7. Glossary

Anonymous Call Rejection Enables a user to reject calls from anonymous parties who have explicitly restricted their Caller ID. By activating, callers without available caller identification are informed that the user is not accepting calls at that time. The users phone does not ring and the user sees or hears no indication of the attempted call.

ATA (Analog Telephone adapter) Device that coverts analog voice signals to digital signals which can then be transmitted over the Internet.

Authentication Authentication is performed upon the registration of an IP phone. This ensures that the user of the device is authorised to gain access into byphone. SIP invites may also be authenticated on an on-going basis at pre-defined intervals. Standard digest authentication is used. The authentication information is configured both in the phone and via the phone system. All initiated calls from unregistered phones are denied.

Auto Attendant An automatic response system, such as a voice presenting options such as press 2 for sales, 3 for accounts, etc., which handles incoming calls and sends them to the appropriate phone or message. See also IVR

Bandwidth Usually measured in 1000 bits per second (kbps), it is the amount of data that can be carried from one point to another in a given time period.

Baud One signalling element per second, not to be confused with bits per second.

BYOD (Bring Your Own Device) Some businesses allow a person to supply their own equipment such as Ipads, laptops, mobile phones etc.

Call Forwarding Enables a user to redirect all incoming calls to another phone number. Users have the option to activate and deactivate the service by dialling a feature code or configuring the service via their web interface. If activated, a user must specify the forwarding number.

Call Forwarding Busy Enables a user to redirect calls to another destination when an incoming call encounters a busy condition. Users have the option to activate and deactivate the service by dialling a feature code or configuring the service via their web interface. If activated, a user must specify the forwarding number.

Call Forwarding No Answer Enables a user to redirect calls to another destination when an incoming call is not answered within a specified number of rings. Users have the option to activate and deactivate the service by dialling a feature code or configuring the service via their web interface. If activated, a user must specify the forwarding number and the number of rings before forwarding.

Call Transfer- Blind Enables a user to transfer a call unattended before or after the call is answered.

Call Transfer – Attended Enables a user to transfer a call attended, providing the recipient with details of the caller before transferring.

Call Waiting Enables a user to answer a call while already engaged in another call. When a second call is received while a user is engaged in a call, the user is informed via a call waiting tone. The user connects with the waiting party and holds the original party. By depressing the flash hook, the user reconnects to the original party and holds the waiting party. The feature completes when any party hangs up.

Caller ID Blocking Enables a user to block delivery of his/her identity to the called party.

Cloud Communications 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.

Codec Normally used to reference to converting analog signals to digital or digital signals to analog. It can be used in conjunction with compression software to compress and decompress these signals to varying degrees.

Data Usually treated as a synonym for information, but when used as a description for network topology refers to all traffic other than voice.

Data Transfer Rate The speed of travel of a given amount of data from one place to another.

Device A device is the phone or computer that you make a call through. A device is linked to a user via an Identity. Many devices can have multiple Identities allowing a user to use each to fulfil different roles.

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 Dialling) A service that allows an enterprise to allocate individual phone numbers to each person within its PBX system.

DND (Do Not Disturb) Allows users to set their station as unavailable so that incoming calls are given a busy treatment. Users have the option to activate and deactivate the service by dialling a feature code or configuring the service via their web portal.

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) it is the signal generated when you press a telephone's touch keys that is sent to the telephone company. These signals are 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 standardised and are uniquely linked to a number (and # or *) on the telephone keypad.

Echo Cancellation 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. Extension Dialling Enables users to dial extensions to call other members of their business group.

Follow Me Enables users to define a list of phone numbers that are alerted sequentially for incoming calls that match specified criteria.

IAD (Integrated Access Device) Equipment at the customers location that is used to convert digital signals back to voice. Usually used in association with a DSL connection.

IAX (Inter-Asterisk eXchange Protocol), (pronounced “eeks”) (now commonly meaning IAX2) is an Asterisk communications protocol for setting up interactive user sessions (both audio and/or video) and supports any type of codec.

Identity A set of SIP credentials owned by a Byphone user, which is then registered using a SIP device facilitating the making and receiving of phone calls on the Byphone system.

IVR (Interactive Voice Response) An integrated software information system that speaks to callers and uses menus and voice responses. By using touch-tone keypad entries to interact with the software, you get voice responses with real time data.

Jitter 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.

Latency 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 recognised with latency higher than 150 milliseconds. Higher than 500 milliseconds and the conversation is going to be very problematic.

LERG (Local Exchange Routing Guide) Is a database of the first 6 digits of a telephone number, updated on a regular basis, that provides information for routing telephone calls over the Public Switching Telephone Network, as well as, enables identification of what local company the number belongs to.

Line Echo Echo that is common in the PSTN network and is created as a result of voice travelling over hybrids or 2 wire to 4 wire conversions.

MOS (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.

MTA (Multimedia Terminal Adapter) A device that connects a traditional telephone to a cable line, converting analogue voice to digital signals.

NAT (Network Address Translation) 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 are a challenge for VoIP and result in one-way audio in some cases.

PBX (Public Branch Exchange) A private telephone switching system that allows outside phone lines from a telecommunications provider to connect to extensions within the office or building. They usually have multiple features including call forwarding, time conditions and voice mail.

POTS (Plain Old Telephone System) The single phone line, single phone number system that has been in existence for many years.

PRI (Primary Rate Interface) The Primary Rate Interface consists of 23 B-channels and one 64 Kpbs Dchannel using a T-1 line and can have up to 1.544 Mbps service. Typically, it is a dynamic circuit that delivers both voice and data, giving preference for voice. When a channel is not carrying voice, it is automatically allocated for data.

PSTN (Public Switched Telephone Service) The combination of local, long-distance and international carriers that make up the worldwide telephone network.

PSU (Power Supply Unit) The electrical plug used to power a handset.

Remote Office Enables users to access and use their byphone service from any end point, on-net or off-net (e.g., home office, mobile phone). This service is especially useful for tele-workers and mobile workers, since calls are still originated from byphone, the service provides an easy mechanism for separating personal and business phone expenses, as well as keeping alternate phone numbers private.

RJ-11 The typical four or 6 wire connector used to connect telephone equipment.

RJ-45 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.

Router A router is a device connected to at least two networks that determines the next network point to forward a packet to. The decision of which way to send each information packet is based on it's current understanding of the networks that it is connected to.

RTP (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) is a signalling 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.

SIP (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 worldwide by connecting to an ITSP (Internet Telephony Service Provider). SIP trunking can save money and offer services to an IP-PBX.

Soft Phone IP telephony software that allows end users to send and receive calls over the computer or hand held PC device (PDA) over the Internet. STUN (Simple traversal of UDP through NATs) is a protocol for assisting devices behind a NAT firewall or router with their packet routing.

Switch A switch is a device that keeps a record of the MAC addresses of all devices connected to it and then channels incoming data from any of multiple input ports to the specific output port that will take the data toward its intended destination.

UDP (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.

VoIP (Voice over Internet Protocol) The transmission of voice over the Internet as digital packets rather than the traditional circuit-committed protocols of the PSTN. VoIP uses real-time protocol (RTP) to help ensure that the packets get delivered in a timely way.