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Quick Links To Resources Below
Technology Defenitions More Technology Defenitions White Papers
Network Devices and Components MPLS (Multiprotocol Label Switching) What’s SIP Got To Do With It?
What is a Proxy Gateway VPLS (Virtual Private LAN Services) Taking the Guesswork Out of IP Telephony
What is a Server VPN (Virtual Private Network) Architecture for Convergence
What is a Router What is Unified Messaging A Roadmap for Convergence
What is a Hub What is Instant Messaging Secure IP Telepony for the Enterprise
What is a Switch 802.11 Wireless LAN Quick Reference SIP Market Overview
Switch Types (Layer) What is WiFi and WLAN  
Networking Standards and Protocols What is WiMax Additional White Papers
LAN Architecture and Topologies RJ (Registered Jacks) 3Com Jargon Buster and Glossary
Ethernet protocols What is the MAC Layer Zultys ROI for a Single Office
Fast Ethernet (100BASE-T) What is a MPEG Tutorial on Bridges, Routers, Switches
Gigabit Ethernet (1000 Mbps) What is a M-JPEG (Motion JPEG) FCC VoIP Consumer Fact Sheet
10-Gigabit Ethernet SMTP (Simple Mail Transfer Protocol) New Approach to Enterprise Comm.
OSI 7 Layers Reference Model Video Streaming Technologies  
IBM SNA (Systems Ntwk. Architecture) RTP (Real-Time Transport Protocol IP-PBX FACTS 
VoIP (Voice Over IP) RTSP (Real-Time Streaming Protocol) Phone Systems for Enhanced Business
H.323 VOIP Protocol TCP (Transmission Control Protocol) IP-PBX Takes Digital Ntwkg. Step Further
SIP (Session Initiation Protocol) TCP/IP protocol  
Megaco ( Media Gateway Control Protocol) UDP (User Datagram Protocol)  
MGCP (Media Gateway Control Protocol) UDP Ports  
PPPoE: Power Over Ethernet Overview What is SNMP  
NAT (Network Address Translation) RMON (Remote Monitoring)  
VoIP for the Telecommuter  APPN (Advanced Peer-to-Peer Ntwkg.)  
Layer 3 IP VPN -- --
 
WHITE PAPERS
 
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)
 

 
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)
 

 
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)
 

 
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)
 

 
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)
 

 
DEFENITIONS  
   
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 IEEEs 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
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What is Voice Over Internet Protocol (VoIP)?  
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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.

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H.323 VOIP Protocol

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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.

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Session Initiation Protocol (SIP)  
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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.

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Megaco/H.248: Media Gateway Control Protocol  
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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.

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MGCP/Media Gateway Control Protocol
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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.

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VoIP for the Telecommuter  
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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.
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PPPoE: Power Over Ethernet Overview
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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.

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NAT: IP Network Address Translator (Network Address Translation)
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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.

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Layer 3 IP VPN: Internet Protocol Virtual Private Network  
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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.

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MPLS: Multiprotocol Label Switching
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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.

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IPsec: Security Architecture for IP Network
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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.

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Layer 2 Ethernet VPN and Virtual Private LAN Services (VPLS)
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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.

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VPN: Virtual Private Network
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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.

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Unified Messaging
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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.

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Instant Messaging
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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.

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802.11 Wireless LAN standard suite IEEE Quick Reference
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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)

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WLAN: Wireless LAN by IEEE 802.11, a, b, g, n 
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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.

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IEEE 802.16: Broadband Wireless MAN Standard (WiMAX) 
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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 ). 
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RJ: Registered Jack
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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.

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MAC Layer
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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.
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MPEG: Moving Picture Experts Group   
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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.
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Motion-JPEG  
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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.
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SMTP: Simple Mail Transfer Protocol   
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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.

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Video Streaming Technologies
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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.

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RTP: Real-Time Transport Protocol
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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.

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RTSP: Real Time Streaming Protocol
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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.

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TCP: Transmission Control Protocol 
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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.

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TCP/IP Protocol Suite Overview
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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.

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UDP: User Datagram Protocol 
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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.

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TCP UDP Port Numbers
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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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