as well as half duplex modes
of operation. Single-mode and multi mode fiber and short-haul
coaxial cable, and twisted pair cables are supported.
- The
IEEE 802.3z defines the Gigabit Ethernet over fiber and
cable, which has a physical media standard 1000Base-X (1000BaseSX
- short wave covers up to 500m, and 1000BaseLX - long wave
covers up to 5km). The IEEE 802.3ab defines the Gigabit
Ethernet over the unshielded twisted pair wire (1000Base-T
covers up to 75m).
- The
Gigabit interface converter (GBIC) allows network managers
to configure each gigabit port on a port-by-port basis for
short-wave (SX), long-wave (LX), long-haul (LH), and copper
physical interfaces (CX). LH GBICs extended the single-mode
fiber distance from the standard 5 km to 10 km.
Gigabit
Ethernet is defined by IEEE (http://www.ieee.org)
802.3z and 802.3ab. |
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10
Gigabit Ethernet Protocol IEEE 802.3ae for LAN, WAN and MAN |
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10-Gigabit
Ethernet, being standardized in IEEE 802.3ae, offers data
speeds up to 10 billion bits per second.Built on the Ethernet
technology used in most of today's local area networks (LANs),
it offers similar benefits to those of the preceding Ethernet
standard. 10-Gigabit Ethernet is used to interconnect local
area networks (LANs), wide area networks (WANs), and metropolitan
area networks (MANs). 10-Gigabit Ethernet uses the familiar
IEEE 802.3 Ethernet media access control (MAC) protocol and
its frame format and size. However, it supports full duplex
mode but not the half-duplex operation mode and it only functions
over optical fiber. So it does not need the carrier-sensing
multiple-access with Collision Detection (CSMA/CD) protocol
as it is used in other Ethernet standards.
The
10 Gigabit specifications, contained in the IEEE 802.3ae supplement
to the 802.3 standard, provides support to extend the 802.3
protocol and MAC specification to an operating speed of 10
Gb/s. In addition to the data rate of 10 Gb/s, 10-Gigabit
Ethernet is able to accommodate slower date rates such as
9.584640 Gb/s (OC-192), through its "WAN interface sublayer"
(WIS) which allows 10 Gigabit Ethernet equipment to be compatible
with the Synchronous Optical Network (SONET) STS-192c transmission
format.
The
10GBASE-SRand 10GBASE-SWmedia types are for use over short
wavelength (850 nm) multimode fiber (MMF), which covers a
fiber distance from 2 meters to 300 meters.. The 10GBASE-SR
media type is designed for use over dark fiber, meaning a
fiber optic cable that is not in use and that is not connected
to any other equipment. The 10GBASE-SW media type is designed
to connect to SONET equipment, which is typically used to
provide long distance data communications.
The
10GBASE-LRand 10GBASE-LWmedia types are for use over long
wavelength (1310 nm) single-mode fiber (SMF), which covers
a fiber distance from 2 meters to 10 kilometers (32,808 feet).
The 10GBASE-LR media type is designed for use over dark fiber,
while the 10GBASE-LW media type is designed to connect to
SONET equipment.
The
10GBASE-ERand 10GBASE-EWmedia types are for use over extra
long wavelength (1550 nm) single-mode fiber (SMF), which covers
a fiber distance from 2 meters up to 40 kilometers (131,233
feet). The 10GBASE-ER media types is designed for use over
dark fiber, while the 10GBASE-EW media type is designed to
connect to SONET equipment.
Finally,
there is a 10GBASE-LX4media type, which uses wave division
multiplexing technology to send signals over four wavelengths
of light carried over a single pair of fiber optic cables.
The 10GBASE-LX4 system is designed to operate at 1310 nm over
multi-mode or single-mode dark fiber. The design goal for
this media system is from 2 meters up to 300 meters over multimode
fiber or from 2 meters up to 10 kilometers over single-mode
fiber.
10
Gigabit Ethernet is defined by IEEE (http://www.ieee.org). |
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OSI
7 Layers Reference Model For Network Communication
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Open
Systems Interconnection (OSI) model is a reference model developed
by ISO (International Organization for Standardization) in
1984, as a conceptual framework of standards for communication
in the network across different equipment and applications
by different vendors. It is now considered the primary architectural
model for inter-computing and internetworking communications.
Most of the network communication protocols used today have
a structure based on the OSI model. The OSI model defines
the communications process into 7 layers, which divides the
tasks involved with moving information between networked computers
into seven smaller, more manageable task groups. A task or
group of tasks is then assigned to each of the seven OSI layers.
Each layer is reasonably self-contained so that the tasks
assigned to each layer can be implemented independently. This
enables the solutions offered by one layer to be updated without
adversely affecting the other layers.
The
OSI 7 layers model has clear characteristics. Layers 7 through
4 deal with end to end communications between data source
and destinations. Layers 3 to 1 deal with communications between
network devices.
On
the other hand, the seven layers of the OSI model can be divided
into two groups: upper layers (layers 7, 6 & 5) and lower
layers (layers 4, 3, 2, 1). The upper layers of the OSI model
deal with application issues and generally are implemented
only in software. The highest layer, the application layer,
is closest to the end user. The lower layers of the OSI model
handle data transport issues. The physical layer and the data
link layer are implemented in hardware and software. The lowest
layer, the physical layer, is closest to the physical network
medium (the wires, for example) and is responsible for placing
data on the medium.
The
specific description for each layer is as follows:
Layer
7: Application Layer
Defines
interface to user processes for communication and data transfer
in network
Provides
standardized services such as virtual terminal, file and job
transfer and operations
Layer
6: Presentation Layer
Masks
the differences of data formats between dissimilar systems
Specifies
architecture-independent data transfer format
Encodes
and decodes data; Encrypts and decrypts data; Compresses and
decompresses data
Layer
5: Session Layer
Manages
user sessions and dialogues
Controls
establishment and termination of logic links between users
Reports
upper layer errors
Layer
4: Transport Layer
Manages
end-to-end message delivery in network
Provides
reliable and sequential packet delivery through error recovery
and flow control mechanisms
Provides
connectionless oriented packet delivery
Layer
3: Network Layer
Determines
how data are transferred between network devices
Routes
packets according to unique network device addresses
Provides
flow and congestion control to prevent network resource depletion
Layer
2: Data Link Layer
Defines
procedures for operating the communication links
Frames
packets
Detects
and corrects packets transmit errors
Layer
1: Physical Layer
Defines
physical means of sending data over network devices
Interfaces
between network medium and devices
Defines
optical, electrical and mechanical characteristics
There
are other network architecture models, such as IBM
SNA (Systems Network Architecture) model . Those models
will be discussed in separate documents.
The
OSI 7 layer model is defined by
ISO in document 7498 and ITU X.200, X.207, X.210, X.211, X.212,
X.213, X.214, X.215, X.217 and X.800. The protocols defined
by ISO based on the OSI 7 layer mode. |
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IBM
SNA - Systems Network Architecture and Protocols |
 |
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SNA
(Systems Network Architecture) is one of the most popular
network architecture models, in addition to the OSI
Model, proposed by IBM. Although SNA model is now considered
a legacy networking model, SNA is still widely deployed. SNA
was designed around the host-to-terminal communication model
that IBM's mainframes use. IBM expanded the SNA protocol to
support peer-to-peer networking. This expansion was deemed
Advanced Peer-to-Peer Networking (APPN)
and Advanced Program-to-Program Communication (APPC). Advanced
Peer-to-Peer Networking (APPN) represents IBM's second-generation
SNA. In creating APPN, IBM moved SNA from a hierarchical,
mainframe-centric environment to a peer-to-peer (P2P) networking
environment. At the heart of APPN is an IBM architecture that
supports peer-based communications, directory services, and
routing between two or more APPC systems that are not directly
attached.
IBM
SNA model has many similarities with the OSI
7 layers model. However, SNA model has only 6 layers and
does not define specific protocols for its physical control
layer. The physical control layer is assumed to be implemented
via other standards. The functions of each SNA layer are described
as follows:
Data
link control (DLC)- Defines several protocols, including the
Synchronous Data Link Control (SDLC) protocol for hierarchical
communication, and the Token Ring Network communication protocol
for LAN communication between peers. SDLC provided a foundation
for ISO HDSL and IEEE 802.2.
- Path
control- Performs many OSI network layer functions, including
routing and datagram segmentation and reassembly (SAR)
- Transmission
control- Provides a reliable end-to-end connection service
(similar to TCP), as well as encrypting and decrypting services
- Data
flow control- Manages request and response processing, determines
whose turn it is to communicate, groups messages, and interrupts
data flow on request
- Presentation
services- Specifies data-transformation algorithms that
translate data from one format to another, coordinate resource
sharing, and synchronize transaction operations
-
Transaction services- Provides application services in the
form of programs that implement distributed processing or
management services
|
--- |
 |
|
What
is Voice Over Internet Protocol (VoIP)? |
|
-- |
VoIP
is a packet technology allowing the analog waves of our spoken
words to be converted to digital signals and then packetized.
Packets are sent over the IP network to the end point for
reassembly and conversion to sound.
Using
VOIP protocols, voice communications can be achieved on any
IP network regardless it is Internet, Intranets or Local Area
Networks (LAN). In a VOIP enabled network, the voice signal
is digitized, compressed and converted to IP packets and then
transmitted over the IP network. VOIP signaling protocols
are used to set up and tear down calls, carry information
required to locate users and negotiate capabilities. The key
benefits of Internet telephony (voice over IP) are the very
low cost, the integration of data, voice and video on one
network, the new services created on the converged network
and simplified management of end user and terminals.
here
are a few VOIP protocol stacks which are derived from various
standard bodies and vendors, namely H.323, SIP,
MEGACO and MGCP. |
--- |
 |
|
H.323
VOIP Protocol
|
 |
-- |
H.323
is the ITU-T's standard, which was originally developed for
multimedia conferencing on LANs, but was later extended to
cover Voice over IP. The standard encompasses both point to
point communications and multipoint conferences. H.323 defines
four logical components: Terminals, Gateways, Gatekeepers
and Multipoint Control Units (MCUs). Terminals,
gateways and MCUs are known as endpoints.
There
are five types of information exchange enabled in the H.323
architecture:
- Audio
(digitized) voice
- Video
(digitized)
- Data
(files or image)
- Communication
control (exchange of supported functions, controlling logic
channels, etc.)
- Controlling
connections and sessions (setup and tear down)
The
H.323 was first published in 1996 and the latest version (v5)
was completed in 2003.
H.323
is an ITU-T (http://www.itu.int/ITU-T/
) standard. |
--- |
 |
|
Session
Initiation Protocol (SIP) |
|
-- |
SIP
is an Internet Engineering Task Force (IETF) standard for
managing the handshake procedures for beginning and ending
real-time communications between IP network end points.
SIP
is a text-based protocol, similar to HTTP and SMTP, for initiating
interactive communication sessions between users. This makes
SIP easy to troubleshoot, enables fast application development,
and presents a stable framework for establishing interoperability
between devices, applications, call controllers, and gateways.
SIP is used to enable human-to-human communications that might
include voice, video, chat, interactive games, and virtual
reality.
SIP
is a component that can be used with other IETF protocols
to build a complete multimedia architecture, such as the Real-time
Transport Protocol (RTP) for transporting real-time data and
providing QoS feedback, the Real-Time streaming protocol (RTSP)
for controlling delivery of streaming media, the Media Gateway
Control Protocol (MEGACO) for controlling gateways to the
Public Switched Telephone Network (PSTN), and the Session
Description Protocol (SDP ) for describing multimedia sessions.
Therefore, SIP should be used in conjunction with other protocols
in order to provide complete services to the users. However,
the basic functionality and operation of SIP does not depend
on any of these protocols.
SIP
provides a suite of security services, which include denial-of-service
prevention, authentication (both user to user and proxy to
user), integrity protection, and encryption and privacy services.
SIP
is defined by IETF (www.ietf.org
) in RFC 3261, 3262, 3263, 3264, and 3265. |
--- |
 |
|
Megaco/H.248:
Media Gateway Control Protocol |
|
-- |
The Media Gateway Control Protocol (Megaco)
is a result of joint efforts of the IETF and the ITU-T (ITU-T
Recommendation H.248). Megaco/H.248 is for control of elements
in a physically decomposed multimedia gateway, which enables
separation of call control from media conversion. Megaco/H.248
addresses the relationship between the Media Gateway (MG),
which converts circuit-switched voice to packet-based traffic,
and the Media Gateway Controller, which dictates the service
logic of that traffic). Megaco/H.248 instructs an MG to connect
streams coming from outside a packet or cell data network
onto a packet or cell stream such as the Real-Time Transport
Protocol (RTP). Megaco/H.248 is essentially quite similar
to MGCP from an architectural standpoint and the controller-to-gateway
relationship, but Megaco/H.248 supports a broader range of
networks, such as ATM.
Megaco/H.248
is defined by IETF (www.ietf.org
) and ITU-T. |
--- |
 |
|
MGCP/Media
Gateway Control Protocol |
 |
-- |
Media
Gateway Control Protocol (MGCP) is used for controlling telephony
gateways from external call control elements called media
gateway controllers or call agents. A telephony gateway is
a network element that provides conversion between the audio
signals carried on telephone circuits and data packets carried
over the Internet or over other packet networks.
MGCP
assumes a call control architecture where the call control
intelligence is outside the gateways and handled by external
call control elements. The MGCP assumes that these call control
elements, or Call Agents, will synchronize with each other
to send coherent commands to the gateways under their control.
MGCP is, in essence, a master/slave protocol, where the gateways
are expected to execute commands sent by the Call Agents.
MGCP
is defined in RFC:
2705 by IETF (www.ietf.org
) and ITU-T. |
--- |
 |
|
VoIP
for the Telecommuter |
|
-- |
Telecommuters
are a tough group to support. They need data lines and separate
voice circuits both of which rack up huge costs. And there
is little hope for seamlessly integrating them into a corporatewide
call-management system because the most-advanced options usually
are telco-provided Centrex services. But VoIP holds tremendous
promise for telecommuters. By providing a single data circuit
and H.323-compliant equipment, you can integrate them into
your network easily, with access to the company operator (just
by dialing "0"), the voicemail system and other
telephony resources. You can even reduce toll charges by letting
the telecommuter place H.323 calls to remote offices. Telecommuters
can also forward calls to a regional office without sacrificing
any features. |
--- |
 |
|
PPPoE:
Power Over Ethernet Overview |
 |
-- |
PPP
over Ethernet (PPPoE) provides the ability to connect a network
of hosts over a simple bridging access device to a remote
Access Concentrator. With this model, each host utilizes it's
own PPP stack and the user is presented with a familiar user
interface. Access control, billing and type of service can
be done on a per-user, rather than a per-site, basis.
To
provide a point-to-point connection over Ethernet, each PPP
session must learn the Ethernet address of the remote peer,
as well as establish a unique session identifier. PPPoE includes
a discovery protocol that provides this.
PPPoE
has two distinct stages. There is a Discovery stage and a
PPP Session stage. When a Host wishes too initiate a
PPPoE session, it must first perform Discovery to identify
the Ethernet MAC address of the peer and establish a PPPoE
SESSION_ID. While PPP defines a peer-to-peer relationship,
Discovery is inherently a client-server relationship. In the
Discovery process, a Host (the client) discovers an Access
Concentrator (the server). Based on the network topology,
there may be more than one Access Concentrator that the Host
can communicate with. The Discovery stage allows the Host
to discover all Access Concentrators and then select one.
When Discovery completes successfully, both the Host and the
selected Access Concentrator have the information they will
use to build their point-to-point connection over Ethernet.
The Discovery stage remains stateless until a PPP session
is established. Once a PPP session is established, both the
Host and the Access Concentrator MUST allocate the resources
for a PPP virtual interface.
PPPoE
is defined by IETF (http://www.ietf.org
) RFC 2516. |
--- |
 |
|
NAT: IP
Network Address Translator (Network Address Translation) |
 |
-- |
Basic
Network Address Translation (Basic NAT) is a method by which
IP addresses are mapped from one group to another, transparent
to end users. Network Address Port Translation, or NAPT is
a method by which many network addresses and their TCP/UDP
ports are translated into a single network address and its
TCP/UDP ports. Together, these two operations, referred to
as traditional NAT, provide a mechanism to connect a realm
with private addresses to an external realm with globally
unique registered addresses.
The
need for IP Address translation arises when a network's internal
IP addresses cannot be used outside the network either for
privacy reasons or because they are invalid for use outside
the network. Network topology outside a local domain can change
in many ways. Customers may change providers, company backbones
may be reorganized, or providers may merge or split. Whenever
external topology changes with time, address assignment for
nodes within the local domain must also change to reflect
the external changes. Changes of this type can be hidden from
users within the domain by centralizing changes to a single
address translation router. Basic Address translation would
allow hosts in a private network to transparently access the
external network and enable access to selective local hosts
from the outside. Organizations with a network setup
predominantly for internal use, with a need for occasional
external access are good candidates for this scheme.
There
are limitations to using the translation method. It is mandatory
that all requests and responses pertaining to a session be
routed via the same NAT router. One way to ascertain
this would be to have NAT based on a border router that is
unique to a stub domain, where all IP packets are either originated
from the domain or destined to the domain. There are other
ways to ensure this with multiple NAT devices.
This
solution has the disadvantage of taking away the end-to-end
significance of an IP address, and making up for it with increased
state in the network. As a result, end-to-end IP network
level security assured by IPSec cannot be assumed to end hosts,
with a NAT device enroute. The advantage of this approach
however is that it can be installed without changes to hosts
or routers.
NAT
is defined by IETF (http://www.ietf.org
) RFC3022. |
--- |
 |
|
Layer
3 IP VPN: Internet Protocol Virtual Private Network
|
 |
-- |
Internet
Protocol Virtual Private Network (IP VPN) is a group of technologies
that are widely used by corporations and service providers
to provide secured, private and scalable communications with
proper QoS, over a public IP based infrastructure such as
the Internet and Service Provider shared IP networks. IP VPN
is replacing the traditional VPN technologies such as ATM
VPN, Frame Relay VPN and TDM based VPN to become the main
stream of the VPN services, though interfaces to the existing
technologies exist in some cases.
The
core technology of VPN is the encapsulation or tunneling algorithms.
Primarily, there are three types of IP VPN technologies:
IPsec based IP VPN, MPLS based IP VPN and SSL base IP VPN.
Different technologies may have different focus of benefits
and serve different business purposes. The following are summaries
of the three types of technologies, their main applications
and limitations:
MPLS
based IP VPN:
MPLS-based
Layer 3 VPNs uses MPLS labeling algorithms and signaling protocols
to encapsulate IP packets and distribute VPN-related information.
MPLS based IP VPN can seamlessly interface with traditional
VPN technologies such as ATM, Frame Relay and TDM etc. It
can be an alternative or a complementary VPN solution to the
legacy deployment. A primary advantage of MPLS is that it
provides the scalability to support both small and very large-scale
VPN deployments. It can support end-to-end QoS, rapid faultcorrection
of link and node failure, bandwidth protection, and a foundation
for deploying additional value-added services. MPLS technology
also simplifies configuration, management, and provisioning,
helping service providers to deliver highly scalable, differentiated,
end-to-end IP based services. The service provider can offer
SLAs by enabling MPLS traffic engineering and fast reroute
capabilities in the core network. MPLS based IP VNP is a network
based VPN technology for site-to-site VPN communications only.
IPsec
Based IP VPN:
IPSec
protocol provides the framework for CPE-based Layer 3 VPNs.
IPSec supports 1)Data confidentiality by encrypting packets
before transmission; 2)Data integrity through authenticating
packets 3)Data origin authentication; 4) Anti-replay; 5) Encapsulating
Security Payload (ESP), for confidentiality. IPSec parameters
are communicated and negotiated between network devices in
accordancewith the Internet Key Exchange (IKE) protocol.The
IPSec protocol provides protection for IP packets by allowing
network designers to specify the traffic that needs protection,
define how thattraffic is to be protected, and control who
can receive the traffic. IPSec VPNs is a replacement technology
to the traditional VPNs such as leased-line, Frame Relay,
or ATM. The advantage of IPSec is that it meets network requirements
more cost effectively and with greater flexibility byusing
the public IP network such as the Internet and service providers¡¯
IP-based networks.IPSec is suitable for both site-to-site
and remote-access VPNs.
SSL
based IP VPN:
Secure
Sockets Layer (SSL) is for remote-access VPNs, instead of
site-to-site VPNs. In the SSL based VPN, the Secure Sockets
Layer protocol is used for packet encapsulation and user authentication.
SSL provides access to Web-based applications from any location
with a Web browser, an Internet connection, and without special
clientsoftware. It provides secure connectivity by authenticating
the communicating parties and encrypting the traffic that
flows between them.SSL-based VPNs only support applications
coded for SSL, including standard e-mail clients, Telnet,
FTP, IP telephony, multicastapplications, and applications
requiring QoS. |
--- |
 |
|
MPLS:
Multiprotocol Label Switching |
 |
-- |
Multiprotocol
Label Switching (MPLS), an architecture for fast packet switching
and routing, provides the designation, routing, forwarding
and switching of traffic flows through the network. More specifically,
MPLS has mechanisms to manage traffic flows of various granularities.
MPLS is independent of the layer-2 and layer-3 protocols such
as ATM and IP. MPLS provides a means to map IP addresses to
simple, fixed-length labels used by different packet-forwarding
and packet-switching technologies. MPLS interfaces to existing
routing and switching protocols, such as IP, ATM, Frame Relay,
Resource ReSerVation Protocol (RSVP) and Open Shortest Path
First (OSPF), etc.
In
MPLS, data transmission occurs on Label-Switched Paths (LSPs).
LSPs are a sequence of labels at each and every node along
the path from the source to the destination. There are several
label distribution protocols used today, such as Label Distribution
Protocol (LDP) or RSVP or piggybacked on routing protocols
like border gateway protocol (BGP) and OSPF. High-speed switching
of data is possible because the fixed-length labels are inserted
at the very beginning of the packet or cell and can be used
by hardware to switch packets quickly between links.
MPLS
is designed to address the network problems such as networks-speed,
scalability, quality-of-service (QoS), and traffic engineering.
MPLS has also become a solution to meet the bandwidth-management
and service requirements for next-generation IP-based backbone
networks. |
--- |
 |
|
IPsec:
Security Architecture for IP Network
|
 |
-- |
IPsec
provides security services at the IP layer by enabling a system
to select required security protocols, determine the algorithm(s)
to use for the service(s), and put in place any cryptographic
keys required to provide the requested services. IPsec can
be used to protect one or more "paths" between a
pair of hosts, between a pair of security gateways, or between
a security gateway and a host.
The
set of security services that IPsec can provide includes access
control, connectionless integrity, data origin authentication,
rejection of replayed packets (a form of partial sequence
integrity), confidentiality (encryption), and limited traffic
flow confidentiality. Because these services are provided
at the IP layer, they can be used by any higher layer protocol,
e.g., TCP, UDP,
ICMP, BGP, etc.
These
objectives are met through the use of two traffic security
protocols, the Authentication Header (AH) and the Encapsulating
Security Payload (ESP), and through the use of cryptographic
key management procedures and protocols. The set of IPsec
protocols employed in any context, and the ways in which they
are employed, will be determined by the security and system
requirements of users, applications, and/or sites/organizations.
When
these mechanisms are correctly implemented and deployed, they
ought not to adversely affect users, hosts, and other Internet
components that do not employ these security mechanisms for
protection of their traffic. These mechanisms also are designed
to be algorithm-independent. This modularity permits selection
of different sets of algorithms without affecting the other
parts of the implementation. For example, different user communities
may select different sets of algorithms (creating cliques)
if required.
A
standard set of default algorithms is specified to facilitate
interoperability in the global Internet. The use of these
algorithms, in conjunction with IPsec traffic protection and
key management protocols, is intended to permit system and
application developers to deploy high quality, Internet layer,
cryptographic security technology. |
--- |
 |
|
Layer
2 Ethernet VPN and Virtual Private LAN Services (VPLS)
|
 |
-- |
Virtual
Private LAN Services (VPLS) is a solution that can provide
layer 2 Virtual Private Network (VPN) services over Ethernet
networks. It uses a combination of Ethernet
and MPLS to meet the needs of carriers
and customers alike. VPLS allows customer networks at geographically
diverse locations to communicate with each other as if they
were in the same LAN. The WAN and MAN becomes transparent
to all customer locations. Ethernet VPN based on VLPS and
MPLS provides more benefits than other alternative layer 2
or 3 VPN technologies:
- Lower
capital expenditure required for deploying Ethernet infrastructure
by the Service Providers and customers
- Better
scalability due to unlimited scalability of MPLS
- Better
reliability because MPLS provides many advantageous reliability
features
- Better
QoS management because the traffic engineering capabilities
in MPLS allow providers to support service level guarantees
across the entire network.
- Improved
OAM: MPLS' dynamic signaling is instrumental in providing
quicker changes and reconfigurations of service.
- Protection
of investment on existing technologies because VPLS can
be used to offer not only Metro Ethernet services but can
also interconnect with existing ATM and Frame Relay access
networks and IP-VPN core networks running over various core
technologies such as Next Generation SONET/SDH and Dense
Wave Division Multiplexing (DWDM).
VPLS
Standards
VPLS
is currently being defined in the Internet Engineering Task
Force (IETF) with the broad support of carriers and vendors.
Most of VPLS related standards are still in the drafting stage
by the l2vpn and pwe3 working groups of IETF. There are primaried
two groups of technologies to address point-to-point communication
and point-to-multi-point commnucation.
Pseudowires
are point-to-point connections setup between pairs of Provider
Edge routers. Their primary function is to emulate services
like ATM, Frame Relay, Ethernet and TDM over an underlying
common MPLS network. To achieve this, each of these technologies
is encapsulated into a common MPLS format. These encapsulation
standards were previously known as the martini drafts. By
encapsulating services into a common MPLS format, pseudowires
allow carriers to converge their services to an MPLS network.
Ethernet encapsulating pseudowires are the building blocks
of VPLS. The pseudowire encapsulation standards are being
defined in the IETF's pwe3 working group.
For
the point-to-multipoint communication network, the customer
sites are connected through a service provider network, which
appears as a Layer 2 switch capable of learning and aging.
Customer sites are connected to the service provider network
at the Provider Edge (PE). All PEs in the network are connected
together in a full mesh of tunnels with each tunnel carrying
multiple pseudowires. Depending on the location and the number
of customer sites, the number of pseduowires setup for a customer/service
may range from one (for a customer with only two locations)
to a full mesh (for a customer who has locations connected
to every PE). All unknown unicast, multicast and broadcast
packets are flooded to all the PEs participating in a customer
VPN. This network model assumes that all PEs in a service
(or VPLS instance) are connected in a full mesh of pseudowires
which obviates the need to keep the network loop free. The
VPLS network model is being standardized as part of the VPLS
drafts in the IETF's l2vpn working group.
To
improve its scalability, Hierarchical VPLS (HVPLS) is introduced.
The HVPLS standards allow the creation of hierarchies with
a hub-and-spoke arrangement. The full mesh of tunnels is maintained
between the hub sites (designated as PEs). The CE equipment
is connected to an MTU-s router, which is connected to a PE
router, thus providing the hierarchy. |
--- |
 |
|
VPN:
Virtual Private Network
|
 |
-- |
Virtual
Private Network (VPN) refers to simulating a private network
over the public Internet by encrypting communications between
the two private end-points. This provides the same connectivity
, QOS and privacy you would find on a typical private network.
Typically, VPNs cab be categorized as follows:
Traditional
VPNs
Frame
Relay VPN (Layer 2)
ATM VPN (Layer 2)
CPE-based VPNs
L2TP
and PPTP VPN (Layer 2)
IPsec VPN (Layer 3)
Provider Provisioned VPNs (PP-VPNs)
BGP/MPLS
VPNs (Layer 2 and 3, RFC 2547bis)
Session based VPN
SSL
VPN (Layer 4 +)
SOCKS VPN (Layer 4 +)
The
traditional VPN technologies have been widely deployed in
the field by Service Providers and Enterprises. However, due
to their high cost and less features, new VNP technologies
such as IPsec VPN, SSL VPN and MPLS VPN are becoming more
and more popular. These new VPN technologies are fully compatible
with TCP/IP, the choice of technology for data routing and
transportation of the world.
The
key technology for VPN is the security of data over a public
network. The three types of security: authentication, encryption
and encapsulation, forms the foundation of virtual private
networking. However, authentication, encryption and encapsulation
can be performed by many different technologies. In addition,
these three sets of technologies can be combined in different
ways.
For
data encapsulation in VPN, many tunneling technologies are
developed, such as Layer 2 Tunneling Protocol (L2TP), Layer
2 Forward protocol (L 2F ) and Point to Point Tunneling Protocol
(PPTP). PPTP provides remote users encrypted, multi-protocol
access to a corporate network over the Internet. Network layer
protocols, such as IPX and NetBEUI, are encapsulated by the
PPTP for transport over the Internet. However, PPTP can support
only one tunnel at a time for each user. Therefore, its proposed
successor, L2TP (a hybrid of PPTP and another protocol, L
2F ) can support multiple, simultaneous tunnels for each user.
PPTP and L2TP are the layer 2 VPN technologies from CPE (customer
premise equipment) to CPE.
Internet
Protocol Security (IPSec), the most
widely deployed VPN technology, is a set of authentication
and encryption protocols developed by the Internet Engineering
Task Force (IETF), to address data confidentiality, integrity,
authentication and key management in the IP networks. The
IPSec protocol typically works on the edges of a security
domain, which encapsulates a packet by wrapping another packet
around it. It then encrypts the entire packet. This encrypted
stream of traffic forms a secure tunnel across an otherwise
unsecured IP network. IPsec is the primary layer 3 VPN technology
providing a CPE to CPE tunnel.
SSL/TLS,
a technology popularly used for secured communication of web
traffic (HTTPS), can also be also used for VPN. SSL VPNs use
the highly mature and widespread SSL/TLS protocol to handle
the tunnel creation and cryptographic elements necessary to
create a VPN. SSL/TLS is much easier to implement than IPSec
and provides a simple and well-tested platform. The RSA handshake
(or DH) is used exactly as IKE in IPSec, and the SSL crypto
library is used to secure the symmetric tunnel after that,
again using similar encryption techniques to those protecting
IPSec tunnels. This tunnel can pass arbitrary traffic, just
like an IPSec VPN.
The
VPN technologies popular among service providers are the border
gateway protocol/multiprotocol label switching (BGP/MPLS)
VPN. BGP/MPLS VPN is introduced to solve the scalability problems
in the traditional ATM and Frame Relay VPNs. In addition,
the MPLS VPN, a connectionless VPN, is fully compatible with
the TCP/IP technologies and the Internet world, which has
significantly lower cost of deployment and operations. The
BGP/MPLS VPN standard is defined in the IETF RFC 2547bis to
provide Layer 3 VPN solutions using BGP to carry route information
over a MPLS core. This Layer 3 MPLS-VPN solution achieves
all of the security of the Layer 2 approach, while adding
enhanced scalability inherent in the use of Layer 3 routing
technology. |
--- |
 |
|
Unified
Messaging |
 |
-- |
Unified
Messaging brings voice mail, e-mail, and fax mail services
together. It can be unified at the end user client (Outlook
or Notes for example), at the server, or at both locations.
eMail,
an integral component of most time-share computing services
in the 1960s and 1970s, only connected the users of the computer
that hosted the service. Not until 1971 when Ray Tomlinson
wrote SNDMSG and READMAIL as a network service did eMail assume
the appearance of today's requisite messaging tool.
Voice
Mail, invented by Gordon Matthews in 1979, delivers
the powerful advantage of allowing users to create their own
greetings and manage their own messages flows.
Instant
Messaging has become as pervasive as voice mail,
but with public services presents some annoying and at times
dangerous consequences, such as unwanted pop-up windows, a
multitude of advertising, and in some cases the delivery of
viruses.
Find
Me/ Follow Me capabilities put the caller on hold
while the system attempts through a pre-configured set of
numbers to notify the called party. If called parties respond
to the system, they are told of the pending call and then
can decide to accept it or not.
Fax
Mail integrates fax functions into e-mail services
and separates transmission from printing. Users send a fax
to a central server which captures the transmission and records
it in a computer-readable format such as a tiff. In some implementations,
an initial recipient can review the fax cover page online
and then forward the fax via e-mail to the correct destination.
Other systems assign each user a unique fax number which allows
incoming faxes to be automatically forwarded to the correct
recipient.
|
--- |
 |
|
Instant
Messaging |
 |
-- |
Most
Internet users know Instant Messaging through services such
as AOL Instant Messaging (AIM) or Microsoft Messenger. AIM
was derived from the ICQ model developed in 1996 as the service
of Mirabillis (acquired by AOL in 1998). According to IDC,
by 2006 there will be 255 million users worldwide—both
consumers and enterprises—of instant messaging, almost
three times the number of users in 2002. The phenomenal growth
within enterprises is equally impressive. Businesses are integrating
presence management with their IP telephony system.
Instant
message (IM) technologies allow people to talk online real
time . To s end a message, you need to open up a small window
where you and your friend can type in messages that both of
you can see. Most of the popular instant-messaging programs
provide a variety of features:
Instant
messages - Send notes back and forth with a friend
who is online
Chat - Create your own custom chat room with
friends or co-workers
Web links - Share links to your favorite
Web sites
Images - Look at an image stored on your
friend's computer
Sounds - Play sounds for your friends
Files - Share files by sending them directly
to your friends
Talk - Use the Internet instead of a phone
to actually talk with friends
Streaming content - Real-time or near-real-time
stock quotes and news
There
are many IM systems, such as AOL IM, Yahoo IM and MSN IM,
which use different technologies and they are often not compatible
with each other. There have been several attempts to create
a unified standard for instant messaging: IETF's SIP (Session
Initiation Protocol) and SIMPLE (SIP for Instant Messaging
and Presence Leverage), APEX (Application Exchange), Prim
(Presence and Instant Messaging Protocol), and the open XML-based
XMPP (Extensible Messaging and Presence Protocol), more commonly
known as Jabber. |
--- |
 |
|
802.11
Wireless LAN standard suite IEEE Quick Reference |
 |
-- |
802.11a
Wireless LAN IEEE standard with speed up to 54 Mbps
802.11b
Wireless LAN IEEE standard with speed up to 11 Mbps
802.11g
Wireless LAN IEEE standard with speed up to 54 Mbps
802.11i Wireless LAN security specification
IEEE
802.15 IEEE
Standard for short range
and low power wireless communication (Bluetooth)
802.16 IEEE Standard for metropolitan range
wireless communication (WiMax)
|
--- |
 |
|
WLAN:
Wireless LAN by IEEE 802.11, a, b, g, n |
 |
-- |
The
Wireless Local Area Networl (WLAN) technology is defined by
the IEEE 802.11 family of specifications. There are currently
four specifications in the family: 802.11, 802.11a, 802.11b,
and 802.11g. All four use the Ethernet protocol and CSMA/CA
(carrier sense multiple access with collision avoidance instead
of CSMA/CD) for path sharing.
- 802.11
-- applies to wireless LANs and provides 1 or 2 Mbps transmission
in the 2.4 GHz band using either frequency hopping spread
spectrum (FHSS) or direct sequence spread spectrum (DSSS).
- 802.11a
-- an extension to 802.11 that applies to wireless LANs
and provides up to 54 Mbps in the 5GHz band. 802.11a uses
an orthogonal frequency division multiplexing (OFDM) encoding
scheme rather than FHSS or DSSS. The 802.11a specification
applies to wireless ATM systems and is used in access hubs.
- 802.11b
(also referred to as 802.11 High Rate or Wi-Fi) -- an extension
to 802.11 that applies to wireless LANS and provides 11
Mbps transmission (with a fallback to 5.5, 2 and 1 Mbps)
in the 2.4 GHz band. 802.11b uses only DSSS. 802.11b was
a ratification to the original 802.11 standard, allowing
wireless functionality comparable to Ethernet.
- 802.11g
-- offers wireless transmission over relatively short distances
at 20 - 54 Mbps in the 2.4 GHz band. The 802.11g also uses
the OFDM encoding scheme.
- 802.11n
- builds
upon previous 802.11 standards by adding MIMO (multiple-input
multiple-output). IEEE 802.11n offers high throughput wireless
transmission at 100Mbps ? 200 Mbps.
The
modulation used in 802.11 has historically been phase-shift
keying (PSK). The modulation method selected for 802.11b is
known as complementary code keying (CCK), which allows higher
data speeds and is less susceptible to multipath-propagation
interference. 802.11a uses a modulation scheme known as orthogonal
frequency-division multiplexing (OFDM) that makes possible
data speeds as high as 54 Mbps, but most commonly, communications
takes place at 6 Mbps, 12 Mbps, or 24 Mbps.
For short range and low power
wireless (less than 10 meters) communications among personal
devices such as PDA, Bluetooth and subsequent IEEE standards
(802.15) are taking effects. For long range wireless communications
in the metropolitan areas, WiMax as defined in the IEEE 802.16
is the standard.
WLAN
protocols are defined by IEEE (http://www.ieee.org
) 802.11 specifications. |
--- |
 |
|
IEEE
802.16: Broadband Wireless MAN Standard (WiMAX) |
 |
-- |
The
IEEE 802.16 defines wireless service that provide a communications
path between a subscriber site and a core network such as
the public telephone network and the Internet. The Wireless
MAN technology is also branded as WiMAX. The WiMAX wireless
broadband access standard provides the missing link for the
"last mile" connection in metropolitan area networks
where DSL, Cable and other broadband access methods are not
available or too expensive.
IEEE
802.16 standards are concerned with the air interface between
a subscriber's transceiver station and a base transceiver
station. IEEE 802.16 is approved by th IEEE in June 2004.
Three working groups have been chartered to produce standards:
Task Group 1 of IEEE 802.16 developed a point-to-multipoint
broadband wireless access standard for systems in the frequency
range 10-66 GHz. The standard covers both the Media Access
Control (MAC) and the physical (PHY) layers. Task groups a
and b are jointly producing an amendment to extend the specification
to cover both the licensed and unlicensed bands in the 2-11
GHz range.
IEEE
802.16 and WiMAX are designed as a complimentary technology
to Wi-Fi and Bluetooth. The following table provides a quick
comparison of 802.16a with to 802.11b:
Parameters
|
802.16a
(WiMAX) |
802.11
(WLAN) |
802.15
(Bluetooth) |
Frequency
Band: |
2-11GHz
|
2.4GHz
|
Varies
|
Range
|
~31
miles |
~100
meters |
~10meters
|
Data
transfer rate: |
70
Mbps |
11
Mbps - 55 Mbps |
20Kbps
- 55 Mbps |
Number
of users: |
Thousands
|
Dozens
|
Dozens
|
Wireless
MAN is defined by the IEEE 802.16 working group (http://www.ieee.org
). |
--- |
 |
|
RJ:
Registered Jack |
 |
-- |
Registered
Jack (RJ) is a general term for electrical connector designs
registered with the US Federal Communications Commission,
including the RJ-11, RJ-14, RJ-25, RJ-48, RJ-61 and RJ-45
connectors. The most familiar registered jacks are the 4-conductor
and the 6-conductor connectors known variously as RJ-11, RJ-12
and RJ-14, and the 8-conductor RJ-45, all sometimes simply
called RJ connectors. These are commonly used in building
wiring for telephone and local area networks. They were originally
invented and patented by Bell Labs.
RJ-11:
Registered Jack-11
Registered Jack-11 (RJ-11), also called plug, is a four- or
six-wire connector used to connect telephone equipment, modems,
fax to a famle RJ-11 jack on the wall. It is occasionally
used to connect some types of local-area networks (LANs) in
some cases.
RJ-22:
Registered Jack-22
Registered Jack-22 (RJ-22) is a four wire modular jack used
for connecting telephone handsets to telephone instruments.
RJ-25C: Registered Jack-25C
Registered Jack 25C (RJ-25C) is a standard for a modular connector
using 6 conductors. It is usually used to implement a 3-line
telephone connection.
RJ-45:
Registered Jack-45
Registered Jack-45 (RJ-45) is an eight-wire connector used
to connect computers onto a local-area networks (LAN), especially
Ethernet. RJ-45 comes in two types: keyed and non-keyed.
|
--- |
 |
|
MAC
Layer
|
 |
-- |
MAC
Layer, short for Media Access Control Layer, is one of two
sublayers that make up the Data Link Layer of the OSI model.
The MAC layer is responsible for moving data packets to and
from one Network Interface Card (NIC) to another across a
shared channel. |
--- |
 |
|
MPEG:
Moving Picture Experts Group
|
|
-- |
Moving
Picture Experts Group (MPEG) is the family of digital video
compression standards and file formats developed by a working
group of ISO. MPEG generally produces better-quality video
than competing formats, such as Video for Windows, Indeo and
QuickTime. MPEG files can be decoded by special hardware or
by software. |
--- |
 |
|
Motion-JPEG |
|
-- |
Motion
JPEG (M-JPEG) is a video codec where each video field is separately
compressed into a JPEG image. The resulting quality of intraframe
video compression is independent from the motion in the image
which differs from MPEG video where quality often decreases
when footage contains lots of movement. In addition, it makes
video editing easier, as cuts may begin on any frame, not
only on the beginning of a group of frames. JPEG (Joint Photographic
Experts Group) is a standard for storing and compressing digital
images. Motion-JPEG extends this standard by supporting videos.
In motion-JPEG, each frame in the video is stored with the
JPEG format. |
--- |
 |
|
SMTP:
Simple Mail Transfer Protocol |
|
-- |
Simple
Mail Transfer Protocol (SMTP) is a protocol designed to transfer
electronic mail reliably and efficiently. SMTP is a mail service
modeled on the FTP file transfer service. SMTP transfers mail
messages between systems and provides notification regarding
incoming mail.
SMTP
is independent of the particular transmission subsystem and
requires only a reliable ordered data stream channel. An important
feature of SMTP is its capability to transport mail across
networks, usually referred to as "SMTP mail relaying".
A network consists of the mutually-TCP-accessible hosts on
the public Internet, the mutually-TCP-accessible hosts on
a firewall-isolated TCP/IP Intranet, or hosts in some other
LAN or WAN environment utilizing a non-TCP transport-level
protocol. Using SMTP, a process can transfer mail to another
process on the same network or to some other network via a
relay or gateway process accessible to both networks.
In
this way, a mail message may pass through a number of intermediate
relay or gateway hosts on its path from sender to ultimate
recipient. The Mail eXchanger mechanisms of the domain name
system are used to identify the appropriate next-hop destination
for a message being transported. |
--- |
 |
|
Video
Streaming Technologies |
 |
-- |
Streaming
is a technique for transferring data such that it can be processed
as a steady and continuous stream. Streaming technologies
are widely used in transmit large multimedia (voice, video
and data) files quickly. With streaming, the client browser
or plug-in can start displaying the multimedia data before
the entire file has been transmitted.
Video
streaming technology is developed based on (2) key technologies,
the video coding technology and scalable video distribution
technology.
Bandwidth
efficiency, scalability and flexibility between a video server
and client machine is a key issue in the video stream the
Internet is the best effort network. The scalable video distributing
technology can automatically adjust the amount of data according
to the change in bandwidth. Video streaming system consists
of an encoder, distribution server and a client that receives
the video data. The distribution server stores the encoded
video data and begins to distribute it on the client's demand.
People can watch the video whenever and wherever by accessing
the server on the Internet. Encoding and distribution is carried
out in real time in the case of live distribution. Load balance
is considered by placing the relay server in the appropriate
location on the network.
The
most important video codec standards for streaming video are
H.261, H.263, MJPEG, MPEG1, MPEG2 and H.264/MPEG4.
Compared to video codecs for CD-ROM or TV broadcast, codecs
designed for the Internet require greater scalability, lower
computational complexity, greater resiliency to network losses,
and lower encode/decode latency for video conferencing. In
addition, the codecs must be tightly linked to network delivery
software to achieve the highest possible frame rates and picture
quality.
The
transport protocols used in the video streaming are TCP,
UDP, RTP and RTSP. For reliable document (such as
HTTP files) transfer, TCP is required. UDP provides un-reliable
transport of information which can be used to stream video.
However, th most porpular transport is the Real Time
Transport Protocol (RTP), which is specially designed
for the transport of real-time data, including audio and video.
The Real Time Streaming Protocol (RTSP) is
another open standard for delivery of real-time media over
the Internet. It defines the connection between streaming
media client and server software, and provides a standard
way for clients and servers from multiple vendors to stream
multimedia content. |
--- |
 |
|
RTP:
Real-Time Transport Protocol |
 |
-- |
The
real-time transport protocol (RTP) provides end-to-end delivery
services for data with real-time characteristics, such as
interactive audio and video or simulation data, over multicast
or unicast network services. Applications typically run RTP
on top of UDP to make use of its multiplexing and checksum
services; both protocols contribute parts of the transport
protocol functionality. However, RTP may be used with other
suitable underlying network or transport protocols. RTP supports
data transfer to multiple destinations using multicast distribution
if provided by the underlying network.
RTP
itself does not provide any mechanism to ensure timely delivery
or provide other quality-of-service guarantees, but relies
on lower-layer services to do so. It does not guarantee delivery
or prevent out-of-order delivery, nor does it assume that
the underlying network is reliable and delivers packets in
sequence. The sequence numbers included in RTP allow the receiver
to reconstruct the sender's packet sequence, but sequence
numbers might also be used to determine the proper location
of a packet, for example in video decoding, without necessarily
decoding packets in sequence. RTP
is defined by IETF (http://www.ietf.org
) in RFC 3550 and 3551. |
--- |
 |
|
RTSP:
Real Time Streaming Protocol |
 |
-- |
The
Real-Time Streaming Protocol (RTSP) establishes and controls
either a single or several time-synchronized streams of continuous
media such as audio and video. RTSP does not typically deliver
the continuous streams itself, although interleaving of the
continuous media stream with the control stream is possible.
In other words, RTSP acts as a "network remote control"
for multimedia servers. RTSP provides an extensible framework
to enable controlled, on-demand delivery of real-time data,
such as audio and video. Sources of data can include both
live data feeds and stored clips. RTSP is intended to control
multiple data delivery sessions, provide a means for choosing
delivery channels such as UDP , multicast
UDP and TCP , and provide a means for choosing
delivery mechanisms bases upon RTP.
There
is no notion of an RTSP connection; instead, a server maintains
a session labeled by an identifier. An RTSP session is in
no way tied to a transport-level connection such as a TCP
connection. During an RTSP session, an RTSP client may open
and close many reliable transport connections to the server
to issue RTSP requests. Alternatively, it may use a connectionless
transport protocol such as UDP.
The
streams controlled by RTSP may use RTP, but the operation
of RTSP does not depend on the transport mechanism used to
carry continuous media. RTSP is intentionally similar in syntax
and operation to HTTP/1.1 so that extension mechanisms to
HTTP can in most cases also be added to RTSP. However, RTSP
differs in a number of important aspects from HTTP:
- RTSP
introduces a number of new methods and has a different protocol
identifier.
- An
RTSP server needs to maintain state by default in almost
all cases, as opposed to the stateless nature of HTTP.
- Both
an RTSP server and client can issue requests.
- Data
is carried out-of-band by a different protocol, in most
cases.
- RTSP
is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,consistent
with current HTML internationalization efforts.
- The
Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1
carries only the absolute path in the request and puts the
host name in a separate header field.
RTSP
is defined by IETF (http://www.ietf.org
) in RFC 2326. |
--- |
 |
|
TCP:
Transmission Control Protocol |
 |
-- |
Transmission
Control Protocol (TCP) is the transport layer protocol in
theTCP/IP protocol suite , which provides
a reliable stream delivery and virtual connection service
to applications through the use of sequenced acknowledgment
with retransmission of packets when necessary. Along with
the Internet Protocol (IP ),
TCP represents the heart of the Internet protocols.
Since
many network applications may be running on the same machine,
computers need something to make sure the correct software
application on the destination computer gets the data packets
from the source machine, and some way to make sure replies
get routed to the correct application on the source computer.
This is accomplished through the use of the TCP "port
numbers". The combination of IP address of a network
station and its port number is known as a socket or an "endpoint".
TCP establishes connections or virtual circuits between two
"endpoints" for reliable communications. Details
of TCP port numbers could be found in the TCP/UDP Port Number
document and in the reference.
Among
the services TCP provides are stream data transfer, reliability,
efficient flow control, full-duplex operation, and multiplexing.
With
stream data transfer,TCP delivers an unstructured stream
of bytes identified by sequence numbers. This service benefits
applications because that the application does not have to
chop data into blocks before handing it off to TCP. TCP can
group bytes into segments and passes them to IP for delivery.
TCP
offers reliability by providing connection-oriented, end-to-end
reliable packet delivery. It does this by sequencing bytes
with a forwarding acknowledgment number that indicates to
the destination the next byte the source expects to receive.
Bytes not acknowledged within a specified time period are
retransmitted. The reliability mechanism of TCP allows devices
to deal with lost, delayed, duplicate, or misread packets.
A time-out mechanism allows devices to detect lost packets
and request retransmission.
TCP
offers efficient flow control - when sending acknowledgments
back to the source, the receiving TCP process indicates the
highest sequence number it can receive without overflowing
its internal buffers.
Full-duplex
operation: TCP processes can both send and receive
packets at the same time.
Multiplexing in TCP: numerous simultaneous
upper-layer conversations can be multiplexed over a single
connection.
TCP
is defined by IETF (http://www.ietf.org
) RFC793. |
--- |
 |
|
TCP/IP
Protocol Suite Overview |
 |
-- |
The
TCP/IP protocol suite establishes the technical foundation
of the Internet. (UDP/IP is part of the the family). Development
of the TCP/IP was started by DOD projects and now, most protocols
in the suite are developed by the industry non-for-profit
organization named Internet Engineering Task Force (IETF)
under the Internet Architecture Board (IAB), an organization
initially sponsored by the US government and now an open and
autonomous organization. The IAB provides the coordination
for the R&D underlying the TCP/IP protocols and guides
the evolution of the Internet. The TCP/IP protocols are well
documented by the Request For Comments (RFC), which are drafted,
discussed, circulated and approved by the IETF committees.
All documents are open and free and could be found online
in the IETF site listed in the reference.
TCP/IP protocols cover 6 layers in the OSI
network architecture 7 layer model and providing functions
from switching (layer 2) such as MPLS
to applications such as mail services (POP3 and SMTP
). Its core functions are addressing and routing (IP /IPv6
in the networking layer) and transport (TCP , UDP
in the transport layer).
IP
- Internet Protocol
Addressing
of network components is a critical issue in the network communications
for information routing and transmission.Each technology has
its own convention for transmitting messages between two machines
within the same network. On a LAN, messages are sent between
machines by supplying the six byte unique identifier (the
"MAC" address). In an SNA network, every machine
has Logical Units with their own network address. DECNET ,
Appletalk , and Novell IPX all have a scheme for assigning
numbers to each local network and to each workstation attached
to the network.
On
top of these local or vendor specific network addresses, IP
assigns a unique number to every network device in the world,
which is called IP address. This IP address is a four byte
value in IPv4 that, by convention, is expressed by converting
each byte into a decimal number (0 to 255) and separating
the bytes with a period. In IPv6, the IP address has been
increased to 16 bytes.
TCP
- Transmission Control Protocol
TCP
provides a reliable stream delivery and virtual connection
service to applications through the use of sequenced acknowledgment
with retransmission of packets when necessary. Among the services
TCP provides are stream data transfer, reliability, efficient
flow control, full-duplex operation, and multiplexing.
TCP/IP
is defined by IETF (http://www.ietf.org
) RFC793. |
--- |
 |
|
UDP:
User Datagram Protocol |
 |
-- |
UDP
is a connectionless transport layer (layer 4) protocol in
OSI mode which provides a simple
and unreliable message service for transaction-oriented services.
UDP is basically an interface between IP and upper-layer processes.
UDP protocol ports distinguish multiple applications running
on a single device from one another.
Since
many network applications may be running on the same machine,
computers need something to make sure the correct software
application on the destination computer gets the data packets
from the source machine, and some way to make sure replies
get routed to the correct application on the source computer.
This is accomplished through the use of the UDP "port
numbers". For example, if a station wished to use a Domain
Name System (DNS) on the station 128.1.123.1, it would address
the packet to station 128.1.123.1 and insert destination port
number 53 in the UDP header. The source port number identifies
the application on the local station that requested domain
name server, and all response packets generated by the destination
station should be addressed to that port number on the source
station. Details of UDP port numbers could be found in the
TCP/UDP Port Number document and
in the reference.
Unlike
the TCP , UDP adds no reliability, flow-control,
or error-recovery functions to IP. Because of UDP's simplicity,
UDP headers contain fewer bytes and consume less network overhead
than TCP.
UDP
is useful in situations where the reliability mechanisms of
TCP are not necessary, such as in cases where a higher-layer
protocol might provide error and flow control, or real time
data transportation is required.
TCP
is defined by IETF (http://www.ietf.org
) RFC768. |
--- |
 |
|
TCP
UDP Port Numbers |
 |
-- |
TCP
and UDP are both transport protocols above the IP layer, which
are interfaces between IP and upper-layer processes. TCP and
UDP protocol port numbers are designed to distinguish multiple
applications running on a single device from one another.
Since
many network applications may be running on the same machine,
computers need something to make sure the correct software
application on the destination computer gets the data packets
from the source machine, and some way to make sure replies
get routed to the correct application on the source computer.
This is accomplished through the use of the TCP or UDP "port
numbers". In the TCP and UDP header, there are "Source
Port" and "Destination Port" fields which are
used to indicate the message sending process and receiving
process identities defined. The combination of the IP address
and the port number is called "socket".
There
three port ranges defined by IETF IANA: The Well Known Ports,
the Registered Ports, and the Dynamic and/or Private Ports.
- The
Well Known Ports are those from 0 through 1023.
- The
Registered Ports are those from 1024 through 49151.
- The
Dynamic and/or Private Ports are those from 49152 through
65535
Partial TCP UDP Port Numbers Well-Known Ports |
|
Port
No.
| Protocol
| Service
Name
| Aliases |
Comment |
7 |
TCP |
echo |
|
Echo |
7 |
UDP |
echo |
|
Echo |
9 |
TCP |
discard |
sink
null |
Discard |
9 |
UDP |
discard |
sink
null |
Discard |
13 |
TCP |
daytime |
|
Daytime |
13 |
UDP |
daytime |
|
Daytime |
17 |
TCP |
qotd |
quote |
Quote
of the day |
17 |
UDP |
qotd |
quote |
Quote
of the day |
19 |
TCP |
chargen |
ttytst
source |
Character
generator |
19 |
UDP |
chargen |
ttytst
source |
Character
generator |
20 |
TCP |
ftp-data |
|
File
Transfer |
21 |
TCP |
ftp |
|
FTP
Control |
23 |
TCP |
telnet |
|
Telnet |
25 |
TCP |
smtp |
mail |
Simple
Mail Transfer |
37 |
TCP |
time |
|
Time |
37 |
UDP |
time |
|
Time |
39 |
UDP |
rlp |
resource |
Resource
Location Protocol |
42 |
TCP |
nameserver |
name |
Host
Name Server |
42 |
UDP |
nameserver |
name |
Host
Name Server |
43 |
TCP |
nicname |
whois |
Who
Is |
53 |
TCP |
domain |
|
Domain
Name |
53 |
UDP |
domain |
|
Domain
Name Server |
67 |
UDP |
bootps |
dhcps |
Bootstrap
Protocol Server |
68 |
UDP |
bootpc |
dhcpc |
Bootstrap
Protocol Client |
69 |
UDP |
tftp |
|
Trivial
File Transfer |
70 |
TCP |
gopher |
|
Gopher |
79 |
TCP |
finger |
|
Finger |
80 |
TCP |
http |
www,
http |
World
Wide Web |
88 |
TCP |
kerberos |
krb5 |
Kerberos |
88 |
UDP |
kerberos |
krb5 |
Kerberos |
101 |
TCP |
hostname |
hostnames |
NIC
Host Name Server |
102 |
TCP |
iso-tsap |
|
ISO-TSAP
Class 0 |
107 |
TCP |
rtelnet |
|
Remote
Telnet Service |
109 |
TCP |
pop2 |
postoffice |
Post
Office Protocol - Version 2 |
110 |
TCP |
pop3 |
postoffice |
Post
Office Protocol - Version 3 |
111 |
TCP |
sunrpc |
rpcbind
portmap |
SUN
Remote Procedure Call |
111 |
UDP |
sunrpc |
rpcbind
portmap |
SUN
Remote Procedure Call |
113 |
TCP |
auth |
ident
tap |
Authentication
Sevice |
117 |
TCP |
uucp-path |
|
UUCP
Path Service |
119 |
TCP |
nntp |
usenet |
Network
News Transfer Protocol |
123 |
UDP |
ntp |
|
Network
Time Protocol |
135 |
TCP |
epmap |
loc-srv |
DCE
endpoint resolution |
135 |
UDP |
epmap |
loc-srv |
DCE
endpoint resolution |
137 |
TCP |
netbios-ns |
nbname |
NETBIOS
Name Service |
137 |
UDP |
netbios-ns |
nbname |
NETBIOS
Name Service |
138 |
UDP |
netbios-dgm |
nbdatagram |
NETBIOS
Datagram Service |
139 |
TCP |
netbios-ssn |
nbsession |
NETBIOS
Session Service |
143 |
TCP |
imap |
imap4 |
Internet
Message Access Protocol |
158 |
TCP |
pcmail-srv |
repository |
PC
Mail Server |
161 |
UDP |
snmp |
snmp |
SNMP |
162 |
UDP |
snmptrap |
snmp-trap |
SNMP
TRAP |
170 |
TCP |
print-srv |
|
Network
PostScript |
179 |
TCP |
bgp |
|
Border
Gateway Protocol |
194 |
TCP |
irc |
|
Internet
Relay Chat Protocol |
213 |
UDP |
ipx |
|
IPX
over IP |
389 |
TCP |
ldap |
|
Lightweight
Directory Access Protocol |
443 |
TCP |
https |
MCom |
|
443 |
UDP |
https |
MCom |
|
445 |
TCP |
|
|
Microsoft
CIFS |
445 |
UDP |
|
|
Microsoft
CIFS |
464 |
TCP |
kpasswd |
|
Kerberos
(v5) |
464 |
UDP |
kpasswd |
|
Kerberos
(v5) |
500 |
UDP |
isakmp |
ike |
Internet
Key Exchange (IPSec) |
512 |
TCP |
exec |
|
Remote
Process Execution |
512 |
UDP |
biff |
comsat |
Notifies
users of new mail |
513 |
TCP |
login |
|
Remote
Login |
513 |
UDP |
who |
whod |
Database
of who's logged on, average load |
514 |
TCP |
cmd |
shell |
Automatic
Authentication |
514 |
UDP |
syslog |
|
|
515 |
TCP |
printer |
spooler |
Listens
for incoming connections |
517 |
UDP |
talk |
|
Establishes
TCP Connection |
518 |
UDP |
ntalk |
|
|
520 |
TCP |
efs |
|
Extended
File Name Server |
520 |
UDP |
router |
router
routed |
RIPv.1,
RIPv.2 |
525 |
UDP |
timed |
timeserver |
Timeserver |
526 |
TCP |
tempo |
newdate |
Newdate |
530 |
TCP,UDP |
courier |
rpc |
RPC |
531 |
TCP |
conference |
chat |
IRC
Chat |
532 |
TCP |
netnews |
readnews |
Readnews |
533 |
UDP |
netwall |
|
For
emergency broadcasts |
540 |
TCP |
uucp |
uucpd |
Uucpd |
543 |
TCP |
klogin |
|
Kerberos
login |
544 |
TCP |
kshell |
krcmd |
Kerberos
remote shell |
550 |
UDP |
new-rwho |
new-who |
New-who |
556 |
TCP |
remotefs |
rfs
rfs_server |
Rfs
Server |
560 |
UDP |
rmonitor |
rmonitord |
Rmonitor |
561 |
UDP |
monitor |
|
|
636 |
TCP |
ldaps |
sldap |
LDAP
over TLS/SSL |
749 |
TCP |
kerberos-adm |
|
Kerberos
administration |
749 |
UDP |
kerberos-adm |
|
Kerberos
administration |
|
|
TCP and UDP port numbers are defined by IETF (http://www.ietf.org
). |
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RMON:
Remote Monitoring MIBs (RMON1 and RMON2) |
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-- |
Remote
Monitoring (RMON) is a standard monitoring specification that
enables various network monitors and console systems to exchange
network-monitoring data. RMON provides network administrators
with more freedom in selecting network-monitoring probes and
consoles with features that meet their particular networking
needs.
RMON
was originally developed to address the problem of managing
LAN segments and remote sites from a central location. The
RMON specification, which is an extension of the SNMP MIB,
is a standard monitoring specification. Within an RMON network
monitoring data is defined by a set of statistics and functions
and exchanged between various different monitors and console
systems. Resultant data is used to monitor network utilization
for network planning and performance-tuning, as well as assisting
in network fault diagnosis.
There
are 2 versions of RMON: RMON1 (RMONv1)and RMON2 (RMONv2).
RMON1 defined 10 MIB groups for basic network monitoring,
which can now be found on most modern network hardware. RMON2
(RMONv2) is an extension of RMON that focuses on higher layers
of traffic above the medium access-control (MAC) layer. RMON2
has an emphasis on IP traffic and application-level traffic. RMON2
allows network management applications to monitor packets
on all network layers. This is difference from RMON which
only allows network monitoring at MAC layer or below.
RMON
solutions are comprised of two components: a probe (or an
agent or a monitor), and a client, usually a management station.
Agents store network information within their RMON MIB and
are normally found as embedded software on network hardware
such as routers and switches although they can be a program
running on a PC. Agents can only see the traffic that flows
through them so they must be placed on each LAN segment or
WAN link that is to be monitored. Clients, or management stations,
communicate with the RMON agent or probe, using SNMP to obtain
and correlate RMON data.
Now,
there are a number of variations to the RMON MIB. For example,
the Token Ring RMON MIB provides objects specific to managing
Token Ring networks. The SMON MIB extends RMON by providing
RMON analysis for switched networks.
RMON
is defined by IETF (http://www.ietf.org
) through a group of RFCs shown in the reference. |
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|
APPN:
Advanced Peer-to-Peer Networking |
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-- |
Advanced
Peer-to-Peer Networking (APPN) is an enhancement to the original
IBM SNA architecture . APPN, which
includes a group of protocols and processors, handles session
establishment between peer nodes, dynamic transparent route
calculation, and traffic prioritization. Using APPN, a group
of computers can be automatically configured by one of the
computers acting as a network controller so that peer programs
in various computers will be able to communicate with other
using specified network routing.
APPN
features include:
- Better
distributed network control; because the organization is
peer-to-peer rather than solely hierarchical, terminal failures
can be isolated
- Dynamic
peer-to-peer exchange of information about network topology,
which enables easier connections, reconfigurations, and
routing
- Dynamic
definition of available network resources
- Automation
of resouce registration and directory lookup
- Flexibility,
which allows APPN to be used in any type of network topology
An
APPN network is composed of three types of APPN node:
- Low
Entry Networking (LEN) Node - APPN LEN node provides
peer to peer connectivity with all other APPN nodes.
- End
Node- An End Node is similar to a LEN node in that it
participates at the periphery of an APPN network. An End
Node includes a Control Point (CP) for network control information
exchange with an adjacent network node.
- Network
Node - The backbone of an APPN network is composed of
one or more Network Nodes which provide network services
to attached LEN and End Nodes.
The
APPN network have the following major functional processors:
Connectivity-
The first phase of operation in an APPN network is to establish
a physical link between two nodes. When it has been established,
the capabilities of the two attached nodes are exchanged using
XIDs. At this point, the newly attached node is integrated
into the network.
Location
of a Targeted LU- Information about the resources (currently
only LUs) within the network is maintained in a database which
is distributed across the End and Network Nodes in the network.
End Nodes hold a directory of their local LUs. If the remote
LU is found in the directory, a directed search message is
sent across the network to the remote machine to ensure that
the LU has not moved since it was last used or registered.
If the local search is unsuccessful, a broadcast search is
initiated across the network. When the node containing the
remote LU receives a directed or broadcast search message,
it sends back a positive response. A negative response is
sent back if a directed or broadcast search fails to find
the remote LU.
Route
Selection- When a remote LU has been located, the originating
Network Node server calculates the best route across the network
for a session between the two LUs. Every Network Node in the
APPN network backbone maintains a replicated topology database.
This is used to calculate the best route for a particular
session, based on the required class of service for that session.
The class of service specifies acceptable values for session
parameters such as propagation delay, throughput, cost and
security. The route chosen by the originating Network Node
server is encoded in a route selection control vector (RSCV).
Session
Initiation - A BIND is used to establish the session.
The RSCV describing the session route is appended to the BIND.
The BIND traverses the network following this route. Each
intermediate node puts a session connector for that session
in place, which links the incoming and outgoing paths for
data on the session.
Data
Transfer- Session data follows the path of the session
connectors set up by the initial BIND. Adaptive pacing is
used between each node on the route. The session connectors
on each intermediate node are also responsible for segmentation
and segment assembly when the incoming and outgoing links
support different segment sizes.
Dependent
LU Requestor- Dependent LUs require a host based System
Services Control Point (SSCP) for LU-LU session initiation
and management. This means that dependent LUs must be directly
attached to a host via a single data link.
High-performance
routing (HPR)- HPR is an extension to the APPN architecture.
HPR can be implemented on an APPN network node or an APPN
end node. HPR does not change the basic functions of
the architecture. HPR has the following key functions:
- Improves
the performance of APPN routing by taking advantage of high-speed,
reliable links
- Improves
data throughput by using a new rate-based congestion control
mechanism
- Supports
nondisruptive re-routing of sessions around failed links
or nodes
- Reduces
the storage and buffering required in intermediate nodes.
APPN
is an IBM network architecture, extended from the IBM
SNA.
|
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IP-PBX
FACTS |
|
Business
Phone Systems for Enhanced Business Operations |
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-- |
When
making a business phone system decision, businesses first
need to determine which type of phone system best meets their
business needs. Will a traditional phone system or an IP phone
system provide the benefits that are key to business operations?
IP
phones systems can provide unique advantages:
Lower
total cost of ownership. An IP phone system can lower total
cost of ownership for businesses. Long distance charges for
remote offices, the expense of teleworking, and international
travellers' phone charges can be dramatically reduced with
an IP business phone system. Moving an employee is as simple
as unplugging a telephone and plugging it in at a new location—as
opposed to costly service calls from legacy phone system vendors.
IP phone system business owners can converge their data and
voice communications, avoiding the need to deploy and manage
two separate networks and infrastructure. For example, running
a traditional phone line to each employee's desk can cost
over $100 per line. When setting up a new office, why pay
the hundreds or thousands of dollars in cabling?
In
addition to the hard dollar savings, an IP phone system can
make workers more productive. A unified dial plan gives employees
easy and fast access to each other, regardless of their location,
saving time every day for every employee. Over the course
of a year that savings can add up. And overworked IT staff
can easily configure the IP phone system using a web browser
instead of the complicated interfaces from legacy vendors.
Enhance
business revenue. The abundance of applications supported
by an IP phone system can enhance customer service and improve
business performance. And integrating the IP phone system
with business software such as call center and conferencing
applications can boost top-line revenue. |
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Copyright
© 2019 LAN-COMM Technologies, Inc. - All rights reserved. |
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