ATA – Analog Telephone Adaptor
An Analog Telephone Adaptor is a device that converts audio, data and video signals into Internet Protocol (IP) packets that can then be sent over the Internet. It can be used to connect legacy telephones to a high bandwidth line to place voice calls using Voice over Internet Protocol (VoIP).
Bandwidth refers to how much data can be transferred during a given time period.
CDR (Call Record Detail)
Details about a specific call that includes duration, origination, destination, and billable information, as well as other pertinent information.
Cloud refers to the Internet. Cloud Communications uses the Internet as a way to have users connect to host equipment at a remote location which then connect to other users allowing phone calls. Synonymous with hosted VoIP or Internet Phone Service.
A codec converts a data stream to be delivered, received, used, stored or encrypted. Codecs are especially useful for video streaming and conferencing. Codecs are so called because they both COmpress or DECompress and encode or decode data packets.
CSR (Customer Service Record)
A document required for all phone numbers that will show information that is tied to that number including services, billing activity, associated address and service orders.
DHCP (Dynamic Host Control Protocol)
A communications protocol that lets network administrators supervise and distribute IP addresses from a central point to each computer or device on a network.
DID (Direct Inward Dialing)
A service that allows an enterprise to allocate individual phone numbers to each person within its PBX system.
DSL (Digital Subscriber Line)
Phone technology that allows a broadband internet digital connection to be carried over existing copper phone lines, while still allowing the phone service carry analog signals over the same line.
DTMF (Dual Tone Multi-Frequency)
Also known as Touchtone, it is the signal generated when you press a telephone's touch keys that is sent to the telephone company. These signals are actually two tones of a specific frequency designed so that a voice cannot duplicate them. The ability for interactive telephone menus to work correctly across different networks and phone systems is due to the fact that DTMF tones are standardized and are uniquely linked to a number (and # or *) on the telephone keypad.
Enhanced 911, or E911 for short, is a system dedicated to connecting mobile and internet phone users to emergency services.
Echo cancellation is the process of eliminating echo from voice communication to improve the quality of the call. It is necessary because speech compression techniques and packet processing delays generate echo, of which there are 2 types, acoustic echo and hybrid echo. Echo cancellation improves voice quality in VoIP calls and also reduces the required bandwidth due to silence suppression techniques.
ECM (Error Correction Mode)
Used in conjunction with memory storing fax machines, ECM allows for the receiving fax machine to request retransmission for a page where some errors were detected in the frames of that page. If the receiving fax machine is unable to receive an error free page the fax transmission may fail and the fax connection terminated. On networks with some packet loss, fax transmissions will routinely fail when ECM is enabled because of the low tolerance allowed for any packet loss.
Frame Relay is a packet switching method that uses available bandwidth only when it is needed. This fast packet switching method is efficient enough to transmit voice communications with proper network management.
G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. G.711 represents logarithmic pulse-code modulation (PCM) samples for signals of voice frequencies, sampled at the rate of 8000 samples/second.
G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus use G.711 or out-of-band methods to transport these signals.
Voice over Internet Protocol (VoIP) systems use gateways to convert voice calls and fax communications received from the Public Switched Telephone Network (PSTN) into digital packets that are then transmitted over an Internet Protocol (IP) network.
Interactive Voice Response (IVR)
An IVR platform uses computer telephony components to translate callers’ touch-tones or voice commands into computer queries after the callers listen to an audio menu such as, “Please press one to be connected to an agent.” These queries are then “fetched” by the IVR platform from the host computer.
Internet/ IP/ VoIP Telephony
Used interchangeably, Internet telephony, IP telephony, and Voice over Internet Protocol (VoIP) all refer to voice communications sent and received over the Internet.
Internet Protocol (IP)
IP is the communications system that routes data packets from one computer to another over the Internet.
An Internet Protocol (IP) address is a fixed or dynamic number ranging from 0.0.0.0 to 255.255.255.255 required for every internet connected device.
As data load increases and decreases, routers on the Internet can create slightly different times that individual packets take to travel from one point to another point. This variation in time is known as jitter.
The time it takes for a packet to reach its destination. Higher delay times can be an issue, especially for VoIP, where voice delay can be recognized with latency higher than 150 milliseconds. Higher than 500 milliseconds and the conversation is going to be very problematic.
LNP (Local Number Portability)
Local Number Portability is the ability of a telephone customer to retain their phone number if they switch to another telephone provider.
MAC Address (Media Access Control Address)
A MAC address is a unique identifier usually assigned by a device manufacturer, and is sometimes referred to as the hardware or physical address. The MAC address enables VOIP traffic to be routed to a specific device, such as an ATA or IP phone.
MOS (Mean Opinion Score)
The Mean Opinion Score provides a numerical indication of the perceived quality of voice transmission after compression and/or transmission and is expressed as a number in the range 1 to 5, where 1 is lowest perceived audio quality and 5 is the highest perceived audio quality measurement.
NAT (Network Address Translation)
NAT is an Internet standard allowing a local network to use one public IP address to connect to the Internet and a set of local IP addresses to identify each PC or device in the local network. NAT translations require specific configurations for VoIP and can result in one-way audio in some cases.
A packet is a logically grouped unit of data. Packets contain the information to be transmitted as well as information regarding the originator, destination, and synchronization. The transmission process allows for packets to be sent along the most optimal route to its destination. Packets are constructed on one end of the communication and de-constructed on the receiving end.
PCAP (Packet Capture)
PCAP is an application programming interface (API) that allows for capturing network traffic. This is useful in diagnosing connectivity issues on VOIP networks and equipment.
PoE (Power over Ethernet)
Power over Ethernet is a technology used to transmit electrical power along with data to remote devices over standard ethernet cable in a network. This technology is useful for powering IP telephones or network cameras where it would be costly to run power seperately.
POTS (Plain Old Telephone System)
The familiar single phone line, single phone number system that has been in existence for many years.
Private Branch Exchange (PBX) or Private Automatic Branch Exchange (PABX)
A Private Branch Exchange (PBX) or Private Automatic Branch Exchange (PABX) systems replaces the traditional telephone switchboard with a private telephone network exchange. PBX phone systems offer additional features such as caller greetings, conference calling, call holding or extension routing.
Public Switched Telephone Network (PSTN)
The public circuit-switched telephone networks make up the PSTN. At one time, the PSTN consisted of analog telephone systems, now the PSTN is almost entirely digital, and includes mobile phones in addition to POTS (Plain Old Telephone Service) lines.
QOS – Quality of Service
Quality of Service in networking terms refers to the ability of a network to give guaranteed performance based on some metric usually by prioritising traffic. For example certain types of network traffic such as voice over ip can be prioritized on a network to make sure that a certain minimum level of quality is alway obtained.
The geographic area used by local exchange carriers to set rate boundaries for issuing phone numbers and for billing. Rate centers are important when considering porting of numbers through LNP to and from VoIP service providers.
The typical four or 6 wire connector used to connect legacy telephone equipment.
An 8 wire connector used to connect Ethernet connections in computers, routers and other Internet devices. This connector is slightly larger than a (RJ-11) telephone connector.
Real Time Transport Protocol An Internet protocol that functions for end-to-end network connections for applications that use audio or video.
SIP (Session Initiation Protocol)
Session Initiation Protocol is a signaling protocol for Internet conferencing, telephony, and instant messaging. It is a request-response protocol, dealing with requests from clients and responses from servers initiating an interactive user session.
Session Initiation Protocol Trunking is the use of VoIP to facilitate the connection of typically a PBX to the Internet, where the Internet replaces the conventional telephone trunk, allowing a business to communicate with traditional PSTN telephone subscribers by connecting to an ITSP (Internet Telephony Service Provider). SIP trunking can save money and offer enhanced services to an IP-PBX.
A softphone is software for desktop, laptop or tablet computers that provides a functional Voice over Internet Protocol (VoIP) telephone service. Softphones receive inputs from a microphone and outputs through the computer’s speakers or a set of headphones.
A recognized standard for sending fax transmissions over an IP network in real time mode. Messages are sent as UDP or TCP/IP packets.
TCP/IP (Transmission Control Protocol/Internet Protocol)
TCP/IP were the first two networking protocols defined and continue to form the basis of internet traffic.
User Datagram Protocol is a communications protocol that does not provide sequencing of the packets. The application must be able to make sure that the entire message has arrived and is in the right order.
Unified Messaging (UM)
UM is a term that refers to integrating different electronic messaging and communications media (e-mail, SMS, fax, voicemail, video messaging, etc.) technologies into a single interface, accessible from a variety of different devices.
Virtual Phone Number
An additional phone number assigned to an existing phone line in order to save a caller long distance fees, such as having a Los Angeles telephone number for a business located in Chicago. The caller pays no long distance and communicates with the party as if they were local.
VoIP – Voice over IP
Voice over Internet Protocol is the routing of voice conversations over the Internet or through any other IP-based network. Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques.
Wi-Fi phones provide voice and data communications over the Internet whenever a wireless signal is available.
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