TAPI 3.0

Microsoft announced TAPI 3.0 in mid-September, 1997. At that time, Mitch Goldberg of Microsoft, said TAPI 3.0 will be available initially in NT 5 beta 1 in September, fully featured in NT 5 beta 2, and available commercially when the NT 5.0 operating system ships (around the middle of 1998). TAPI 3.0 will also be available for Windows 98 ("Memphis") in the NT 5.0 time frame. microsoft.com/communications

At the same time, Microsoft released a White Paper titled "IP Telephony with TAPI 3.0." Here are key excerpts. For the full paper, www.ctexpo.com. TAPI 3.0 is an evolutionary API providing convergence of both traditional PSTN telephony and IP Telephony. IP Telephony is an emerging set of technologies which enables voice, data, and video collaboration over existing LANs, WANs and the Internet. TAPI 3.0 enables IP Telephony on the Microsoft Windows operating system platform by providing simple and generic methods for making connections between two or more machines, and accessing any media streams involved in the connection. TAPI 3.0 supports standards based H.323 conferencing and IP multicast conferencing. It utilizes the Windows NT 5.0 operating system's Active Directory service to simplify deployment within an organization, and includes quality of service (QoS) support to improve conference quality and network manageability.

What is IP Telephony? IP Telephony is an emerging set of technologies that enables voice, data, and video collaboration over existing IP-based LANs, WANs, and the Internet. Specifically , IP Telephony uses open IETF and ITU standards to move multimedia traffic over any network that uses IP (the Internet Protocol). This offers users both flexibility in physical media (for example, POTS lines, ADSL, ISDN, leased lines, coaxial cable, satellite, and twisted pair) and flexibility of physical location. As a result, the same ubiquitous networks that carry Web, e-mail and data traffic can be used to connect to individuals, businesses, schools and governments worldwide. TAPI 3.0 is an evolutionary API that supports convergence of both traditional PSTN telephony and telephony over IP networks.

What are the Benefits of IP Telephony? IP Telephony allows organizations and individuals to lower the costs of existing services, such as voice and broadcast video, while at the same time broadening their means of communication to include modern video conferencing, application sharing, and whiteboarding tools. In the past, organizations have deployed separate networks to handle traditional voice, data, and video traffic. Each with different transport requirements, these networks were expensive to install, maintain, and reconfigure. Furthermore, since these networks were physically distinct, integration was difficult if not impossible , limiting their potential usefulness . IP Telephony blends voice, video and data by specifying a common transport, IP, for each, effectively collapsing three networks into one. The result is increased manageability, lower support costs, a new breed of collaboration tools, and increased productivity. Possible applications for IP Telephony include telecommuting , real-time document collaboration, distance learning, employee training, video conferencing, video mail, and video on demand.

What is TAPI 3.0? As telephony and call control become more common in the desktop computer, a general telephony interface is needed to enable applications to access all the telephony options available on any machine. Additionally, it is imperative that the media or data on a call is available to applications in a standard manner. TAPI 3.0 is an architecture that provides simple and generic methods for making connections between two or more machines, and accessing any media streams involved in that connection. It abstracts call-control functionality to allow different, and seemingly incompatible, communication protocols to expose a common interface to applications. IP Telephony is a demand poised for explosive growth, as organizations begin an historic shift from expensive and inflexible circuit-switched public telephone networks to intelligent , flexible and inexpensive IP networks.

Inside TAPI 3.0: TAPI 3.0 integrates multimedia stream control with legacy telephony. Additionally, it is an evolution of the TAPI 2.1 API to the COM model, allowing TAPI applications to be written in any language, such as JavaT, C/C++ and the Microsoft Visual Basic programming system. Besides supporting classic telephony providers, TAPI 3.0 supports standard H.323 conferencing and IP multicast conferencing. TAPI 3.0 utilizes the Windows NT 5.0 Active Directory service to simplify deployment within an organization, and it supports quality of service (QoS) features to improve conference quality and network manageability.

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There are four major components to TAPI 3.0:

  1. TAPI 3.0 COM API

  2. TAPI Server

  3. Telephony Service Providers

  4. Media Stream Providers

In contrast to TAPI 2.1, the TAPI 3.0 API is implemented as a suite of Component Object Model (COM) objects. Moving TAPI to the object-oriented COM model allows component upgrades of TAPI features. It also allows developers to write TAPI-enabled applications in any language, such as Java, Visual Basic, or C/C++. The TAPI Server process (TAPISRV.EXE) abstracts the TSPI (TAPI Service Provider Interface) from TAPI 3.0 and TAPI 2.1, allowing TAPI 2.1 Telephony Service Providers to be used with TAPI 3.0, maintaining the internal state of TAPI. Telephony Service Providers (TSPs) are responsible for resolving the protocol-independent call model of TAPI into protocol-specific call control mechanisms. TAPI 3.0 provides backward compatibility with TAPI 2.1 TSPs. Two IP Telephony service providers (and their associated MSPs) ship by default with TAPI 3.0: the H.323 TSP and the IP Multicast Conferencing TSP. TAPI 3.0 provides a uniform way to access the media streams in a call, supporting the DirectShow API as the primary media stream handler. TAPI Media Stream Providers (MSPs) implement DirectShow interfaces for a particular TSP and are required for any telephony service that makes use of DirectShow streaming. Generic streams are handled by the application.

There are five objects in the TAPI 3.0 API:

  1. TAPI

  2. Address

  3. Terminal

  4. Call

  5. CallHub

The TAPI object is the application's entry point to TAPI 3.0. This object represents all telephony resources to which the local computer has access, allowing an application to enumerate all local and remote addresses. An Address object represents the origination or destination point for a call. Address capabilities, such as media and terminal support, can be retrieved from this object. An application can wait for a call on an Address object, or can create an outgoing call object from an Address object. A Terminal object represents the sink, or renderer, at the termination or origination point of a connection. The Terminal object can map to hardware used for human interaction, such as a telephone or microphone, but can also be a file or any other device capable of receiving input or creating output. The Call object represents an address' connection between the local address and one or more other addresses (This connection can be made directly or through a CallHub). The Call object can be imagined as a first-party view of a telephone call. All call control is done through the Call object. There is a call object for each member of a CallHub. The CallHub object represents a set of related calls. A CallHub object cannot be created directly by an application. They are created indirectly when incoming calls are received through TAPI 3.0. Using a CallHub object, a user can enumerate the other participants in a call or conference, and possibly (because of the location independent nature of COM) perform call control on the remote Call objects associated with those users, subject to sufficient permissions:

Using TAPI Objects to Place a Call:

  1. Create and initialize a TAPI object.

  2. Use the TAPI object to enumerate all available Address objects on a machine (for example, network cards, modems, and ISDN lines).

  3. Enumerate the supported address types of each Address object (for example, a phone number, IP address, and so on).

  4. Choose an Address object based on queries for support for appropriate media (audio, video, and so on) and address types.

  5. Use the CreateCall() method of the Address object to create a Call object associated with a particular address.

  6. Select appropriate Terminals on the Call object.

  7. Call the Connect() method of the Call object to place the call. To Answer a Call:

  8. Create and initialize a TAPI object.

  9. Use the TAPI object to enumerate all available Address objects on a machine (for example, network cards, modems, and ISDN lines).

  10. Enumerate the supported address types of each Address object (e.g. a phone number, IP address, etc.).

  11. Choose an Address object based on queries for support of appropriate media (audio, video, and so on) and address types.

  12. Register an interest in specific media types with the appropriate Address object.

  13. Register a call event handler (i.e. implement an ITCallNotification interface) with the Address object.

  14. TAPI notifies the application of a new call through ITCallNotification and creates a Call object.

  15. Select appropriate Terminals on the Call object.

  16. Call the Connect() method of the Call object to place the call.

  17. Call the Answer() method of the Call object to answer the call. Media Streaming Model: The Windows operating system provides an extensible framework for efficient control and manipulation of streaming media called the DirectShow API. DirectShow, through its exposed COM interfaces, provides TAPI 3.0 with unified stream control. At the heart of the DirectShow services is a modular system of pluggable components called filters, arranged in a configuration called a filter graph. A component called the filter graph manager oversees the connection of these filters and controls the stream's data flow. Each filter's capabilities are described by a number of special COM interfaces called pins. Each pin instance can consume or produce streaming data, such as digital audio. While COM objects are usually exposed in user mode programs, the DirectShow streaming architecture includes an extension to the Windows driver model that allows the connection of media streams directly at the device driver level.

These high-performance streaming extensions to the Windows driver model avoid user- to-kernel mode transitions, and allow efficient routing of data streams between different hardware components at the device driver level. Each kernel mode filter is mirrored by a corresponding user mode proxy that facilitates connection setup and can be used to control hardware-specific features. DirectShow network filters extend the streaming architecture to machines connected on an IP network. The Real-Time Transport protocol (RTP), designed to carry real-time data over connectionless networks, transports TAPI media streams and provides appropriate time stamp information. TAPI 3.0 includes a kernel mode RTP network filter. TAPI 3.0 utilizes this technology to present a unified access method for the media streams in multimedia calls. Applications can route these streams by manipulating corresponding filter graphs; they can also easily connect streams from multiple calls for bridging and conferencing capabilities.

What is H.323?: H.323 is a comprehensive International Telecommunications Union (ITU) standard for multimedia communications (voice, video, and data) over connectionless networks that do not provide a guaranteed quality of service, such as IP-based networks and the Internet. It provides for call control, multimedia management, and bandwidth management for point-to-point and multipoint conferences. H.323 mandates support for standard audio and video codecs and supports data sharing via the T.120 standard. Furthermore, the H.323 standard is network, platform, and application independent, allowing any H.323 compliant terminal to interoperate with any other.

H.323 allows multimedia streaming over current packet-switched networks. To counter the effects of LAN latency, H.323 uses as a transport the Real-time Transport Protocol (RTP), an IETF standard designed to handle the requirements of streaming real-time audio and video over the Internet. The H.323 standard specifies three command and control protocols:

  1. H.245 for call control

  2. Q.931 for call signaling

  3. The RAS (Registration, Admissions, and Status) signaling function The H.245 control channel is responsible for control messages governing operation of the H.323 terminal, including capability exchanges, commands, and indications . Q.931 is used to set up a connection between two terminals, while RAS governs registration, admission, and bandwidth functions between endpoints and gatekeepers (RAS is not used if a gatekeeper is not present).

H.323 defines four major components for an H.323-based communications system:

  1. Terminals

  2. Gateways

  3. Gatekeepers

  4. Multipoint Control Units (MCUs) Terminals are the client endpoints on the network. All terminals must support voice communications; video and data support is optional.

A Gateway is an optional element in an H.323 conference. Gateways bridge H.323 conferences to other networks, communications protocols, and multimedia formats. Gateways are not required if connections to other networks or non-H.323 compliant terminals are not needed. Gatekeepers perform two important functions which help maintain the robustness of the network - address translation and bandwidth management.

Gatekeepers map LAN aliases to IP addresses and provide address lookups when needed. Gatekeepers also exercise call control functions to limit the number of H.323 connections, and the total bandwidth used by these connections, in an H.323 "zone." A Gatekeeper is not required in an H.323 system-however, if a Gatekeeper is present, terminals must make use of its services.

Multipoint Control Units (MCU) support conferences between three or more endpoints. An MCU consists of a required Multipoint Controller (MC) and zero or more Multipoint Processors (MPs). The MC performs H.245 negotiations between all terminals to determine common audio and video processing capabilities, while the Multipoint Processor (MP) routes audio, video, and data streams between terminal endpoints.

Any H.323 client is guaranteed to support the following standards: H.261 and G.711.

H.261 is an ITU-standard video codec designed to transmit compressed video at a rate of 64 Kbps and at a resolution of 176x44 pixels (QCIF). G.711 is an ITU-standard audio codec designed to transmit A-law and -law PCM audio at bit rates of 48, 56, and 64 Kbps. Optionally, an H.323 client may support additional codecs: H.263 and G.723.

H.263 is an ITU-standard video codec based on and compatible with H.261. It offers improved compression over H.261 and transmits video at a resolution of 176 x 44 pixels (QCIF). G.723 is an ITU-standard audio codec designed to operate at very low bit rates.

The H.323 Telephony Service Provider (along with its associated Media Stream Provider) allows TAPI-enabled applications to engage in multimedia sessions with any H.323-compliant terminal on the local area network. Specifically, the H.323 Telephony Service Provider (TSP) implements the H.323 signaling stack. The TSP accepts a number of different address formats, including name, machine name , and e-mail address. The H.323 MSP is responsible for constructing the DirectShow filter graph for an H.323 connection (including the RTP, RTP payload handler, codec, sink, and renderer filters).

H.323 telephony is complicated by the reality that a user's network address (in this case, a user's IP address) is highly volatile and cannot be counted on to remain unchanged between H.323 sessions. The TAPI H.323 TSP uses the services of the Windows NT Active Directory to perform user-to-IP address resolution. Specifically, user-to-IP mapping information is stored and continually refreshed using the Internet Locator Service (ILS) Dynamic Directory, a real-time server component of the Active Directory.

The following user scenario illustrates IP address resolution in the H.323 TSP:

  1. John wishes to initiate an H.323 conference with Alice, another user on the LAN. Once Alice's video conferencing application creates an Address object and puts it in listen mode, Alice's IP address is added to the Windows NT Active Directory by the H.323 TSP. This information has a finite time to live (TTL) and is refreshed at regular intervals via the Lightweight Directory Access Protocol (LDAP):

  2. John's H.323 TSP then queries the ILS Dynamic Directory server for Alice's IP address. Specifically, John queries for any and all RTPerson objects in the Directory associated with Alice:

  3. Armed with Alice's up-to-date IP address, John initiates an H.323 call to Alice's machine, and H.323-standard negotiations and media selection occurs between the peer TSPs on both machines. Once capability negotiations have been completed, both H.323 Media Stream Providers (MSPs) construct appropriate DirectShow filter graphs, and all media streams are passed off to DirectShow to handle. The conference then begins.

What is IP Multicast Conferencing? IP Multicast is an extension to IP that allows for efficient group communication. IP Multicast arose out of the need for a lightweight, scalable conferencing solution that solved the problems associated with real-time traffic over a datagram, "best-effort" network. There are many advantages to using IP Multicast: scalability, fault tolerance, robustness, and ease of setup. The IP Multicast conferencing model incorporates the following key features:

  1. No global coordination is needed to add and remove members from a conference.

  2. To reach a multicast group, a user sends data to a single multicast IP address. No knowledge of the other users in a group is necessary.

  3. To receive data, users register their interest in a particular multicast IP address with a multicast aware router. No knowledge of the other users in a group is necessary. Routers hide the multicast implementation details from the user.

Traditional connection-oriented conferencing suffers from a number of problems:

  1. User complexity: Users must know the location of every user they wish to converse with, limiting scalability and fault-tolerance and rendering it difficult for users to add and remove themselves from a conference.

  2. Wasted bandwidth: A user wishing to broadcast data to N users must send data through N connections.

The total bandwidth required for multiparty conferences in which all users are sending data goes up as N squared the number of parties involved, leading to huge scalability problems. IP Multicast takes advantage of the actual network topology to eliminate the transmission of redundant data down the same communications links.

IP Multicast implements a lightweight, session-based communications model, which places relatively little burden on conference users. Using IP Multicast, users send only one copy of their information to a group IP address that reaches all recipients. IP Multicast is designed to scale well as the number of participants expands. Adding one more user does not add a corresponding amount of bandwidth. Multicasting also results in a greatly reduced load on the sending server. IP Multicast routes these one-to-many data streams efficiently by constructing a spanning tree, in which there is only one path from one router to any other. Copies of the stream are made only when paths diverge.

Without multicasting, the same information must either be carried over the network multiple times, one time for each recipient, or broadcast to everyone on the network, consuming unnecessary bandwidth and processing. IP Multicast uses Class D Internet Protocol addresses to specify multicast host groups, ranging from to Both permanent and temporary group addresses are supported. Permanent addresses are assigned by the Internet Assigned Numbers Authority (IANA) and include, the "all-hosts group" used to address all multicast hosts on the local network, and, which addresses all routers on a LAN. The range of addresses between and is reserved for routing and other low-level network protocols. Other addresses and ranges have been reserved for applications, such as to for Net News (for more information, see RFC 1700, "Assigned Numbers " at ftp://ftp.internic.net/rfc/rfc1700.txt).

The transport protocol for IP Multicast is RTP (Real-time Transport Protocol), which provides a standard multimedia header giving timestamp, sequence numbering, and payload format information. Applications for IP Multicast include video and audio conferencing, telecommuting, database and Web-site replication, distance learning, dissemination of stock quotes, and collaborative computing. At present, the largest demonstration of the capabilities of IP Multicast is the Internet MBONE (Multicast Backbone). The MBONE is an experimental, global multicast network layered on top of the physical Internet. It has been in existence for about five years , and presently carries IETF meetings, NASA space shuttle launches, music, concerts, and many other live meetings and performances . www. mbone .com.

The IP Multicast Conferencing TSP is chiefly responsible for resolving conference names to IP multicast addresses, using the Session Description Protocol (SDP) conference descriptors stored in the ILS Dynamic Directory Conference Server. It is complemented by the Rendezvous conference controls, described later. The IP Multicast Conferencing MSP is responsible for constructing an appropriate DirectShow filter graph for an IP multicast connection (including RTP, RTP payload handler, codec, sink, and renderer filters).

TAPI 3.0 uses the IETF standard Session Description Protocol to advertise IP multicast conferences across the enterprise. SDP descriptors are stored in the Windows NT Active Directory ” specifically, in the ILS Dynamic Directory Conference Server. In contrast to the Dynamic Directory servers used by the H.323 TSP, there is only one ILS Conference Server per enterprise, since conference announcements are not continually refreshed, therefore consuming little bandwidth.

TAPI 3.0's IP multicast conference mechanism is illustrated in the following scenario, in which John wishes to initiate a multicast conference:

  1. John's TAPI 3.0-enabled application uses the Rendezvous Controls to create an SDP session descriptor on the ILS Conference Server. The SDP descriptor contains, among other things, the conference name, start and end time information, the IP multicast address of the conference, and the media types used for the conference.

  2. Jim queries the ILS Conference Server for SDP descriptors of conferences matching his criteria.

  3. Mary and Alice perform similar queries and use the SDP information they receive to decide to participate in John's conference. Armed with the multicast IP address of the conference, they join the multicast host group.

The Rendezvous Controls are a set of COM components that abstract the concept of a conference directory, providing a mechanism to advertise new multicast conferences and to discover existing ones. They provide a common schema (SDP) for conference announcement, as well as scriptable interfaces, authentication, encryption, and access control features.

The user may add, delete, and enumerate multicast conferences stored on an ILS Conference Server via the Rendezvous Controls. These controls manipulate conference data via the Lightweight Directory Access Protocol (LDAP). The conferencing application uses the Rendezvous Controls to obtain session descriptors for the conferences that match the user's criteria. Access control lists (ACLs) protect each of the stored conference announcements, and whether or not an announcement is visible and accessible depends upon the user's credentials. Once the user has chosen a conference, the user application searches for all Address objects that support the address type "Multicast Conference Name." The application then uses the conference name from the SDP descriptor as a parameter to the CreateCall() method of the appropriate Address object, passes the appropriate Terminal objects to the returned Call object, and calls Call->Connect(). The Rendezvous Controls store the conference information on an ILS Conference Server in a format defined by the Session Description Protocol (SDP), an IETF standard for announcing multimedia conferences. The purpose of SDP is to publicize sufficient information about a conference (time, media, and location information) to allow prospective users to participate if they so choose. Originally designed to operate over the Internet MBONE (IP Multicast Backbone), SDP has been integrated by TAPI 3.0 with the Windows NT Active Directory, thereby extending its functionality to local area networks. An SDP descriptor advertises the following information about a conference.

A session description is broken into three main parts : a single Session Description, zero or more Time Descriptions, and zero or more Media Descriptions. The Session Description contains global attributes that apply to the whole conference or all media streams. Time Descriptions contain conference start, stop, and repeat time information, while Media Descriptions contain details that are specific to a particular media stream.

While traditional IP multicast conferences operating over the MBONE have advertised conferences using a push model based on the Session Announcement Protocol (SAP), TAPI 3.0 utilizes a pull-based approach using Windows NT Active Directory services. This approach offers numerous advantages, among them bandwidth conservation and ease of administration.

TAPI 3.0's conference security system addresses the following needs:

  • Controlling who can create, delete, and view conference announcements.

  • Preventing conference eavesdropping. TAPI 3.0 utilizes the security features of the Windows NT Active Directory and LDAP to provide for secure conferencing over insecure networks such as the Internet. Each object in the Active Directory can be associated with an Access Control List (ACL) specifying object access rights on a user or group basis. By associating ACLs with SDP conference descriptors, conference creators can specify who can enumerate and view conference announcements. User authentication is provided by the Windows NT security subsystem.

Session Descriptors are transmitted from the ILS Conference Server to the user over LDAP in encrypted form, via a Secure Sockets Layer (SSL) connection, ensuring that the SDP is safe from eavesdroppers. IP Multicast makes no provision for authenticating users. Any user may anonymously join a multicast host group. To keep conferences private, TAPI 3.0 allows an IP multicast conference to be encrypted, with the encryption key distributed from within the conference descriptor. Only users with sufficient permissions have access to a conference's SDP descriptor, and therefore the Multicast Encryption Key. Once an authenticated user fetches the encryption key, he or she can participate in the conference.

In contrast to traditional data traffic, multimedia streams, such as those used in IP Telephony or videoconferencing, may be extremely bandwidth and delay sensitive, imposing unique quality of service (QoS) demands on the underlying networks that carry them. Unfortunately, IP, with a connectionless, "best-effort" delivery model, does not guarantee delivery of packets in order, in a timely manner, or at all. In order to deploy real-time applications over IP networks with an acceptable level of quality, certain bandwidth, latency, and jitter requirements must be guaranteed, and must be met in a fashion that allows multimedia traffic to coexist with traditional data traffic on the same network.

Bandwidth: Multimedia data, and in particular video, may require orders of more bandwidth than traditional networks have been provisioned to handle. An uncompressed NTSC video stream, for example, can require upwards of 220 megabits per second to transmit. Even compressed, a handful of multimedia streams can completely overwhelm any other traffic on the network.

Latency: The amount of time a multimedia packet takes to get from the source to the destination (latency) has a major impact on the perceived quality of the call. There are many contributors towards latency, including transmission delays, queuing delays in network equipment, and delays in host protocol stacks. Latency must be minimized in order to maintain a certain level of interactivity and to avoid unnatural pauses in conversation.

Jitter: In contrast to data traffic, real-time multimedia packets must arrive in order and on time to be of any use to the receiver. Variations in packet arrival time (jitter) must be below a certain threshold to avoid dropped packets (and therefore irritating shrieks and gaps in the call). Jitter, by determining receive buffer sizes, also affects latency.

Coexistence: In comparison with multimedia traffic, data traffic is relatively bursty , and arrives in unpredictable chunks (for instance, when someone opens a Web page, or downloads a file from an FTP site). Aggregations of such bursts can clog routers and cause gaps in multimedia conferences, leaving calls at the mercy of everyone on the network (including other IP Telephony users). Multimedia bandwidth must be protected from data traffic, and vice versa.

Public-switched telephone networks guarantee a minimum quality of service by allocating static circuits for every telephone call. Such an approach is simple to implement, but wastes bandwidth, lacks robustness, and makes voice, video, and data integration difficult. Furthermore, circuit-switched data paths are impossible to create using a connectionless network such as IP.

QoS support on IP networks offers the following benefits:

  1. Support for real-time multimedia applications.

  2. Assurance of timely transfers of large amounts of data.

  3. The ability to share the network in a manner that avoids starving applications of bandwidth.

Quality of service in TAPI 3.0 is handled through the DirectShow RTP filter, which negotiates bandwidth capabilities with the network based on the requirements of the DirectShow codecs associated with a particular media stream. These requirements are indicated to the RTP filter by the codecs via its own QoS interface. The RTP filter then uses the COM Winsock2 GQoS interfaces to indicate , in an abstract form, its QoS requirements to the Winsock2 QoS service provider (QoS SP). The QoS SP, in turn , invokes a number of varying QoS mechanisms appropriate for the application, the underlying media, and the network, in order to guarantee appropriate end-to-end QoS. These mechanisms include:

  1. The Resource Reservation Protocol (RSVP)

  2. Local Traffic Control: Packet Scheduling; 802.1p; Appropriate layer 2 signaling mechanisms

  3. IP Type of Service and DTR header settings The Resource Reservation Protocol (RSVP) is an IETF standard designed to support resource (for example, bandwidth) reservations through networks of varying topologies and media. Through RSVP, a user's quality of service requests are propagated to all routers along the data path, allowing the network to reconfigure itself (at all network levels) to meet the desired level of service. The RSVP protocol engages network resources by establishing flows throughout the network. A flow is a network path associated with one or more senders, one or more receivers, and a certain quality of service. A sending host wishing to send data that requires a certain QoS will broadcast, via an RSVP-enabled Winsock Service Provider, "path" messages toward the intended recipients. These path messages, which describe the bandwidth requirements and relevant parameters of the data to be sent, are propagated to all intermediate routers along the path. A receiving host, interested in this particular data, will confirm the flow (and the network path) by sending "reserve" messages through the network, describing the bandwidth characteristics of data it wishes to receive from the sender. As these reserve messages propagate back toward the sender, intermediate routers, based on bandwidth capacity, decide whether or not to accept the proposed reservation and commit resources. If an affirmative decision is made, the resources are committed and reserve messages are propagated to the next hop on the path from source to destination.

Packet Scheduling: This mechanism can be used in conjunction with RSVP (if the underlying network is RSVP-enabled) or without RSVP. Traffic is identified as belonging to one flow or another, and packets from each flow are scheduled in accordance with the traffic control parameters for the flow. These parameters generally include a scheduled rate (token bucket parameter) and some indication of priority. The former is used to pace the transmission of packets to the network. The latter is used to determine the order in which packets should be submitted to the network when congestion occurs.

801.2p: Traffic control can also be used to determine the 802.1 User Priority value (a MAC header field used to indicate relative packet priority) to be associated with each transmitted packet. 802.1p-enabled switches can then give preferential treatment to certain packets over others, providing additional quality of service support at the data link layer level.

Layer 2 Signaling Mechanisms: In response to Winsock 2 QoS APIs, the QoS service provider may invoke additional traffic control mechanisms depending on the specific underlying data link layer. It may signal an underlying ATM network, for instance, to set up an appropriate virtual circuit for each flow. When the underlying media is a traditional 802 shared media network, the QoS service provider may extend the standard RSVP mechanism to signal a Subnet Bandwidth Manager (SBM). The SBM provides centralized bandwidth management on shared networks.

Each IP packet contains a three-bit Precedence field, which indicates the priority of the packet. An additional field can be used to indicate a delay, throughput, or reliability preference to the network. Local traffic control can be used to set these bits in the IP headers of packets on particular flows. As a result, packets belonging to a flow will be treated appropriately later by three devices on the network. These fields are analogous to 802.1p priority settings but are interpreted by higher layer network devices. www.microsoft.com/communications. See H.323.


Tenant Package.


TAR is the UNIX standard program for combining and compressing files. It's like a UNIX Winzip program. I believe it stands for "Tape Archiver". The UNIX System V manual lists the command description as "tape file archiver". One no longer uses it just for tape archiving, but that was its initial usage.


TARFU is a military term used on TTY circuits to state that "things are really F...ked up". In the military there were three stages to a project, which is not going well ” SNAFU, FUBAR and TARFU. See SNAFU and FUBAR.


Truevision Advanced Raster Graphics Adapter.


A SCSI device that performs an operation requested by an initiator.

Target Host Number

The number that identifies the destination software program during the user logon process.

Target ID Address Resolution Protocol.

TARP is the established multivendor standard for SONET network elements that support TL1 Operating System interfaces.

Target Token Rotation Time

TTRT. An FDDI (Fiber Distributed Data Interface)token travels along the network ring from node to node. If a node does not need to transmit data, it picks up the token and sends it to the next node. If the node possessing the token does need to transmit, it can send as many frames as desired for a fixed amount of time.


Documents filed by a regulated telephone company with a state public utility commission or the Federal Communications Commission. The tariff, a public document, details services, equipment and pricing offered by the telephone company (a common carrier) to all potential customers. Being a "common carrier" means it (the phone company) must offer its services to everybody at the prices and at the conditions outlined in its public tariffs. Tariffs do not carry the weight of law behind them. If you or the telephone company violate them, no one will go to jail. The worst that can happen to you, as a subscriber, is that your service will be cut off, or threatened to be cut off. Regulatory authorities do not normally approve tariffs. They accept them ” until they are successfully challenged before a hearing of the regulatory body or in court (usually Federal Court). Many tariffs were accepted by regulatory commissions only to be struck down in court as unlawful , discriminatory, not cost-justified, etc. Monies collected under the tariff have been refunded and unnecessary equipment removed. In these new, competitive days, many telephone companies are violating their own tariffs by charging less money than their tariffs say they should, or bundling services together at a discount. They are also providing service and equipment on terms less onerous than outlined in their tariffs. Many users now regard tariffs as starting bargaining points, rather than ending bargaining points.

Tariff 12

A user-specific long distance tariff of AT&T. Tariff 12 gives AT&T the ability to price its long distance services for one company practically any which way it feels ” giving them a mix of services at stable prices over the long term with significant volume discounts . As this dictionary was going to the printer, a federal appeals court overturned the Federal Communications Commission's April 1989 decision allowing AT&T to offer custom networks and ordered the FCC to reopen its investigation into the legality of the Tariff 12 deals. There are still some users. They are "grandfathered" until we get a final say on the tariff. And, as we go to press, AT&T can offer Tariff 12 customized services to any company ” but cannot include 800 services in its Tariff 12 pricing.

Tariff 15

A user-specific long distance tariff of AT&T. Tariff 15 gives AT&T the ability to price its long distance services for one company practically any way it feels. Tariff 15 is single-customer discounting. Some of AT&T competitors claim the tariff is "illegal."

Tariff Rebalancing

Largely an initiative of the FCC in the US, national, regional (e.g., EU) and international (ITU) regulatory authorities and policy-making bodies are considering the rebalancing of tariffs in order that they be more closely related to the costs of providing the various telecommunications products and services. While most attention is focused on the accountings rate for long distance, both domestic and international, tariff rebalancing encompasses all tariffed products (including equipment rentals) and services (e.g., local service), and across both the business and consumer domains. Historically, tariffs for individual products and services were designed in the context of the overall tariff structure, which addressed the full range of such products and services. Within this overall structure existed a complex set of cross-subsidies which generally resulted in low tariffs for basic consumer services, such as residential local service. Relatively high tariffs existed both for optional services, such as long distance, international and custom calling features. The primary justification for this arrangement was that of the desire to gain "universal service" - a phone in everybody's home. Mixed in with this was the concept of "ability to pay." Further, individual consumers vote, while companies do not. As consumer rights advocates became more vocal during the past twenty years or so, the pressure to retain these cross- subsidies increased. It generally is recognized, however, that the traditional tariff structure places incumbent carriers and service providers at a decided disadvantage in a competitive environment, as the pricing policies of the newer competitors are not constrained by tariffs, nor strange societal concepts of universal service and ability to pay. Further ” and this is perhaps a major impetus ” there exists a clear imbalance in the accounting rates for international long distance calls. For instance, a call from Argentina to the U.S. is much more expensive than is a call in the reverse direction, even though the costs for call origination are roughly equal to the costs of termination. This imbalance in international long distance tariffs has led to a huge imbalance of trade, to the detriment of the US. Therefore, both domestic US deregulation, and the resulting increase in competition, and the international imbalance of trade resulting from imbalance in international long distance tariffs have led to the FCC initiative. Many other nations and regions have followed the U.S. lead in terms of deregulation and competition, and are in the process of rebalancing domestic tariffs. Although the WTO (World Trade Organization) has taken the initiative in terms of the rebalancing of international long distance tariffs, many nations are resisting, citing their opinions that the WTO is unduly influenced by the U.S. Further, many developing nations rely heavily on this imbalance of trade as a major source of hard currency. See also Cross Subsidization, Separations and Settlements, Tariff and Universal Service.


Telephone Answering and Recording Machines.


Target ID Address Resolution Protocol. TARP is the established multivendor standard for SONET network elements that support TL1 Operating System interfaces.


Telephone Answering Service.


Telecommunications Alarm, Surveillance, and Control system. Expands the scope of maintenance from the traditional alarm monitoring and control functions to include performance monitoring and fault locating.


Time Assignment Speech Interpolation. A money saving analog multiplexing procedure which keeps the connection to the circuit as long as someone is speaking and lets other conversations use the circuit during the intervals (measured in microseconds) when there's no speaking. Since a long distance circuit is usually only half used ” one person speaking, one person listening ” at least 50% of the circuit can be used by someone else. TASI is typically used by long distance companies on submarine cable across the Atlantic and the Pacific. Unfortunately, the flip-flopping around of circuit allocation by TASI means that the first tiny bit of a conversation is often lost. This can be disastrous for data. The key in data is to keep transmitting. The key in voice is to say something like "Ah" to seize the channel and then say what you want. TASI is very much like a very fast version of mobile dispatch radio. A modern version of TASI is called Digital Speech Interpolation (DSI). TASI is somewhat comparable to statistical multiplexing of data. Ray Horak, my Contributing Editor, recently had a gentleman from Cable & Wireless attending one of his public seminars . Apparently this gentleman was one of the engineers working on an early submarine cable between the US and the UK. The cable system used TASI, which was by no means perfect, but it certainly enhanced the voice traffic carrying capacity of that very limited 50- pair cable system. The system worked fine most of the time, but occasionally crashed at no particular time and for no apparent reason. The engineers were baffled. Finally, they recorded the conversations until the system crashed. It seems that the TASI approach would fail when required to support a large number of conversations between women. It seems that women have a tendency to talk and listen at the same time, and to do so quite effectively. Men, it seems, either lack this skill or prefer the "I talk and you listen, then you talk and I listen" protocol. Mystery solved! (Ladies, please don't call me to complain about this definition. Ray swears that it is true. Call him.)

Task Management

Allocating resources and overseeing the sequence of tasks completed by the computer.

Task Switching

You have a computer and you want to have it do several tasks at once. There are two ways. One is multi tasking. The computer will keep working on several tasks at once, though you may not see them on your screen. For example, you pay start a spreadsheet recalculating. And then you may call your electronic mail system. While you're receiving your mail, your spreadsheet is still recalculating. When you're through, you can switch back to your spreadsheet and see the final results. That's called multi tasking and MS-DOS doesn't have it. MS-DOS 5.0 through its Shell has something called task switching, whose idea is that you can load several programs into your computer and switch quickly between them. But the programs you put in background won't run. They stop the moment you put them in background. When you cycle back to them, they will start running again. Sadly, MS-DOS 5.0 Shell's task switching capabilities are very weak. Your programs will lock up and you will lose your data. I do not recommend using Task Switching in MS-DOS 5.0. Windows 3.1 is alleged to have a form of multi tasking. I don't trust it either yet. Windows has locked up on me on several occasions.

Task-To-Task Communication

The process whereby one computer program exchanges data with another. May also be called program-to-program communications.

Tauzin-Dingell Bill

A House of Representatives bill circa 2002 pushing to allow the Regional Bell Operating Companies (RBOCs) to offer broadband Internet services over long-distance lines without opening up their local phone service monopolies to outside competition. The Tauzin-Dingell bill, or more formally H.R. 1542, the Internet Freedom and Broadband Deployment Act, is legislation written by Representatives John D. Dingell (DMA) and Billy Tauzin (R-LA).

Tax Identification Number

A unique identifier for business organizations that is used for reporting tax payments to the government (similar to the social security number for individuals).


Transparent Asynchronous Transmitter Receiver Interface. 100-Mbps ATM physical interface specification based on the FDDI PHY.


Twain noticed one day that "taxidermist" was listed just prior to "taxman" in the dictionary. He concluded that he was always much happier to see the taxi- dermist coming that the taxman...because he knew the taxidermist was only going to take the skin.


The orderly classification of things according to their natural relationships. The Internet Engineering Task Force (IETF) uses this term a lot in their documents to describe the orderly interrelationships between protocols used in the Internet.


TransAtlantic Telephone cable. A bunch of phone companies pool their monies and build an undersea cable connecting their various countries . They typically own the percentage of the cable which they contributed in money. In August, 2001, TAT-14 was completed. It joined Denmark, France, Germany, the Netherlands, the U.K. and the U.S. at a cost of $1.4 billion. There were 48 participating telecommunications carriers. Capacity on the cable was 640 giabits per second. That's 640,000,000,000 bits per second.


An ATM term. Transparent Bridging: An IETF bridging standard where bridge behavior is transparent to the data traffic. To avoid ambiguous routes or loops , a Spanning Tree algorithm is used.


TeraBits per Second. See also Tera.


To Be Announced. Pricing of a product that may exist and that may, one day, be priced. You often see TBA after products which are hyperware, vaporware, mirrorware or smokeware.


  1. Transnational Broadband Backbone. An international, high-speed (T-3 or faster), backbone system of transmission facilities and, perhaps, switching systems.

  2. Telecommunications Bonding Backbone. A copper conductor extending from the telecommunications main grounding busbar (TMGB) to the farthest floor telecommunications grounding busbar (TGB). The idea is to eestablish an electrically conductive path for telecommunications, the metallic parts of which path are permanently joined to ensure 1) electrical continuity, 2) the capacity to conduct safely any current likely to be imposed, and

  3. the ability to limit dangerous electrical potentials. In other words, a TBB is a permanently hard-wired metallic cable system, which is safely grounded and surge-protected in order to protect your equipment.


TBB Interconnecting Bonding Conductor. A conductor used specifically to interconnect metallic wires in a TBB (Telecommunications Bonding Backbone). See also TBB.


To Be Determined.


Transient Buffer Exposure: This is a negotiated number of cells that the network would like to limit the source to sending during startup periods, before the first RM- cell returns.


Transport Bandwidth Manager. A TBM consists of a OC-3 or OC-12 terminal shelf. It is an optical fibre term.


Telemetry Byte Oriented Serial protocol. TBOS is a protocol for transmitting alarm, status, and control points between NE and OS. TBOS defines one physical interface for direct connection between telemetry remote and the monitored equipment. This is a point- to-point communication, RS 422A modified four wire, two to Tx (remote to monitored) and two to Tx (monitored to remote), 26 gauge, max 4 kft remote-to-monitored. Remote sees a 100 to 180 ohms resistor at monitored terminal.


Tera bit per second. One million million bits per second. See Tera.


Timed BReak.


  1. Tc (Time committed). Also known as Measurement Interval. A Frame Relay term defining the interval of time which the carrier uses to measure data rates that burst above the CIR (Committed Information Rate). See also Committed Information Rate.

  2. Transmission Convergence. The TC sublayer is a dimension of the ATM Physical Layer (PHY), working in tight formation with the Physical Medium (PM) sublayer. The TC accepts frames of data transmitted across the PM, delivering them to the ATM Layer for segmentation into cells, generates the Header Error Check (HEC), and sends idle cells when the ATM layer has none to send. On reception , the TC sublayer delineates individual cells in the received bit stream and reconstitutes the frames of data, using the HEC to detect and correct received errors.

  3. Telecommunications Closet. A closet which houses telecommunications wiring and telecom wiring equipment. It contains the BHC (Backbone to Horizontal Cross-connect). It may also contain the network demarcation , or MC (Main Cross-connect). The telecommunications closet is used to connect up telecom wiring. The closet typically has a door. It's a good idea to lock the door and not put anything else in the closet, like mops, buckets and brooms.

  4. Transmission Control.

  5. Technical Consultant.


Trellis-Coded Pulse Amplitude Modulation, a robust modulation format used in SHDSL. TC-PAM provides adaptability in bit rate and line reach. With SHDSL, rates of 192 kBps can be attained on line reaches of more than 20,000 feet (6 km), and higher rates, up to 2.3 MBps can be attained over line reaches of more than 10,000 feet (3 km). SHDSL is spectrally compatible with ADSL. See ADSL, PAM, SHDSL and Trellis Coding.


TeleCommunications Association. A not-for-profit users association of communications management professionals. Formed in 1961, TCA recently expanded its focus to position itself as "The Information Technology and Telecommunications Association." Most of its members are West-of-the-Rockies telecommunications management professionals. For many years, TCA held a great annual conference in San Diego. The TCA conference was huge, and it was lots of fun. The vendors spent fortunes on their exhibits. They spent even more on hospitality suites and lavish dinner parties. Over time, the show declined as the conference business became for-profit, as the technologies moved ever more quickly, as the number of conferences proliferated, as the conference business became more highly focused and more fragmented , and as companies downsized and could no longer afford to subsidize associations like TCA. The TCA show moved to Reno, Nevada in 1997. That was the last show. Several of the TCA chapters still hold annual events, but they are much smaller. www.tca.org


TeleCommunications Access Method. A popular telecommunications software package to run on IBM 370 computers. See IBM.


Telecommunications Access Method. An IBM macro language for creating communications applications programs and message control.


Transactional Capabilities Application Part. Provides the signaling function for network data bases. TCAP is an ISDN application protocol. In addition to PRA and ISUP, the third major ISDN protocol in the delivery of advanced network services is TCAP, a CCS7 application protocol that provides the platform to support non-circuit related, transaction- based information exchange between network entities. This capability is required by transaction-based services that must exchange information between a pair of signaling nodes in a CCS7 network. Examples of these services include enhanced dial-800 service, automated credit card calling and virtual private networking. The TCAP protocol enables these services to access remote databases called service control points (SCPs) to process part of the call. The SCP supplies the translation and routing data needed to deliver advanced network services ” like translating a dial calls into the required routing number. TCAP is useful also in coordinating some enhanced call-related services. For example, network ring again requires the connection of two users when both stations become idle. In this case, TCAP is used to coordinate between the users' switches while waiting for each line to become idle. And it can do this without tying up network trunks.

One of the major advantages of TCAP is that it provides a set of protocol building blocks for use in a variety of service definitions. The TCAP building blocks are subdivided into the transaction sublayer and the component sublayer. For more on TCAP, see the 1988-3 issue of Northern Telecom's Telesis publication.


T-Carrier Administration System. Provides mechanized support for the facility maintenance and administration center to achieve centralized administration and control of the digital network.


Telephone Country Code.


Telephone Company Circuit ID.


Training Check Frame. Last step in a series of signals in a fax transmission called a training sequence, designed to let the receiver adjust to telephone line conditions.


A Yiddish word meaning trinkets, best exemplified by the giveaway junk we often pick up at telecommunications trade shows. It's pronounced choch-kuh in Yiddish. The plural is pronounced choch-key. See Yiddish.


Trunk Circuit Identification Code. Only relevant to SS7. A number that uniquely identifies a trunk between an origination point code and a destination point code. An example would be between two telephone company switches.


TeleCommunications Industry Forum. A voluntary special interest group under ATIS (Alliance for Telecommunications Industry Solutions). TCIF addresses areas such as electronic commerce, including bar coding and EDI (Electronic Data Interchange). www.atis.org/atis/tcif/index.html See also ATIS.


  1. Tool Command Language. TcI is actually two things: a language and a library. First, Tcl is a simple textual language, intended primarily for issuing commands to interactive programs such as text editors, debuggers , illustrators, and shells . It has a simple syntax and is also programmable, so Tcl users can write command procedures to provide more powerful commands than those in the built-in set. Second, Tcl is a library package that can be embedded in application programs. The Tcl library consists of a parser for the Tcl language, routines to implement the Tcl built-in commands, and procedures that allow each application to extend Tcl with additional commands specific to that application. The application program generates Tcl commands and passes them to the Tcl parser for execution. Commands may be generated by reading characters from an input source, or by associating command strings with elements of the application's user interface, such as menu entries, buttons , or keystrokes. When the Tcl library receives commands it parses them into component fields and executes built-in commands directly. For commands implemented by the application, Tcl calls back to the application to execute the commands. In many cases commands will invoke recursive invocations of the Tcl interpreter by passing in additional strings to execute (procedures, looping commands, and conditional commands all work in this way). www.neosoft.com/tcl/default.html.

  2. Tool Command Language.


  1. Traveling Class Mark.

  2. Trellis Coding Modulation.

  3. Time Compression Multiplexing. A digital transmission technique that permits full duplex data transmission by sending compressed bursts of data in a "ping-pong" fashion.

  4. Telecommunications Manager. The TCM is the manager of the department that plans, controls, and administers the telephony and telecommunications assets of the company. He ensures that the telephone and telecommunications systems are well-run and functioning smoothly. These assets may include the PBX and ISDN, T-1, local and long distance telephone lines, telephone sets, authorization codes, cable pairs, WANs, Fax machines, voice mail systems, automated attendants, interactive voice response systems, automatic call distribution, multiplexors, modem pools, etc. The internal data facilities such as LANs and routers may be under the administration of the TCM, or could be the responsibility of the Management Information Systems (MIS) department. But since the TCM has responsibility for both the inside wiring and the outside Carrier facilities, close coordination would be required if the internal data facilities are controlled by the MIS department.

The following are the functions of the TCM;

  • Operating, administering, monitoring, and maintaining the existing telecommunications systems.

  • Dealing with the various vendors and providers, including verifying and paying the bills.

  • Preparing and managing the Telecommunications budget.

  • Keeping abreast of changes in technology, services, industry structure, and rates.

  • Assisting company management in developing a corporate telecommunications policy that meets business objectives.

  • Developing and implementing company telephone and telecommunications procedures for efficient and cost effective use, and training company employees in these procedures.

  • Upgrading, procurement, selecting, contracting, or purchasing a system, new system, equipment, or services.

  • Planning and analyzing for growth, new requirements, or future functionality. The goal of the TCM is to provide good telecommunications services for an organization and its employees at the lowest possible cost. This definition courtesy , Robert J. Perillo, Perillo@dockmaster.ncsc.mil.


Collective Number Group Table.


Thomas Conrad Networking System is a 100 million bit per second proprietary networking system (LAN) based on ARCnet that can use most standard ARCnet drivers on any network operating systems.


Total Cost of Ownership. A term coined by The Gartner Group to bring attention to the actual, total cost to the enterprise of owning a PC. In 1997 that figure was $29,353 for owning a standard, networked, Windows 95 PC for a period of three years. It's probably a lot higher now. The point is clear and fairly obvious ” consider not only the acquisition/implementation cost of a workstation (networked or not), but also consider the total cost, including administration, maintenance, support, software upgrades and training. Of course, one problem with TCO is that it has now way to measure the actual benefits of owning that equipment. A product can also have a low cost of ownership, but not be the best product for the job. IN short, like all statistics, you can reach pretty well whatever conclusion you would like.


  1. Transmission Control Protocol. ARPAnet-developed transport layer protocol corresponding to OSI Layer 4, the Transport Layer. TCP is a transport layer, connection-oriented, end-to-end protocol. It provides reliable, sequenced , and unduplicated delivery of bytes to a remote or local user. TCP provides reliable byte stream communication between pairs of processes in hosts attached to interconnected networks. It is the portion of the TCP/IP protocol suite that governs the exchange of sequential data. UDP (User Datagram Protocol) is a Layer 4 option that does not provide the same level of reliability, but does involve much less overhead. See Internet and TCP/IP for a much longer explanations .

  2. An ATM term. Test Coordination Procedure: A set of rules to coordinate the test process between the lower tester and the upper tester. The purpose is to enable the lower tester to control the operation of the upper tester. These procedures may, or may not, be specified in an abstract test suite.

  3. Total Call Processor.


According to Microsoft: Transmission Control Protocol/Internet Protocol (TCP/IP) is a networking protocol that provides communication across interconnected networks, between computers with diverse hardware architectures and various operating systems. TCP (Transmission Control Protocol) and IP (Internet Protocol) are only two protocols in the family of Internet protocols. Over time, however, "TCP/IP" has been used in industry to denote the family of common Internet protocols. The Internet protocols are a result of a Defense Advanced Research Projects Agency (DARPA) research project on network interconnection in the late 1970s. It was mandated on all United States defense long-haul networks in 1983 but was not widely accepted until the integration with 4.2 BSD (Berkeley Software Distribution) UNIX. The popularity of TCP/IP (Harry's note: it's the Internet's networking protocol) is based on:

  • Robust client-server framework. TCP/IP is an excellent client-server application platform, especially in wide-area network (WAN) environments.

  • Information sharing. Thousands of academic, defense, scientific, and commercial organizations share data, electronic mail and services on the connected Internet using TCP/IP.

  • General availability. Implementations of TCP/IP are available on nearly every popular computer operating system. Source code is widely available for many implementations . Additionally, bridge, router and network analyzer vendors all offer support for the TCP/IP protocol family within their products. TCP/IP is the most complete and accepted networking protocol available. Virtually all modern operating systems offer TCP/IP support, and most large networks rely on TCP/IP for all their network traffic. Microsoft TCP/IP provides cross-platform connectivity and a client-server development framework that many software vendors and corporate developers are using to develop distributed and client-server applications in heterogeneous enterprise networks over TCP/IP.

How TCP Works: TCP is a reliable, connection-oriented protocol. Connection-oriented implies that TCP first establishes a connection between the two systems that intend to exchange data. Since most networks are built on shared media (for example, several systems sharing the same cabling), it is necessary to break chunks of data into manageable pieces so that no two communicating computers monopolize the network. These pieces are called packets. When an application sends a message to TCP for transmission, TCP breaks the message into packets, sized appropriately for the network, and sends them over the network.

Because a single message is often broken into many packets, TCP marks these packets with sequence numbers before sending them. The sequence numbers allow the receiving system to properly reassemble the packets into the original message. Being able to reassemble the original message is not enough, the accuracy of the data must also be verified . TCP does this by computing a checksum. A checksum is a simple mathematical computation applied, by the sender, to the data contained in the TCP packet. The recipient then does the same calculation on the received data and compares the result with the checksum that the sender computed. If the results match, the recipient sends an acknowledgment (ACK). If the results do not match, the recipient asks the sender to resend the packet. Finally, TCP uses port IDs to specify which application running on the system is sending or receiving the data.

The port ID, checksum, and sequence number are inserted into the TCP packet in a special section called the header. The header is at the beginning of the packet containing this and other "control" information for TCP.

How IP Works: IP is the messenger protocol of TCP/IP. The IP protocol, much simpler than TCP, basically addresses and sends packets. IP relies on three pieces of information, which you provide, to receive and deliver packets successfully: IP address, subnet mask, and default gateway.

The IP address identifies your system on the TCP/IP network. IP addresses are 32-bit addresses that are globally unique on a network. They are generally represented in dotted decimal notation, which separates the four bytes of the address with periods. An IP address looks like this:

Although an IP address is a single value, it really contains two pieces of information:

(a.) Your system's network ID, and (b.) Your system's host (or system) ID.

The subnet mask, also represented in dotted decimal notation, is used to extract these two values from your IP address. The value of the subnet mask is determined by setting the network ID bits of the IP address to ones and the host ID bits to zeros. The result allows TCP/IP to determine the host and network IDs of the local workstation. Here's how to understand an IP address. For example:

When the IP address is (specified by the user)

And the subnet mask is (specified by the user)

The network ID is 102.54 (IP address and subnet mask) And the host ID is 94.97 (IP address and subnet mask)

OK. the above was Microsoft's definition. Here's my definition, which covers some areas Microsoft doesn't. TCP/IP is a set of protocols developed by the Department of Defense to link dissimilar computers across many kinds of networks, including unreliable ones and ones connected to dissimilar LANs. TCP/IP is the protocol used on the Internet. It is, in essence, the glue that binds the Internet. Developed in the 1970s by the U.S. Department of Defense's Advanced Research Projects Agency (DARPA) as a military standard protocol, its assurance of multi vendor connectivity has made it popular among commercial users as well, who have adopted TCP/IP. Consequently, TCP/IP now is supported by many manufacturers of minicomputers, personal computers, mainframes, technical workstations and data communications equipment. It is also the protocol commonly used over many Ethernet LANs (as well as X.25) networks. It has been implemented on everything from PC LANs to minis and mainframes.

TCP/IP currently divides networking functionality into only four layers: A Network Interface Layer that corresponds to the OSI Physical and Data Link Layers . This layer manages the exchange of data between a device and the network to which it is attached and routes data between devices on the same network.

An Internet Layer which corresponds to the OSI network layer. The Internet Protocol (IP) subset of the TCP/IP suite runs at this layer. IP provides the addressing needed to allow routers to forward packets across a multiple LAN inter network. In IEEE terms, it provides connectionless datagram service, which means it attempts to deliver every packet, but has no provision for retransmitting lost or damaged packets. IP leaves such error correction, if required, to higher level protocols, such as TCP.

IP addresses are 32 bits in length and have two parts: the Network Identifier (Net ID) and the Host Identifier (Host ID). Assigned by a central authority, the Net ID specifies the address, unique across the Internet, for each network or related group of networks. Assigned by the local network administrator, the Host ID specifies a particular host, station or node within a given network and need only be unique within that network.

A Transport Layer, which corresponds to the OSI Transport Layer. The Transmission Control Protocol (TCP) subset runs at this layer. TCP provides end-to-end connectivity between data source and destination with detection of, and recovery from, lost, duplicated , or corrupted packets ” thus offering the error control lacking in lower level IP routing. In TCP, message blocks from applications are divided into smaller segments, each with a sequence number that indicates the order of the segment within the block. The destination device examines the message segments and, when a complete sequence of segments is received, sends an acknowledgement (ACK) to the source, containing the number of the next byte expected at the destination.

An Application Layer, which corresponds to the session, presentation and application layers of the OSI model. This layer manages the function required by the user programs and includes protocols for remote log-in (Telnet), file transfer (FTP), and electronic mail (SMTP).

See also OSI and Public Key Encryption.


  1. Transaction Confirmation Report. A report from a fax machine listing the faxes received and transmitted. It provides details about each fax, including date, time, the remote fax's number, results, total pages.

  2. An ATM term. Tagged Cell Rate: An ABR service parameter, TCR limits the rate at which a source may send out-of-rate forward RM-cells. TCR is a constant fixed at 10 cells/second.


  1. Transmission Convergence Sublayer: This is part of the ATM physical layer that defines how cells will be transmitted by the actual physical layer.

  2. TCS is a manufacturer of excellent call center manpower scheduling software packages. It is based in Nashville, TN.


A Bluetooth term. A set of AT-commands by which a mobile phone and modem can be controlled in the multiple usage models. In BT, AT-commands are based on ITU-T recommendation v.250 and ETS 300 916(GSM 07.07). In addition, the commands used for fax services are specified by the implementation. TCS-AT will also be used for dial-up networking and headset profiles.

TCS Binary

A Bluetooth term. Bluetooth Telephony Control protocol Specification using bit-Oriented protocol. It is also referred to as the TCS-BIN system. TCS-BIN will be used for cordless telephony profiles.


  1. Timing Control Unit.

  2. Closed User Group Table.


Address Conversion Table.


Time Division - Synchronous Code Division Multiple Access. A unique 3G wireless cell phone standard being developed by China for use inside China. The spectral efficiency of radio systems is 3 to 5 times higher than that of GSM. When TD-SCDMA base- stations are connected to an existing GSM system, the existing transmission links may be reused. Network operators can offer 3G voice, video and data services without installation of completely new networks. Instead, selective insertion of TD-SCDMA network elements into existing GSM radio access network structures allows a gradual growth of 3G coverage alongside the existing cell phone infrastructure. See 3G.


See Traffic Data Administration System.


  1. Time Division Controller.

  2. Dial Code Table.


  1. Time Division Duplex. A method used in cellular and PCS networks employing TDMA (Time Division Multiple Access) to support full duplex communications. Each radio channel is divided into multiple time slots through TDMA, thereby supporting multiple conversations. TDD supports transmission in the forward direction (from the cell phone to the cell site) through one radio frequency channel and one time slot. Another radio channel and time slot supports transmission in the backward direction (from the cell site to the cell phone). TDD is flexible with regard to allocation of timeslots for forward and reverse, making it best suited for asymmetrical applications, such as web browsing. See also FDD.

  2. IPWireless's term for UMTS TD-CDMA. They claim that their TDD implementation allows a datacom wireless service provider to transmit data on a single radio channel alternating between uplink and downlinks as opposed to a traditional Frequency Division Duplex (FDD) approach of first and second generation cellular, which requires separate channels for each of the uplink and downlink channels.

  3. Telecommunications Device for the Deaf. Under the Communications Act of 1934, a TDD is defined as a machine "that employs graphic communication in the transmission of coded signals through a wire or radio." TDD devices (which typically look like simple computer terminals) use the Baudot method of communications. Most TDD devices are acoustically coupled and are slow, running at 300 baud.

There is a special TDD/TTY Operator Services number. It's 800-855-1155. Users of TDDs often abbreviate commonly used words or expressions to save time. Here are some of the most frequently used:

ANS Answer                        R Are

CUD Could                            REC Receive

GA Go Ahead                    SK Stop Keying

LTR Letter                              THRU Through

MSG Message                THX Thank You

MIN Minute                            U You

NITE Night                              UR Your

PLS Please                            WUD Would

QQ Question                        XOX Hugs & Kisses


Telephone Disclosure and Dispute Resolution Act. A US federal act passed in 1992 which required both the FCC and the FTC to prescribe regulations governing pay-per- call services. Subsequently, the FTC adopted its 900-Number Rule, which became effective November 1, 1993. Under the TDDRA, a consumer's telephone service cannot be disconnected for failure to pay charges for a 900-number call, and 900-number blocking must be made available to consumer who do not wish to have access to 900-number services. See also 900-Number Rule.


Technical Development and Evaluation Laboratory.


  1. Trunk Distributing Frame.

  2. Transborder Data Flows are movements of machine-readable data across international boundaries. TDF legislation began in the 1970s and has been put into effect by many countries in an attempt to protect personal privacy of citizens . This term has particular meaning as it relates to electronic commerce or EDI and is becoming more and more relevant with the use of the Internet as a means to conduct global business.


Time Domain Harmonic Scaling


Transmit Division Intertie.


See Time Division Multiplex .


Time Division Multiple Access. One of several technologies used to separate multiple conversation transmissions over a finite frequency allocation of through-the-air bandwidth. As with FDMA (Frequency Division Multiple Access), TDMA is used to allocate a discrete amount of frequency bandwidth to each user, in order to permit many simultaneous conversations. However, each caller is assigned a specific timeslot for transmission. A digital cellular telephone system using TDMA assigns 10 timeslots for each frequency channel, and cellular telephones send bursts, or packets, of information during each timeslot. The packets of information are reassembled by the receiving equipment into the original voice components. TDMA promises to significantly increase the efficiency of cellular telephone systems, allowing a greater number of simultaneous conversations. See CDMA, FDMA.


An abbreviated way of describing a TDMA system which can support 3 calls in single radio frequency "carrier." IS-54 TDMA divides a carrier frequency into 6 time slots. Initial "full-rate" vocoders will require 2 time slots per user and thus support 3 users on each on each carrier frequency. This is called TDMA-3. Future "half-rate" vocoders will require only 1 time slot per user and support 6 users per carrier frequency, yielding TDMA- 6.


An abbreviated way of describing a TDMA system which is capable of supporting 6 users per carrier frequency. IS-54 TDMA divides a carrier frequency into 6 time slots per user and thus support 3 users on each carrier frequency. Future "half-rate" vocoders will require only 1 time slot per user and thus will support 6 users per carrier frequency, yielding TDMA-6.


  1. Technical Document Management Systems.

  2. Time Division Multiplex System.

  3. Transmission Distortion Measuring Set.

Newton[ap]s Telecom Dictionary
Newton[ap]s Telecom Dictionary
ISBN: 979387345
Year: 2004
Pages: 133

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