 |
ANI Automatic Number Identification; A telephone function that transmits the billing number of the incoming call (Caller ID, for example). BLI Busy Lamp Indicator; A light or LED on a
telephone that shows which line is in use. CoS - Class of Service.
Class of Service (CoS) is a classification method only. CoS does NOT
ensure a level of Quality of Service (QoS), but is the method used by
queuing mechanisms to limit delay and other factors to improve QoS.
Most CoS strategies assign a priority level, usually 0–7 or 0-63, to a
frame or packet respectively. Common CoS models include the IP TOS
(Type Of Service) byte, Differentiated Services Code Point DID Direct Inward Dialing; The ability to make a telephone call directly into an internal extension without having to go through the operator. Diff-Serv Differentiated Services; The Diff-Serv model divides traffic into a small number of classes to provide quality of service (QoS). DNIS Dialed Number Identification Service; A telephone function that sends the dialed telephone number to the answering service. DTMF Dual-Tone Multifrequency; The type of audio signals generated when you press the buttons on a touch-tone telephone. 802.1p An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level. 802.1q An IEEE standard for providing virtual
LAN (VLAN)
identification and QoS levels. Three bits are used to allow eight
priority levels, and 12 bits are used to identify up to 4,096 VLANs. Gateway - A gateway is basically a protocol converter, i.e. a network point that connects networks using different protocols so that data can be exchanged seamlessly between endpoints. For example, a POTS-to-VoIP Gateway connects PSTNs and packet-switched networks, translating the media into IP packets, so that "legacy" telephony becomes Voice-over-IP. H.323 - An ITU standard for real-time interactive voice and videoconferencing over LANs and the Internet.H.323 is an ITU standard for the usage of multimedia communication via packet-oriented networks that guarantees interoperability between different equipment vendors. The largest packet-oriented network is the Internet but also WAN, ISDN or dialup connections on which data is transported in packets (e.g. PPP) belong into this group. H.323 describes the general infrastructure and the utilization of different speech coders and protocol signalling stacks. The speech coders are defined in their respective sub standards, e.g. G.711 (Alaw and ulaw used in ISDN), G.722, G.723.1 and G.729.A for speech encoding. H.323 is definitely the most widely deployed and mature standard, but it is also criticized for being complicated to implement by vendors and uses a lot of resources which are not abundant (especially in terminals). H.450 - The H.450 Supplementary Services is a series of standards that define extended functionality and distribution thereof in a H.323 infrastructure. These services are called supplementary since they extend the basic services of H.323 which essentially boil down to being able to establish and release a connection. Examples of such Supplementary Services are Hold (local and remote), Call Waiting (an indication that a person is trying to reach someone who busy talking to someone else), Call Diverting (call is transferred when busy), Call Redirect (a call is transferred to a mobile after working hours), Pickup, Parking, etc – features that a classical PBX and ISDN offer. In addition, mechanisms are provided that enable vendors to tunnel proprietary supplementary services if need be – of course this is not intended to become a standard but is a workaround until these features are interesting enough to be integrated. IP PBX is a customer premises telephone system that manages telephones in the enterprise and acts as the gateway to external networks. Unlike a conventional PBX that requires two separate networks, one each for data and voice, an IP PBX is based on converged networks that enable true one-wire to the desktop connection. An IP PBX can be used with IP phones, softphones and traditional phones connected to Ethernet adapters (ATA) or PCs. IP telephony - (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems. IVR - Interactive Voice Response is a software application that accepts a combination of voice telephone input and touch-tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and perhaps other media. IVR is usually part of a larger application that includes database access. Common IVR applications include: bank and stock account balances and transfers, surveys and polls, caller authorization centers (for example, the MCCP), etc. Jitter is the variance of latency (i.e. delay) in a connection. The problem is that audio devices or connection-oriented systems (e.g. ISDN or PSTN) need a continuous stream of data. In order to compensate for this, VoIP terminals and gateways implement a jitter buffer that collect the packets before relaying them onto their audio devices or connection-oriented lines (e.g. ISDN), respectively. An increase in the jitter buffer size decreases the likelihood of data being missed but also has the drawback that it increases latency of a connection. Latency The delay or time span between the voice being digitalized at the senders Location and then output at the receivers end is the latency of a connection. Latency is influenced by the distance the data has to travel, the packet size, the number and delay time of network elements between the terminals and of course the latency generated by the terminals themselves when sending, receiving, encoding, decoding and compensating jitter. LPCP (Lightweight Phone Control Protocol) is a standard that is used to control telephones in a pragmatic and simple way. Thus, the memory and resources needed on VoIP telephones and terminals can be reduced to the minimum needed which in turn will be more cost effective. The call signalling (SIP / H.323) is done for the phone on a server that has enough memory resources. Megaco / H.248 - This is also a media gateway control protocol such as LPCP but more complex and general. MGCP Media Gateway Control Protocol; A protocol for IP telephony that enables a caller with a PSTN phone number to locate the destination device and establish a session. PBX Private Branch eXchange; An in-house telephone switching system that interconnects telephone extensions to each other as well as to the outside telephone network. PRI Primary Rate Interface; An ISDN service that provides 23 64-Kbps B (Bearer) channels and one 64-Kbps D (Data) channel (23 B and D). PSTN Public Switched Telephone Network; The
worldwide voice telephone network. QoS - Quality of Service pertains to
the quality of a connection and this
is especially important for connections relaying voice since the
user feels the impact immediately. A retransmission cannot make
up for the lost data. The internet protocol was devised as a “best
effort” data network and thus it does consider jitter, latency or
even data loss a problem. Ergo, it does not handle voice well per
se. To make the transmission of voice possible it must be given
the necessary priority and bandwidth. There are mechanisms for
reserving
bandwidth (see RSVP) but they add network equipment with an additional
burden of handling this functionality and slow down establishing
connections. The other pragmatic approach to this problem is to
acknowledge that normally the access point (interconnection between
LAN and WAN) is the most critical section. By prioritising the packets
(see ToS) (of course the network equipment has to Support this)
and ensuring that the access point is not overloaded good QoS can
be achieved. The data traffic load in the backbone is about 10 times
that of voice (thanks to WWW) of carriers so this should not be
the problem. RADIUS - Remote
Authentication Dial-In User Service. A client/server
protocol and software that is used for dial-in clients to connect to
other computers and networks. It provides authentication and accounting
when using PPTP or L2TP tunneling. RTP (realtime transport protocol)
labels all information
transferred by a sender with a timestamp. By examining the timestamps
the receiver is able to sort the packets in the original order and
synchronize real time streams and/or compensate jitter in audio data. RTCP (realtime transport control
protocol) was devised to give
Applications a status on the quality of a network. With this
information parameters affecting the transmission of data, e.g. the
jitter buffer size, can be optimized. RSVP (resource reservation protocol) makes it possible to reserve bandwidth in non-terminal network elements such as routers. This and prioritisation (see TOS) is done to practically eliminate latency, jitter and loss of packet problems for realtime application such as VoIP. SIP (Session
Initiation Protocol) is a highly pragmatic,
ASCII-based
protocol and competing standard to H.323. Its main advantages are
that it is easy to implement, debug and to integrate applications.
It is newer than H.323 and but does not standardize many of the
supplementary services, yet. At the moment, it is not pushing H.323
out of the scene. A protocol that provides telephony services
similar to H.323, but is less complex and uses less resources. It
creates, modifies, and terminates sessions with one or more
participants. Such sessions include Internet telephony and multimedia
conferences. SIP is a request-response protocol, dealing with requests
from clients and responses from servers. Session border controller (SBC)
A new category of network equipment that enables
interactive
communications across IP network borders. SBCs closely integrate
signaling and media control and serve as a transit point for all
signaling and media streams going through the service provider's
network. The ability to traverse firewalls and network address
translators ensures ubiquity of network reach, whilst advanced routing
and interworking capabilities maintain mission-critical quality of
service. Softswitch - (Also referred
to as media gateway controller or call agent).
The generic name for a new approach to telephony switching that has
evolved to enable transporting voice traffic over packet-switched
networks. At the most basic level, a softswitch is defined as media
gateway controller software that provides call control and resource
management for a media gateway. Call control relates to the setup and
termination of calls, including call routing. A softswitch also
provides call authentication and authorization, and accounting services
by accessing information available in an existing Signaling System 7
(SS7) network.
TAPI Telephony API; A programming interface that
allows Windows client applications to access voice services on a server. TOS - In order to specify the priority of a packet the internet protocol has a ToS (type of service) field. Trunk A communications channel between two points, typically referring to large-bandwidth telephone channels between switching centers that handle many simultaneous voice and data signals. CDR: Call Detail Record |