The Synchronous Optical Network (SONET) and Synchronous Digital Hierarchy (SDH) are a set of related standards for synchronous data transmission over fiber optic networks that are often used for framing and synchronization at the physical layer. SONET is the
SONET/SDH can be used in an ATM or non-ATM environment. Packet Over SONET/SDH (POS) maps IP datagrams into the SONET frame payload using Point-to-Point Protocol (PPP). In the ATM environment, connections to SONET/SDH lines may be via multi-mode, single-mode or UTP.
The Asynchronous Transfer Mode (ATM) composes a protocol suite which establishes a mechanism to carry all traffic on a stream of fixed 53-byte packets (cells). A fixed-size packet can ensure that the switching and multiplexing function could be carried out quickly and easily. ATM is a connection-oriented technology, i.e.; two systems on the network should inform all intermediate switches about their service requirements and traffic parameters in order to establish communication.
The ATM reference model, which has two forms - one for the user-to-network interface (UNI) and the other for the network-to-node interface (NNI), is divided into three layers: the ATM adaptation layer (AAL), the ATM layer, and the physical layer. The AAL interfaces the higher layer protocols to the ATM Layer, which relays ATM cells both from the upper layers to the ATM Layer and vice versa. When relaying information received from the higher layers, the AAL segments the data into ATM cells. When relaying information received from the ATM Layer, the AAL must reassemble the payloads into a format the higher layers can understand. This is called Segmentation and Reassembly (SAR). Different AALs are defined in supporting different types of traffic or service expected to be used on ATM networks.
The ATM layer is responsible for relaying cells from the AAL to the physical layer for transmission and from the physical layer to the AAL for use at the end systems, it determines where the incoming cells should be forwarded to, resets the corresponding connection identifiers and forwards the cells to the next link, as well as buffers cells, and handles various traffic management functions such as cell loss priority marking, congestion indication, and generic flow control access. It also monitors the transmission rate and conformance to the service contract (traffic policing).
The physical layer of ATM defines the bit timing and other characteristics for encoding and decoding the data into suitable electrical/optical waveforms for transmission and reception on the specific physical media used. In addition, it also provides frame adaptation function, which includes cell delineation, header error check (HEC) generation and processing, performance monitoring, and payload rate matching of the different transport formats used at this layer. SONET, DS3, Fiber, twisted-pair are few media often used at the physical layer.
Customer Premises Equipment - refers to all ISDN compatible equipment connected at the user side. Examples of devices are telephone, PC, Telex, Facsimile, etc. The exception is the FCC definition of NT1. The FCC views the NT1 as a CPE because it is on the customer side, but the CCITT views NT1 as part of the network.
DSLAM: Digital Subscriber Line Access Multiplexer
Digital Subscriber Line Access Multiplexer (DSLAM), a system in the DLS technology architecture, links and aggregates many customer DSL connections to a single high-speed ATM line. To enable DSL technology, service providers must have a DSLAM located in their networks to interact with the customer premises equipment (CPE) at the end-user location.
DSL technology is a platform for delivering broadband services to homes and small businesses. DSL can support a wide variety of high-bandwidth applications, such as high-speed Internet access, telecommuting, virtual private networking, and streaming multimedia content. DSL can transmit more than 8 Mbps to a subscriber enough to provide Internet access, video on demand (VOD), and local-area network (LAN) access. This increases the existing access capacity by more than fifty fold, enabling the transformation of the existing public network. No longer is this network limited to voice, text, and low-resolution graphics. It promises to be nothing less than a ubiquitous system that can provide multimedia (including full-motion video) around the globe.
A DSLAM separates the voice-frequency signals from the high-speed data traffic and controls and routes digital subscriber line (xDSL) traffic between the subscriber's end-user equipment (router, modem, or network interface card [NIC] and the network service provider's network.
A multiservice DSLAM is a broadband-access network element (NE) that combines support for multiple DSL transmission types. When coupled with high-capacity asynchronous transfer mode (ATM) switching, multiservice DSLAMs deliver scalability, port density, and a redundant architecture for reliability. Multiservice DSLAMs, together with various CPE elements, can enable the relatively efficient deployment of broadband networks for high-speed Internet access as well as voice and video applications. Such DSLAMs often allow for full ATM switching, traffic management, and quality of service (QoS), in addition to the delivery of a full range of services. These services include analog, ISDN, IDSL, SDSL, rate-adaptive DSLCcompetive access provider (RADSLCCAP), and RADSLC discrete multitone (DMT) on a single platform.The multiservice DSLAM can also be configured to add value in the form of routing and security functionality. The device is intended to enable service providers to optimize the bandwidth of existing infrastructure as well as deliver high-speed, integrated services over a single physical-access medium.
DSL, xDSL
DSL (Digital Subscriber Line) is a modem technology for broadband data access over ordinary copper telephone lines (POTS) from homes and businesses. xDSL refers collectively to all types of DSL, such as ADSL (and G.Lite), HDSL, SDSL, IDSL and VDSL etc. They are sometimes referred to as last-mile (or first mile) technologies because they are used only for connections from a telephone switching station to a home or office, not between switching stations.
xDSL is similar to ISDN in as much as both operate over existing copper telephone lines (POTS) using sophisticated modulation schemes and both require the short runs to a central telephone office (usually less than 20,000 feet). However, xDSL offers much higher speeds - up to 32 Mbps for upstream traffic, and from 32 Kbps to over 1 Mbps for downstream traffic.
Several modulation technologies are used by various kinds of DSL:
- Discrete Multitone Technology (DMT)
- Simple Line Code (SLC)
- Carrierless Amplitude Modulation (CAP)
- Multiple Virtual Line (MVL)
- Discrete Wavelet Multitone (DWMT)
To interconnect multiple DSL users to a high-speed backbone network, the telephone company uses a Digital Subscriber Line Access Multiplexer (DSLAM). The DSLAM aggregates data transmission from all access DSL lines and then connects to an asynchronous transfer mode (ATM) network. At the other end of each transmission, a DSLAM demultiplexes the signals and forwards them to appropriate individual DSL connections.
CSU/DSU: Channel Service Unit/Data Service Unit
Channel Service Unit is a device that converts a digital data frame from the communications technology used on a local area network (LAN) into a frame appropriate to a wide-area network (WAN) and vice versa. The DSU is a device that performs protective and diagnostic functions for a telecommunications line. Typically, the two devices are integrated into a single unit. It is actually a complicate and expensive modem. As an example of using the CSU/DSU, if you have leased a digital line (perhaps a T1, T3 or fractional T1/T3 line) for your business from a phone company or an ISP, you need a CSU/DSU at your end and the phone company or ISP need to have a CSU/DSU at its end.
The Channel Service Unit (CSU) receives and transmits signals from and to the WAN line and provides a barrier for electrical interference from either side of the unit. The CSU can also echo loopback signals from the phone company for testing purposes. The Data Service Unit (DSU) manages line control, and converts input and output between RS-232C, RS-449, or V.35 frames from the LAN and the time-division multiplexed (TDM) DSX frames on the T-1 line. The DSU manages timing errors and signal regeneration. The DSU provides a modem-like interface between the computer as Data Terminal Equipment (DTE) and the CSU.
CSU/DSUs are made as separate products or are sometimes part of a T-1 WAN card. A CSU/DSU's Data Terminal Equipment interface is usually compatible with the V.xx and RS-232C or similar serial interface.
X.25: ITU-T Protocol for WAN Communications
X.25, an ITU-T protocol for WAN communications, is a packet switched data network protocol which defines the exchange of data as well as control information between a user device, called Data Terminal Equipment (DTE) and a network node, called Data Circuit Terminating Equipment (DCE).
X.25 is designed to operate effectively regardless of the type of systems connected to the network. X.25 is typically used in the packet-switched networks (PSNs) of common carriers, such as the telephone companies. Subscribers are charged based on their use of the network. X.25 utilizes a Connection-Oriented service which insures that packets are transmitted in order.
X.25 sessions are established when one DTE device contacts another to request a communication session. The DTE device that receives the request can either accept or refuse the connection. If the request is accepted, the two systems begin full-duplex information transfer. Either DTE device can terminate the connection. After the session is terminated, any further communication requires the establishment of a new session. X.25 uses virtual circuits for packets communications. Both switched and permanent virtual circuits are used.
X.75 is the signaling protocol for X.25, which defines the signaling system between two PDNs. X.75 is essentially an Network to Network Interface (NNI).
X.25 protocol suite comes with three levels based on the first three layers of the OSI seven layers architecture.
The Physical Level: describes the interface with the physical environment. There are three protocols in this group: 1) X.21 interface operates over eight interchange circuits; 2) X.21bis defines the analogue interface to allow access to the digital circuit switched network using an analogue circuit; 3) V.24 provides procedures which enable the DTE to operate over a leased analogue circuit connecting it to a packet switching node or concentrator.
The Link Level: responsible for the reliable communication between the DTE and the DCE. There are four protocols in this group: 1) LAPB, derived from HDLC and the most commonly used, enables to form a logical link connection besides all the other characteristics of HDLC; 2) Link Access Protocol (LAP) is an earlier version of LAPB and is seldom used today; 3) LAPD, derived from LAPB and used for ISDN, enables data transmission between DTEs through D channel, especially between a DTE and an ISDN node; 4) Logical Link Control (LLC), an IEEE 802 LAN protocol, enables X.25 packets to be transmitted through a LAN channel.
The Packet Layer Protocol (PLP): describes the data transfer protocol in the packet switched network at the network layer (layer 3). PLP manages packet exchanges between DTE devices across virtual circuits. PLPs also can run over Logical Link Control 2 implementations on LANs as well as over ISDN interfaces running LAPD. The PLP operates in five distinct modes: call setup, data transfer, idle, call clearing, and restarting.
- Call setup mode is used to establish SVCs between DTE devices.
- Data transfer mode is used for transferring data between two DTE devices across a virtual circuit.
- Idle mode is used when a virtual circuit is established but data transfer is not occurring.
- Call clearing mode is used to end communication sessions between DTE devices and to terminate SVCs.
Virtual Switching
The Virtual Switching is a technology allowing multiple switching functions to happen in one physical device or single functional switching to happen in multiple physical devices across network, while in the situation of a real switch, switching is conducted in one physical switch. Virtual switching technology enables a single switch to be used for many different applications. Each different function may have its own discrete performance and security controls. Using virtual switching technology, Service providers can create a dynamic service mix, as desired, without requiring new hardware and enabling the gradual evolution to include new services or a common control plane. Virtual switching is a core concept in the Multi service network switching architecture.
Multi service Switching Forum (MSF) defined an architectural framework for a multiplane system to provide Virtual Switching functions, which provides a high degree of flexibility, simultaneously supporting Asynchronous Transfer Mode (ATM), voice, and Internet Protocol (IP) services, with separate control planes for maximum service efficiency.
Unified Messaging
Unified Messaging (UM) is the integration of different streams of communication (e-mail, SMS, Fax, voice, video, etc.) into a single, or, unified 'message store', accessible from a variety of different devices. Unified messaging is a subset of a fully integrated Unified communications system.
Having access to e-mail, voice mail and faxes via a common computer application or by telephone. For example, unified messaging may send faxes and digitized voice mail to a mail server that turns them into e-mail attachments. Audio-based systems convert e-mail messages to speech (text-to-speech) in order to deliver messages to a desk phone or cell phone.
Softswitch
Softswitch is the next generation voice and multimedia switch based on the IP technologies. It is design to replace the Class 5 and Class 4 switches based on the circuit switching technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers, while the traditional Class 5 and Class 4 switches are rely on dedicated facilities for inter-connection and are designed primarily for voice communications. Sometimes Call Agent or Media Gateway Controller, a key component in the VOIP solution, is also called Softswith, though the definition of Softswith is often extend to the whole solution.
- The advantages of the Softswitch vs. the traditional circuit switch are:
- New services and revenue stream for service providers
- Flexibility in deployment and operation
- Unified messaging
- Easy integration of dissimilar networks and components
- Lower cost of solution deployment and total ownership
Softswitch technology enables connectivity between the Internet, wireless networks, cable networks and traditional wireline telephony network, which results a converged network.
Circuit Switching
Circuit switching is a process that establishes connections on demand and permits exclusive use of those connections until they are released. A circuit-switched network is a type of network in which a physical path is obtained for, and dedicated to, a single connection between two end-points in the network during the connection. The traditional voice phone service using PSTN (not the voice over IP) is circuit-switched. The telephone company reserves a specific physical path to the number you are calling during your call and the physical lines involved are used exclusively between the parties at the two end-points.
Circuit-switching is often compared with packet-switching. The main difference in Packet Switching from Circuit Switching is that the communication lines are not dedicated to passing messages from the source to the destination. Circuit Switching is ideal when data must be transmitted quickly, must arrive in sequencing order and at a constant arrival rate. Thus, when transmitting real time data, such as audio and video when quality of service (QOS) is highly desired, a Circuit Switched network is often used. Packet Switching is more efficient and robust for data that is bursty in its nature and can withstand delays and jitter in transmission, such as e-mail messages and Web pages.
Some packet-switched networks, such as the X.25 and ATM networks, are able to have virtual circuit-switching. A virtual circuit-switched connection is a dedicated logical connection that allows sharing of the physical path among multiple virtual circuit connections.
Voice Over IP and VOIP Protocols
Voice over IP (VOIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network. 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.
There are a few VOIP protocol stacks which are derived from various standard bodies and vendors, namely H.323, SIP, MEGACO and MGCP.
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.
Session Initiation Protocol (SIP) is the IETF's standard for establishing VOIP connections. SIP is an application layer control protocol for creating, modifying and terminating sessions with one or more participants. The architecture of SIP is similar to that of HTTP (client-server protocol). Requests are generated by the client and sent to the server. The server processes the requests and then sends a response to the client. A request and the responses for that request make a transaction.
Media Gateway Control Protocol (MGCP) is a Cisco and Telcordia proposed VOIP protocol that defines communication between call control elements (Call Agents or Media Gateway) and telephony gateways. MGCP is a control protocol, allowing a central coordinator to monitor events in IP phones and gateways and instructs them to send media to specific addresses. In the MGCP architecture, The call control intelligence is located outside the gateways and is handled by the call control elements (the Call Agent). Also the call control elements (Call Agents) will synchronize with each other to send coherent commands to the gateways under their control.
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.
SIP: Session Initiation Protocol
SIP supports five facets of establishing and terminating multimedia communications:
- User location: determination of the end system to be used for communication;
- User availability: determination of the willingness of the called party to engage in communications;
- User capabilities: determination of the media and media parameters to be used;
- Session setup: "ringing", establishment of session parameters at both called and calling party;
- Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.
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.
Callers and callees are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers. SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations such as Click to talk.