|
|
Glossary
Glossary of VoIP Terms
1-9
A
B
C
D
E
F
G
H
I
J
K
L
M
N
O
P
Q
R
S
T
U
V
W
X
Y
Z
| 1-9 |
 |
2.5G
The introduction and incremental speed increases of digital cellular, featuring integrated voice and data communications.
3G The third generation mobile communications
digital wireless devices on packet switched networks. Supporting simultaneous Voice and Data calls. Also supporting live video streaming.
3GPP Handsets (3rd Generation Partnership Project)
Handsets that comply with the worldwide standard for the creation, delivery and playback of multimedia over 3rd generation, high-speed wireless networks
|
|
| A |
|
API (Application Program Interface)
A set of calling conventions defining how a service is invoked through a software package.
Application Server
System software that provides a middleware interface between an operating system and the application programs of users.
ASP (Application Service Provider)
An independent, third-party provider of software-based services delivered to customers across a wide area network (WAN).
ASR (Answer-Seizure Ratio)
The ratio of successfully connected calls to attempted calls (also called 'Call Completion Rate').
ATM (Asynchronous Transfer Mode)
A technology for switched, connection-oriented transmission of voice, data and video.
|
|
| B |
|
Backbone
A very-high-speed network spanning the world from one major metropolitan area to another. Such networks are typically provided by national Internet service providers (ISPs). Local ISPs connect to the backbone in order to transport data.
Bandwidth
The maximum data carrying capacity of a transmission link. For networks, bandwidth is usually expressed in bits per second (bps).
Broadband
Descriptive term for evolving digital technology that provides consumers a single switch facility offering integrated access to voice, high-speed data service, video demand services, and interactive delivery services.
|
|
| C |
|
Call
Establishment of (or an attempt to establish) voice connection between two endpoints, or between two points which provide a partial link (e.g. a trunk) between two endpoints.
Call deflection
Call Deflection is a feature under H.450.3 Call Diversion (Call Forwarding) that allows a called H.323 endpoint to redirect the unanswered call to another H.323 endpoint."
Carrier Class
Products designed specifically to meet the capacity, performance scalability, availability and network management requirements of network service providers.
Codec (Compression-decompression)
A voice compression-decompression algorithm that defines the rate of speech compression, quality of decompressed speech and processing power requirements. The most popular codecs in VoIP are ITU-T G.723.1 and G.729 (AB).
Compression
compression is used at anywhere from 1:1 to 12:1 ratios in VOIP applications to consume less bandwidth and leave more for data or other voice/fax communications. The voice quality may decrease with increased compression ratios.
Congestion
The situation in which the traffic present on the network exceeds available network bandwidth/capacity.
Connection-oriented
Mode of communication in which a connection must be established between the transmitter and receiver before transmission of user data. This can be done by switching a circuit or by setting up a logical channel. A well-known connection-oriented protocol is TCP. Connection-oriented is the opposite of connectionless.
Connectionless
Mode of communication in which a connection (circuit or logical channel) does not need to be set up for data transmission between the transmitter and receiver. It is the underlying protocol for packet-switched transmission. The individual data packets can go from the transmitter to the receiver via different paths. A well-known connectionless protocol is UDP.
Converged Network
A single network capable of carrying a mixture of voice (telephone), video (production and training), and application data.
CSMA/CD
Carrier Sense Multiple Access/Collision Detection This is the access procedure to the Ethernet in which the participating stations physically monitor the traffic on the line. If no transmission is taking place at the time the particular station can transmit. If two stations attempt to transmit simultaneously this causes a collision which is detected by all participating stations. After a random time interval the stations that collided attempt to transmit again. If another collision occurs the time intervals from which the random waiting time is selected are increased step by step. Networks using the CSMA/CD procedure are simple to implement but do not have deterministic transmission characteristics. The CSMA/CD method is internationally standardized in IEEE 802.3 and ISO 8802.3.
|
|
| D |
|
DiffServ
DiffServ (Differentiated Services) is a quality of service protocol that prioritizes IP voice and data traffic to help preserve voice quality even when network traffic is heavy.
|
|
| E |
|
E&M
(Ear and Mouth) is the interface on a VOIP device that allows it to be connected to analog PBX trunk ports (tie lines).
E.164
The international public telecommunication numbering plan. An E.164 number uniquely identifies a public network termination point and typically consists of three fields, CC (country code), NDC (national destination code), and SN (subscriber number), up to 15 digits in total.
E1
A wide-area digital transmission scheme (European): 2,048 Mbits/s; 31 channels, 64 Kbps each.
End Date
Date determining the end of the period during which IPCB.net Members (Sellers) will terminate calls to the Gateways specified in a particular Contract.
Endpoint
SIP or H.323 terminal or Gateway. An endpoint can Call and be Called. It generates and terminates the information stream.
|
|
| F |
|
Firewall
A system designed to prevent unauthorized access to or from a private network. Firewalls can be implemented as hardware, software, or a combination of both. All messages entering or leaving the intranet pass through the firewall, which examines each message and blocks those that do not meet the security criteria specified on the firewall.
Forward Error Correction
increases voice quality by recovering lost or corrupted packets.
Framework
A SIP stack component that offers the common architecture, generic classes, structures and mechanisms to all other components and allows the user to create specialized services. It adds at a common level that is required to support the M5T SAFE product line such as: IP, TLS sockets, Kerberos, S/MIME, Crypto algorithms, Certificate and Secure RND.
FXO
(foreign exchange office) is the interface on a VOIP device for connecting to an analog PBX extension.
FXS
(foreign exchange station) is the interface on a VOIP device for connecting directly to phones, faxes, and CO ports on PBXs or key telephone systems.
|
|
| G |
|
G.711
An ITU-T PCM half-duplex codec that uses either A-law or ?-law compression (64 kbps, high quality, minimum processor load).
G.723.1
An ITU-T double rate CELP codec (6.4/5.3 kbps, medium quality, high processor load).
G.726
An ITU-T ADPCM wave form codec (16/24/32/40 kbps, good quality, low processor load).
G.728
An ITU-T low delay CELP codec (16 kbps, medium quality, very high processor load).
G.729
An ITU-T ACELP codec (8 kbps, medium quality, high processor load).
G.7xx
A family of ITU standards for audio compression
Gatekeeper
The central control entity that performs management functions in a Voice and Fax over IP network and for multimedia applications such as video conferencing. Gatekeepers provide intelligence for the network, including address resolution, authorization, and authentication services, the logging of Call Detail Records, and communications with network management systems. Gatekeepers control bandwidth, provide interfaces to existing legacy systems, and monitor the network for engineering purposes as well as for real-time network management and load balancing
Gateway
In IP telephony, a network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of an IP gateway include voice and fax compression/ decompression, packetization, call routing, and control signalling. Additional features may include interfaces to external controllers, such as Gatekeepers or Softswitches, billing systems, and network management systems.
|
|
| H |
|
H.225
Protocols (RAS, RTP/RTCP, Q.931 call signalling) and message formats for H.323.
H.245
A protocol for capability negotiation, messages for opening and closing channels for media streams, etc. (i.e. media signalling).
H.323
An ITU-T "umbrella" of standards for Packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoints, Gateways, Multipoint Conferencing Units (MCUs), and Gatekeepers -- and their interaction. This standard is used for many Voice-over-IP applications, and is heavily dependent on other standards, mainly H.225 and H.245.
Hairpin
Telephony term that means to send a call back in the direction that it came from. For example, if a call cannot be routed over IP to a gateway that is closer to the target telephone, the call typically is sent back out the local zone, back the way from which it came.
Hop off
Point at which a call transitions from H.323 to non-H.323, typically at a gateway. "be hopped-off locally" means "be hairpinned" Example from documentation: "If the called address does not match any known zone prefixes, the gatekeeper will attempt to hairpin the call out through a local gateway
|
|
| I |
|
IAD/CPE (Integrated Access Device / Customer Premises Equipment)
Equipment, such as terminals and modems, supplied by the telephone company, that is installed at customer sites and connected to the telephone company network.
Instant Messaging
A form of electronic communication which involves immediate correspondence between two or more users who are all online simultaneously.
Integrated Circuits (IC)
a small electronic device made out of a semiconductor material, integrated circuits are used for a variety of devices, including audio equipment. Integrated circuits are often classified by the number of transistors and other electronic components they contain
Integrated T-1
Comprised of 24 64Kbps channels, T1 lines can be used for a diverse number of applications. Commonly referred to as an integrated T1 or channelized T1, this highly flexible circuit is designed for businesses that need to run multiple services over the same line. Common applications for integrated T1 service include, Frame Relay/dedicated long distance and Internet/point-to-point. Often confused with a fractional T1, integrated service is made up of multiple fractional T1 services.
Internet Telephony
Refers to technology that enables routing of voice conversations over the Internet or any other IP network. The voice data flows over a general-purpose packet-switched network, instead of the traditional dedicated, circuit-switched voice transmission lines.
IP Centrex IP
Centrex delivers such services as call hold, call transfer, last number look-up and redial, call forward, three-way calling, but does it on a packet-based network.
IP PBX
(The Internet Protocol Private Branch eXchange)Telephone switching equipment that resides in a private business instead of the telephone company. An IP PBX delivers employees dial-tone, the ability to conference, transfer, and dial other employees by extension number as well as many other features. Voice transmissions are sent via data packets over a data network instead of the traditional phone network
IP Telephony
The transmission of voice and fax phone calls over data networks that uses the Internet Protocol (IP). IP telephony is the result of the transformation of the circuit-switched telephone network to a packet-based network that deploys voice-compression algorithms and flexible and sophisticated transmission techniques, and delivers richer services using only a fraction of traditional digital telephony’s usual bandwidth. Compare with VoIP.
ITSP
Internet Telephony Service Provider.
ITU-T
ITU standards for telecommunications.
|
|
| J |
|
Java
An multi-platform, object-oriented programming language from Sun Microsystems. The Java language syntax is somewhat similar to C. Java can be used to program applications and applets.
Jitter
The variation in the amount of Latency among Packets being received
|
|
| K |
|
Kerberos
An authentication system designed to enable two parties to exchange private information across an otherwise open network. It works by assigning a unique key, called a ticket, to each user that logs on to the network. The ticket is then embedded in messages to identify the sender of the message.
|
|
| L |
|
LAN
A local area network (LAN) is a group of computers and associated devices that share a common communications line or wireless link and typically share the resources of a single processor or server within a small geographic area (for example, within an office building).
Latency (Also called Delay)
The amount of time it takes a Packet to travel from source to destination. Together, Latency and Bandwidth define the speed and capacity of a network.
Load Balancing
Distribution of calls among terminating Gateways based on the Priorities and Weights assigned by the Buyer.
|
|
| M |
|
Managed LAN
The Managed LAN Switch feature enables the control of the four switch ports in Cisco 831, 836, and 837 routers. Each switch port is associated with a Fast Ethernet interface.
MGCP (Media Gateway Control Protocol)
A protocol complementary to H.323 and SIP, designed to control media gateways from external call control elements in decomposed gateway architectures. Working in conjunction with the Gateway Location Protocol (GLP), MGCP enables a caller with a PSTN phone number to locate the destination device and establish a session. It provides the gateway-to-gateway interface for the Session Initialization Protocol (SIP). MGCP is meant to simplify standards for the new Voice over Packet technology by eliminating the need for complex, processor-intense IP telephony devices, thus simplifying and lowering the cost of these terminals.
Mgmt-IX (Management Information exchange)
A C++ based set of classes that provides the provisioning for application modules. Mgmt-IX establishes and enhances information exchange capabilities between multiple entities.
MIKEY (Multimedia Internet KEYing)
A protocol designed for exchanging cryptography keys quickly for real-time multimedia streams.
Minimum Duration
The minimum billed call duration up to which all shorter calls are rounded in seconds.
MMoIP (Multimedia over IP)
Multimedia messaging technology that allows the transmission of voice, data, and moving images over the IP.
Mobile Telephone
A device which behaves as a normal telephone whilst being able to move over a wide area. Mobile telephones allow connections to be made to the telephone network, normally by directly dialling the other party's number on an inbuilt keypad.
|
|
| N |
|
|
|
| O |
|
|
|
| P |
|
P2P (Peer-to-Peer)
A decentralized network with no fixed service which routes the data packets from one user to another. A large number of users on the network could become the decentralized servers depending on the availability of resources.
Packet
The basic logical unit of information transferred
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.
PTT (Push-to-Talk)
A method of conversing on half-duplex communication lines, including two-way radio, by pushing a button in order to send, allowing voice communication to be transmitted from you, and releasing to let voice communication be received.
|
|
| Q |
|
Q.931
ISDN connection control protocol, roughly comparable to TCP in the Internet protocol stack. Q.931 doesn't provide flow control or perform retransmission, because the underlying layers are assumed to be reliable and the circuit-oriented nature of ISDN allocates bandwidth in fixed increments of 64 kbps. Q.931 does manage connection setup and breakdown. In H.323 scenario, this protocol is encapsulated in TCP and sent to port 1720.
QoS
Quality of Service. Measure of performance for a transmission system that reflects its transmission quality and service availability. Standards based QOS for VoIP usually involves the implementation of ethernet standards 802.1p and 802.1q at layer 2 across an Ethernet. At layer 3, the IP standard DiffServ defines bits settings in the TOS (type of service) in the IP header which will identify packets as being associated with a specific service.
QSIG Q (point of the ISDN model)
Signalling. Signalling standard. Common channel signalling protocol based on ISDN Q.931 standards and used by many digital PBXs.
|
|
| R |
|
Redundant
Redundant describes computer or network system components, such as fans, hard disk drives, servers, operating systems, switches, and telecommunication links that are installed to back up primary resources in case they fail. A well-known example of a redundant system is the redundant array of independent disks (redundant array of independent disks).
Ring
A ring is a network topology or circuit arrangement in which each device is attached along the same signal path to two other devices, forming a path in the shape of a ring. Each device in the ring has a unique address. Information flow is unidirectional and a controlling device intercepts and manages the flow to and from the ring.
RTCP (Real-Time Control Protocol)
the control protocol for RTP (Real-time Transport Protocol). It is used to periodically transmit control packets to participants in a streaming multimedia session. RTCP's primary function is to provide feedback on the quality of service being provided. This feedback may be used to scale back the sender for flow-control reasons or to keep from congesting the network.
RTP (Real-Time Transport Protocol)
Commonly used with IP networks. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides such services as payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications.
|
|
| S |
|
SDP (Session Description Protocol)
A protocol that describes a format for conveying descriptive information about multimedia sessions
Server
A host computer on a network that holds information and responds to requests for information from it (. The Server is also used to refer to the software that makes the act of serving information possible.
SIGCOMP (Signalling Compression)
Enables compression and decompression of messages sent across the network, which have stringent bandwidth restrictions.
SIP (Session Initiation Protocol)
An application-layer control protocol, a Signalling protocol for Internet Telephony. SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, e-mail, and chat services. In addition to user authentication, redirect and registration services, SIP Server supports traditional telephony features such as personal mobility, time-of-day routing and call forwarding based on the geographical location of the person being called.
SIP Application
A managed code application that runs in a separate process. It consists of an application manifest and at least one Server Agent object.
SIP Client
An RTC (real-time communications) client that uses SIP (Session Initiation Protocol) to establish and maintain RTC sessions with a SIP server.
SIP Server
A server that uses SIP (Session Initiation Protocol) to manage real-time communication among SIP clients.
SIP Stack
A SIP Stack set of security tools, protocols and components targeted at facilitating SIP based VoIP product and solution development. The SIP Stack is comprised of SIP User Agent and other SIP Server components such as MIKEY, SIGCOMP, RTP, SRTP, STUN, MGCP
SIP-UA (SIP User Agent)
A device that transmits SIP packets over IP
SIP/PSTN gateway
A protocol conversion application situated at the boundary between the PSTN (public switched telephone network) and an IP network enabled for SIP (Session Initiation Protocol). The gateway converts SIP signalling protocols to PSTN equivalents, and vice-versa, thereby enabling call conversion between the two networks.
Softswitch
(Also called a Proxy Gatekeeper, Call Server, Call Agent, Media Gateway Controller, or Switch Controller) Software used to bridge a public switched telephone network and voice over Internet by separating the call control functions of a phone call from the media gateway (transport layer). Softswitch performs call control functions such as protocol conversion, authorization, accounting and administration operations.
SRTCP (Secure Real-time Transport Control Protocol)
the control protocol for SRTP (Secure Real-time Transport Protocol).
SRTP (Secure Real-time Transport Protocol)
A security profile for RTP that adds confidentiality, message authentication, and replay protection to that protocol. It is specified in RFC 3711.
STUN (Simple Traversal of UDP through NATs (Network Address Translation))
A protocol for assisting devices behind a NAT firewall or router with their packet routing. STUN enables a device to find out its public IP address and the type of NAT service its sitting behind. It operates on TCP and UDP port 3478
STUN Client
An entity that generates STUN requests. A STUN client can execute on an end system, such as a user's PC, or can run in a network element, such as a conferencing server
STUN Serveran entity that receives STUN requests, and sends STUN responses. STUN servers are generally attached to the public Internet.
|
|
| T |
|
T1
1.544-Mbps point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels.
TAPI Telephony API;
A programming interface that allows Windows client applications to access voice services on a server.
TCP (Transmission Control Protocol)
Connection-oriented transport layer protocol that provides reliable full-duplex data transmission. TCP is part of the TCP/IP protocol stack.
ToS (Type of Service)
An 8-bit field in the IP datagram header that identifies the relative priority of one packet over another. Networking devices use this field to prioritize packets appropriately and place them in different queues if necessary.
TOS Type of Service;
A method of setting precedence for a particular type of traffic for QoS.
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.
Trunking
Trunking means that several connections in a network may be established simultaneously, and that setup of connections proceeds automatically using the channels available at the time in question. In this way many users may share a few connections, and if the number of connections is increased, the capacity of the network is increased more than proportionally. This means that an optimal trunking effect is obtained in very large networks.
|
|
| U |
|
UAC (User Agent Client)
A client application that initiates the SIP request
UAS (User Agent Server)
A server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects or redirects the request
|
|
| V |
|
VoIP Trunking
Service providers can use this application to connect enterprise and call center customers directly to their VoIP network. By bypassing local operators and toll charges, the VoIP Trunking application enables service providers to offer attractive termination rates for both domestic and international long distance calling. This application connects seamlessly to the enterprise/call center's PBX, allowing employees to make off-net calls at reduced rates.
VoIP Voice over IP
The capability to carry normal telephony-style voice over an IP-based Internet with POTS-like functionality, reliability, and voice quality. VoIP enables a router to carry voice traffic (for example, telephone calls and faxes) over an IP network. In VoIP, the DSP segments the voice signal into frames, which then are coupled in groups of two and stored in voice packets. These voice packets are transported using IP in compliance with ITU-T specification H.323.
VPDN virtual private dial-up network
Also known as virtual private dial network. A VPDN is a network that extends remote access to a private network using a shared infrastructure. VPDNs use Layer 2 tunnel technologies (L2F, L2TP, and PPTP) to extend the Layer 2 and higher parts of the network connection from a remote user across an ISP network to a private network. VPDNs are a cost effective method of establishing a long distance, point-to-point connection between remote dial users and a private network
VPN Virtual Private Network
Enables IP traffic to travel securely over a public TCP/IP network by encrypting all traffic from one network to another. A VPN uses “tunnelling” to encrypt all information at the IP level.
|
|
| W |
|
Weight
A number (10-100) assigned to a Contract or Route when ordering the Contract/Route. If several Contracts/Routes for the same destination have the same Priority assigned, calls to the destination are distributed among the Contracts/Routes according to their relative Weights.
Wireless
Communications, monitoring or control systems in which electromagnetic or acoustic waves carry a signal through atmospheric space rather than along a wire.
|
|
| X |
|
|
|
| Y |
|
|
|
| Z |
|
|
Products /