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PCS Glossary

Due to the rapid pace of technological change, terms and definitions tend to vary among different individuals and situations. This glossary contains definitions for terms as they are used in our software, training materials, and online help.

A-B

Term	Definition
A Record	Also known as an Address Record, this is a DNS (domain name system) parameter that associates an IP address with a domain (or website) name. This allows DNS to locate the correct physical device on the internet when you enter a website name. NOTE: When your portal was built, an A Record was created to associate the IP address of a CoreDial server with your Portal Domain Name.
ACD (Automated Call Distributor)	Also called Automatic Call Distribution system, this refers to the software that runs a call center. ACDs are primarily responsible for sending (or routing) incoming calls to the correct destinations and typically include auto attendant functions, features for managing call center agents and queues, tools to collect and report statistics, and customizable routing options.
Analog	In telephony, analog refers to a process of converting sound into electronic signals and sending them between two devices. Analog signals travel in a continual stream (or wave) that fluctuates as the sound level changes. The PSTN (public switched telephone network) was originally built with analog technology; however, it’s now mostly digital. The terms analog, landline, and POTS (plain old telephone service) typically refer to a single connection to the PSTN such as a traditional household phone line.
Analog Gateway	Also known as PSTN Gateways, these devices generally connect LANs to the PSTN, so that IP (LAN) phones can access the PSTN. They may also connect analog phones directly to IP LANs, so that analog phones can communicate over a VoIP network, (small gateways are often called Analog Telephone Adapters). Note: Analog Gateways are not used to connect legacy PBXs to CoreDial SIP Trunks.
Analog Telephone Adapter (ATA)	ATA refers to a category of devices that connect analog phones to VoIP networks. ATAs generally include at least one RJ-11 (or FXS) phone port and one ethernet port. This allows analog phones to connect directly to a LAN and access an IP PBX or an internet-based VoIP service such as CoreDial’s Hosted PBX. There are many varieties of ATAs, and models with multiple phone ports are often called Analog Gateways. Some modems include built-in ATAs that allow a few remote phones to use an internet VoIP service via direct cable or DSL connection (without a LAN).
API (Application Programming Interface)	APIs provide a way for two different programs (or applications) to interact with each other. CoreDial’s open (public) API exposes active call control and dialing data.
Asynchronous Transfer Mode (ATM)	The ATM standard was developed in the 1980s to transmit voice, video, and data over a single connection. It encapsulates network traffic into small, fixed-sized cells (or packets) and transmits them over a virtual, point-to-point, high-speed connection. This differs from internet protocol (IP), which uses variable sized packets and multi-point connections. As IP evolved and became more efficient, it essentially replaced ATM and became the dominant standard for transmitting voice, video, and data over a network.
Attended Transfer	This refers to a notification process that occurs before transferring a call.  It is also known as Announced, Warm, or Supervised Transfer.When a call needs to be transferred, it is first put on hold.  The intended recipient is then called and notified of the party on hold (who cannot hear the exchange). If the recipient agrees to accept the call, the transfer is completed.
Auto Attendant (AA)	Short for Automated Attendant, AAs are also known as “Virtual” Attendants, Receptionists, or Operators. Their primary function is to automate the process of answering and routing incoming calls. AAs answer calls with a prerecorded message that offers a menu of options for completing the call. Callers select an option by pressing a button on their phone, and the AA sends (or routes) the call to the correct destination.  The Account Manager AA can route calls to 19 different destinations such as voicemail, external numbers (including mobile), ring groups, extensions, company directories, other AAs, general mailbox, etc.Note: In the field of telephony, AAs and Integrated Voice Response (IVR) systems are considered to be distinctly separate features of a phone system.
Bandwidth	Bandwidth is the maximum amount of data that can be sent, in one direction, between your device and the internet. It is commonly referred to as internet speed, connection speed, or line speed; and is represented as the amount of data transferred per second. For example, a 500 Mbps connection can send 500 megabits of data in one second.Bandwidth can be misleading for two reasons:It is not the same as throughput. Bandwidth rates are based on optimal conditions; therefore, any network problems will cause the actual throughput (data successfully transmitted per second) to be lower than the bandwidth.Internet connections are typically asymmetric, meaning they are configured with more download bandwidth (from the internet to you) than upload bandwidth.  For example, a 500 Mbps connection can download data at a rate of 500 Mbps, but may only be able to upload data at a rate of 100 Mbps.It’s important to consider the following factors are when determining VoIP bandwidth:VoIP requires the same amount of bandwidth in both directions.CoreDial uses G711 encoding, which transmits VoIP packets at a rate of 80 Kbps (0.64 Mbps).VoIP bandwidth requirements should be based on the peak number of concurrent calls.When adding VoIP to an existing internet connection, the bandwidth must be sufficient for concurrent, peak voice and data transmissions.
Barge	Short for “Barge In,” this call center feature allows an authorized user to listen in (hear both parties) during an active call. The call center agent is aware the call is being monitored, but the outside party is not.  Barge is typically used to monitor or train call center agents. CoreDial’s platform includes the following Barge options:Listen Live – the authorized user’s phone is muted while they are listening to a call.Barge – the authorized user can speak to the call center agent during a call. The outside party cannot hear the Barge conversation. This option is commonly referred to as “whisper.”
Broadband	Broadband refers to a general category of long distance, high-speed methods for sending information between devices. It is available on a variety of physical media such as fiber, FIOS, DSL, Wi-Fi, (coaxial) cable, cellular, satellite, etc; and each technology has its own pros and cons. Regardless of the medium, all broadband implementations share the following traits:The connection is always available or “always on.”The medium (e.g., wire) is divided into multiple pathways called channels or bands, allowing a single medium to simultaneously transmit different types of information (e.g., video and data) to different destinations.Information is downloaded (received by a device) at a minimum speed of 25 Mbps and uploaded (sent) at a minimum speed of 3 Mbps.Broadband is now the default term commonly used to describe any type of internet connection that isn’t dialup.
BTN (Billing Telephone Number)	This is a 10-digit number that the local phone company (ILEC or CLEC) uses to bill a customer for services.When carriers issue phone numbers to customers, they select one of the numbers to be the account billing number. This is referred to as the BTN, and it’s used to bill the customer for all services delivered to a specific location. For example, a customer might have five phone numbers (DIDs) for their business. One number is designated as the BTN, and services for all five numbers are billed under that BTN.  Customers with multiple locations may have different BTNs for each address.Contrary to popular belief, phone numbers are actually owned by the carriers who issue them versus the customers or businesses who use them. Regardless of whether a customer is using VoIP or non-VoIP phone service, there is always a carrier and a BTN associated with their account. The BTN is required in order to move (or port) a customer’s phone numbers onto CoreDial’s platform. If the customer doesn’t know their BTN, they can request it from their current phone service provider.
Bursting	Bursting is a general term that describes a sudden, and usually temporary, spike in the amount of data traveling over a specific connection. In VoIP, bursting occurs when a customer’s total call activity exceeds the combined capacity of all their connections (or call paths). For example, a customer with three prepaid call paths will reach maximum capacity when three users are simultaneously on calls (to/from external numbers). Bursting would occur if a fourth call arrived during that time.CoreDial’s Hosted PBX and SIP Trunking solutions can dynamically add more call paths as needed when bursting occurs. This ensures callers will not receive busy signals during peak activity times.
Busy Call Forwarding	This phone system feature automatically forwards an incoming call to another destination if the called number is busy or has DND turned on.In Account Manager, this is one of several Inbound Dialing Rules that can be configured for an extension. Calls can be sent to any routable destination such as another phone number or extension, ring group, voice mailbox, etc.

C

Term	Definition
CALEA	The Communications Assistance for Law Enforcement Act (CALEA) is a federal law that requires VoIP telecommunications service providers to cooperate with court ordered electronic phone surveillance operations. CoreDial Partners must file a CALEA SSI Plan with the FCC.  This is a document that provides contact information and identifies internal company procedures to be followed in the event an investigation is required.Additional information is available on the following websites:https://www.fcc.gov/public-safety-and-homeland-security/policy-and-licensing-division/general/communications-assistancehttps://ndcac.fbi.gov/calea/about-calea
Call Blocking	“Black list” phone numbers to block them from calling your Hosted PBX.
Call Control	Refers to the software within a telephone switch that supplies its central function.  Call control decodes addressing information and routes phone calls from one end point to another.  It also creates the features that can be used to adapt standard switch operation to the needs of users.  Examples include Call Waiting and Call Forward on Busy.
Call Forwarding	Forwards calls via the portal, or via your device or softphone.  Calls may be forwarded to any extension or phone number.Note: Device or softphone forwarding functionality may vary by manufacturer.
Call Hold	Place calls on hold and play music or a commercial while a caller is on hold.
Call Park	Unlike a call placed on hold, a parked call may be picked up from another extension.
Call Path	See Virtual Call Path
Call Queue	Call Queues are used to route calls on a first-in first-out basis to the appropriate extension or group. These extensions can be agents logged into the system.  Call queues are commonly used with an ACD, where callers hear an announcement such as “Thank you for calling, all available agents are busy, please hold for the next available agent, or press “1” to leave a message”.  When the call is ready to be routed, the ACD handles the routing rules.
Call Routing Time Frames	Allows routing decisions based on time and date.  Multiple schedules can be configured for departments with different hours of operation (e.g. business hours, after hours and holiday hours).
Caller ID	This refers to the process of displaying a caller’s 10 digit number on the recipient’s phone, while the phone is ringing.  This allows the recipient to identify the caller before answering the phone. By default, carriers transmit Caller ID with outbound calls; however, most phones allow users to modify the default behavior.  Caller ID is also referred to as Calling Line Number.Caller ID Name (CNAM) is an enhancement to Caller ID that displays the caller’s name along with the phone number.  The terms Caller ID, CNAM, and Calling Line ID are often used interchangeably.
Carrier	Short for Common Carrier, this refers to a company that sells phone services to the general public, and also owns the transport lines and equipment. Telephone service is a public utility, so carriers are regulated by the FCC and required to charge all customers the same price (for the same service).Following the Telecommunications Deregulation Act of 1996, carriers were categorized as either ILECs or CLECs.  Note: CoreDial is not a carrier.
Circuit Switched Network	In telephony, a circuit refers to single, dedicated path (usually a wire) between two points through which audio signals can travel in both directions. A switch is a device that receives a signal from one location and forwards (or switches) it to another location (either the final destination or another switch en route to the destination). A circuit switched network combines both technologies. It’s a group of geographically distributed switches connected together via backbone circuits, and end point devices such as phones are connected to the switches via local circuits (phone wires).  This allows a connection (or circuit) to be established between any two end points on the network regardless of their actual locations. The PSTN is a world-wide circuit switched network.
CLEC (Competitive Local Exchange Carrier)	CLECs are companies that have their own access to the PSTN and are able to sell phone services directly to consumers. They may own and operate regional networks (local loops), switches, and inter-switch connections; or they may lease some local or backbone PSTN components from ILECs.  CLEC customers are generally unaware of the underlying ILEC relationship.  Note: CoreDial is not a CLEC.
Click-to-Dial	Click-to-Dial is the ability to initiate a phone call from the contact list on your computer with the click of a mouse.
CNAM (Caller ID Name)	Caller ID Name (CNAM) is an enhancement to Caller ID that allows the caller’s name to be displayed along with the phone number.  CNAM databases are maintained by carriers (who own the numbers) or by third party services. When you receive a call, your phone service provider checks the incoming number against a CNAM database, retrieves the CNAM, and displays it with the Caller ID.  This process is referred to as “dipping” and there is often a fee associated with it. Although the PSTN restricts a CNAM to 15 characters, VoIP does not have this limitation. However, longer CNAMs will only be displayed for calls that originate and terminate on a VoIP network.  When a VoIP call terminates on the PSTN, the CNAM will be truncated after the first 15 characters (including spaces). For this reason, most CNAMs are still 15 characters or less.Note: CNAM has no correlation to CNAME, which refers to a DNS resource record.
Codec	Codec is an abbreviation of the words “compression” and “decompression.” The function of a VoIP codec is to convert an analog voice signal into a compressed digital signal that can be sent over the internet; and to reverse the process on the other end. The two standard VoIP codecs are G.711 uLaw and G.729a, and each has different properties. The choice of codec is essentially a trade-off between bandwidth utilization and voice quality.G.711 uLaw provides the highest quality: 200 bytes/packet, full duplex, 80kb/sec/call, and requires 800kb/sec bandwidth for every 10 concurrent calls.  CoreDial recommends you use this codec wherever possible.G.729a provides good quality, but not as high as G.711: 60 bytes/packet, full duplex, 30kb/sec/call, and requires 300kb/sec for every 10 concurrent calls.  CoreDial recommends you use this codec only as temporary solution when needed.
Concurrent Call Utilization	A concurrent call is defined as a call to/from somewhere on the CoreDial network to/from somewhere outside of the CoreDial network such as the PSTN or cellular network. While all calls made from a SIP device on CoreDial service will consume bandwidth on the local network (LAN), not all calls will consume concurrent call paths. It is important to make this distinction, as it is integral in the planning for network capacity and for contracted line count as both represent a cost to the end-user.
Converged Network	Network convergence (also called media convergence) refers to using a single network for multiple types of media such as voice and data. Converged networks are becoming more commonplace as technology improves and costs decrease; therefore, this module focuses on optimizing VoIP in converged networks.  Converged networks can share the same LAN infrastructure, WAN infrastructure, or both.  A common reason for switching from a dedicated to converged network is the lower cost of a single communications line.  With proper configuration and management, converged networks are highly reliable.SOHO (small office home office) networks with up to five VoIP phones are essentially very small converged networks.  Because of the relatively light traffic loads, VoIP performance is usually acceptable without changing the LAN configuration.  SOHO networks are also called flat networks.  Dedicated and SOHO networks are generally simpler to configure and manage, and use the same VoIP optimization techniques as converged networks.
CPNI	CPNI, or Customer Proprietary Network Information, is the data collected about an individual users calls, such as time, date, duration and destination number of each call.
Customer Service Record (CSR)	A CSR is a record of the customer’s account information as stored in their Telecommunication Service Provider’s (or carrier’s) database.  CSRs are held by carriers and provided to customers upon request. A CSR includes the service location, billing address, billing telephone number, account number, and authorized contact information for the account, which may be different from the information printed on the customer’s invoice.  CSRs are often used when porting numbers and are also known as Internal Account Records, Service Records, and Billing Services Records. A customer’s CSR is the most accurate source of porting information and helps ensure a fast, error free process.  CSRs are not required for porting toll free numbers. Instead, the customer’s most recent invoice is used.

D-F

Term	Definition
Dedicated Network	Traditionally, different types of media such as voice and data had their own physical (combined LAN/WAN) infrastructures, known as dedicated networks. For example, voice networks consisted of handsets, prem based PBXs, and a connection to the PSTN.  Data networks included computers, switches, routers, firewalls, and a connection to the internet.Dedicated networks are highly reliable and can provide guaranteed WAN access speeds.  Dedicated LANs/WANs provide the best possible VoIP service. They are easier to install, manage, and troubleshoot but are also more expensive to operate.
Digital Signal Processor (DSP)	A DSP processes a digital signal converted from an analog signal.  VoIP DSPs are are application specific and convert analog voice to digital voice over internet protocol (IP).
Direct Inward Dial (DID)	DID is used for call routing. Through DID, external callers are able to contact a user directly at their unique phone number.  Set up a telephone number to dial directly to a device or extension.
Direct Inward System Access (DISA)	Allows remote users to dial into their Hosted PBX from an outside line and make outbound calls that will display Caller ID information from a DID within their office.
Directed Call Pickup	Allows users to dial *8 plus an extension number to answer a call ringing at that extension.Note: This feature does not work with a Group or queue call, only direct extensions.
Do Not Disturb	A device or softphone feature that simulates a phone being off-the-hook and sends incoming calls directly into voicemail.  Other routing options are also available.
Dynamic Host Configuration Protocol (DHCP)	DHCP is a network service that automatically provides an internet protocol (IP) host with its network address and other related configuration information such as the subnet mask and default gateway as a person shifts from one network to another.Benefits: The DHCP server minimizes configuration errors caused by manual IP address configuration, such as typographical errors or address conflicts caused by the assignment of an IP address to more than one computer at the same time.  It also reduces network administration by i.e. efficient handling of IP address changes for customers that must be updated frequently, such as those for portable computers that move to different locations on a wireless network. DHCP allows the customer to move their device from one place to another without needing to make any configuration changes.
E911	The term “E911” refers to the legal requirement for ITSPs (Internet Telephony Service Providers) to ensure that dialing 911 (from any device) will connect to the appropriate emergency response center (PSAP) and automatically transmit the caller’s address and phone number.E911 laws are imposed at the state and local levels and vary by region. It is the ITSP’s responsibility to identify E911 laws and configure the Account Manager E911 Location feature correctly for each Customer Account.
Endpoint	An endpoint is an IP telephone, softphone, or analog telephone adapter device. Every device (used for making calls) must have an endpoint configured in the Hosted PBX.  Every endpoint must have at least one extension assigned however a single endpoint can have multiple extensions.
Extension: Cloud	A cloud extension (i.e. a voice mailbox) is virtual meaning it is not associated with a physical endpoint.  You cannot make an outbound call from a cloud extension since it is virtual.  A cloud extension forwards (or routes) incoming calls to other extensions. Cloud extensions are great for external workers at a company whereby calls to their corporate cloud extension can be forwarded to their outside number, i.e. a personal cellphone using a feature like Find me, follow me.Note: Cloud extensions cannot be associated with an endpoint.
Extension: Standard	A standard extension is an individual user account associated with a physical endpoint by a two to six digit number.  A standard extension is associated with an endpoint, and the endpoint is associated with a device.  The extension provides a pathway to the endpoint and its device.  Standard extensions on Hosted PBXs are also referred to as SIP extensions.  You can receive incoming calls and make outbound calls on a standard extension.
Find me, follow me	Find me, follow me is generally used as a call-forwarding feature. It improves worker productivity and customer service by ensuring that every call reaches the right person, regardless of where he or she is working.  The Find Me feature attempts to locate you by dialing up to 5 locations until you either accept or reject the call.
Firewall	A firewall is a key security features that sits between two networks, such as a company’s internal network and the Internet.  Firewalls prevents unauthorized people from accessing the internal network.
Firm Order Completion (FOC) Date	The FOC Date is the date when the local carrier will commit to transferring the customers existing phone number to another carrier.
FXS and FXO	These are the names given to ports that are used by analog phone lines (also known as POTS- Plain Old Telephone Service) or phones.

G-I

Term	Definition
Graphical User Interface (GUI)	GUIs allow users to interact with computers through graphical icons and visual indicators as opposed to text-based (command line) interfaces.
Hop	Data packets pass through bridges, routers and gateways along the way.  Each time a packet is passed to the next device, a hop occurs.
Hosted Fax	This feature allows you to send faxes from your computer (as a PDF attachment), which can then be received via email or routed to a physical fax device.  This feature provides immediate access to faxes- anywhere, anytime, and from any device.
Hosted PBX	This is a “software-based” PBX service that is available through the internet. It’s an alternative to a traditional “premise based” PBX and provides the same functions (e.g., auto attendants, voice mail, extensions, etc.). A Hosted PBX requires VoIP phones. These are connected to the internet, and all telephone services are delivered through the internet. The end user customer manages the Hosted PBX through a GUI that is similar to the prem-based PBX screens. They pay an ISP for internet access and CoreDial’s Partners (ITSPs) for Hosted PBX access and VoIP phone services. CoreDial manages the call flow between the Hosted PBX and the PSTN.A Hosted PBX is commonly referred to as a Virtual PBX or Hosted VoIP.
ILEC (Incumbent Local Exchange Carrier)	ILECs are what the original telephone monopoly companies became after the Telecommunications Deregulation Act of 1996.Prior to the Act, these companies were the sole providers of phone service, and they owned and controlled the PSTN infrastructure throughout the US. After the Act, the original companies were divided into regional companies (ILECs) and required to lease PSTN components (e.g., backbone lines) to newly formed, competitor phone companies called CLECs.
Incoming Privacy Screening	Force callers with “No Caller ID” or “Blocked Caller ID” to enter a number that will be presented as their Caller ID.
Integrated Access Device (IAD)	This is a customer premise device that provides access to WANs and the internet.  It aggregates multiple channels of information such as voice and data across a shared access link to a service provider.  Examples of an access link include a T1 line or a DSL connection.  The IAD is installed by the service provider on the customer’s premises and thus allows the service provider to control and manage the features and operation of the access link.
IP Address	An IP Address is used to uniquely identify devices on a network and are classified as either ‘public’ or ‘private’ IP addresses. The customer’s ISP will typically distribute one or more public IP address to the customer and the public IP will be assigned to the WAN interface of the customer’s router or modem/router.
IP Phones	IP phones plug directly into the network and perform analog-to-digital and/or digital-to-analog conversions.
ISP (Internet Service Provider)	An ISP is the company you purchase your internet access from.  Most IPSs provide additional services such web site hosting, etc. As the internet has grown, the industry has steadily consolidated, and is now dominated by large regional or national ISPs such as Comcast, Verizon, etc.
ITSP (Internet Telephony Service Provider)	ITSP is the standard industry designation for businesses that sell VoIP services. CoreDial Partners are ITSPs. NOTE: The FCC’s definition of ITSP is Interstate Telecommunications Service Providers, which refers to companies that sell multi-state phone services via the PSTN.  The FCC categorizes businesses that sell VoIP services (e.g., CoreDial Partners) as Interconnected VoIP Providers.

J-L

Term	Definition
Jitter	Jitter is a measure of the variation in packet arrival times from phone to server.  Jitter is generally caused by network congestion, which is created by delay, variation or timing issues when the packets arrive. The result is poor and/or unacceptable voice quality. Jitter often manifests itself as a “delayed response” from one caller to the other caller or users sounding like they are talking over each other. Sometimes jitter could sound like “squawking” noises, if packets are played out of order.  In order to maintain quality voice communications, try to look for Jitter measurements of less than 5ms.  The best solution for jitter is to increase the capacity of the congested device or link.
Latency	Latency is the time it takes for a caller’s voice to be transported – packetized (see packet), sent over the network, depacketized and replayed – to the other person. Too much latency is bad, making for a disjointed conversation flow. Ideally, latency should not exceed 150 milliseconds (one-way). Geographical distance (i.e. wireless) or a lower-speed network connection can cause latency issues to get worse.
Letter Of Authorization (LOA)	An LOA is a document authorizing a telecommunications provider to act on the customer’s behalf.  If porting a toll free number, a RespOrg LOA is used instead of a regular LOA.
Listen Live	Allows you to listen in on a selected extension, but not to speak.
Live Person Answering	Set up a telephone number to ring a specific extension or a Ring Group – sequentially or simultaneously.  This option enables your company to use a live person to answer the caller instead of an Auto Attendant.
Local Number Port (LNP)	LNP is the process of transferring standard telephone numbers (or DIDs) from one carrier to another.  This allows customers to keep their existing phone number(s) when they switch telecommunications service providers.  It is commonly known as Porting or LNP.

M-O

Term	Definition
Managed IP Telephony Services	Managed IP Telephony Services is another phrase for “Hosted” services. Typically, the end-customer business customer owns the IP PBX and related equipment, while the carrier or VAR provides management and maintenance for the phone system.
Mean Opinion Score (MOS)	Mean Opinion Score is a test that has been used in telephony networks for decades to obtain a user’s view of quality on the network.  In measuring VoIP, it is a calculation based on performance over the IP network in which it is carried.  The range is 1 to 5, where 1 is lowest perceived audio quality and 5 is the highest.  A typical range for VoIP would be 3.5 to 4.2.  Use an Edge Device that collects MOS data.  Common Edge Device brands are Adtran, Cisco, Edgemarc, Peplink, and Sonic Wall.Note that scores are an aggregate of many factors and must be read in the correct context.  For example, a two hour call with poor quality during the last few minutes will have a high MOS value.  These devices also track additional performance factors.
Modem	A modem is provided with services such as cable Internet and DSL and is the DMARC between the ISP and the customer’s internal network. An Ethernet handoff is typically provided from the modem to the customer. In some cases the modem provided by the ISP actually contains router and/or firewall functionality as described below. When a separate router/firewall is to be used by the customer, the modem must be set to “bridge mode” to avoid issues with double NAT (Network Address Translation) that can negatively impact voice traffic.
Multi Protocol Label Switching (MPLS)	This is a virtual, private network.  It is a protocol that directs and carries data through high-performance networks from one network node to the next by using labels.  Think of it as a cloud service whereby when your data leaves your location and enters the service provider’s network, it’s given a label that tells it where to go and what path it needs to take in order to reach its intended endpoint.
Multicast Paging	Dial a Ring Group and make an announcement through the loudspeaker of each phone in the group. In most multi-cast paging environments, administrators will specify a single multi-cast paging address in the IP PBX setup. They will then configure each paging endpoint such as an IP Phone, to “listen in” on that multicast address.In a multicast scenario, when a user initiates a page, the page originates from the IP PBX, and the PBX only sets up a single SIP and RTP audio path to the multicast paging addresses. The IP Phones are always “listening” to this address and when RTP packets or audio is heard, the phone or paging endpoints play that audio stream.Note: this feature is phone specific; check per brand or model.
Music On Hold (Custom or Default)	Gives you the choice to upload custom music files, or use CoreDial’s default music to be played while callers are on hold.
Network Access Translation (NAT)	NAT is the mechanism that allows you to have many computers on your LAN all connected to the internet through a single external IP address.  You can now get a range of devices sold as Firewalls or Routers that combine Firewall, Router and NAT as a device all in the one box.
No Answer Call Forwarding	Automatically forwards your calls to an extension, group, or phone number when you do not answer your phone.
Office Intercom	Dial another user’s extension and activate their phone speaker to make an announcement.Note: this feature is phone specific; check per brand or model.
On-Demand Call Paths	Also known as burstable Virtual Call Paths, this feature allows you to provide your customers with a call path “buffer.”  When the system detects bursting, it will temporarily assign more call paths to the customer in order to handle the increased activity. These extra paths are named “On-Demand” because they are only triggered when all prepaid call paths are in use. If On-Demand Call Paths are not added when bursting occurs, then inbound callers will receive a busy signal. On-Demand Call Paths can be used for Hosted PBX and SIP Trunk installations and are configured on a per customer basis. Each customer can have up to 10 On-Demand Call Paths for voice calls (including conference calls) and for T.38 Fax calls. There are two system settings for this feature:Quantity of On-Demand Call Paths – These settings are located in the services section of customer’s Settings tab.  You will be need to set the number of On-Demand Call Paths for each new customer when creating their account for the first time.Price per On-Demand Call Paths – For voice, the price is set at the item level, and the same price will be used for all customers that use that call path item.  If you need multiple On-Demand price points, then create additional items with different On-Demand prices.
Option 66	Option 66 (also known as DHCP Option 66) allows you to provision IP phones on a mass scale as opposed to individually.  It enables you to complete “zero-touch” provisioning through a web interface. Option 66 automates the task of manually inserting the provisioning link in each IP phone.  By using Option 66 you are less inclined to make mistakes and your implementation will be faster.
Origination	Origination refers to inbound calls or minutes from the PSTN.
Outbound Dialing Rules	Configure which types of outbound calls users can make, i.e. disable international calling or prevent calls to specific numbers or services.

P-R

Term	Definition
Packet	A packet is a small chunk of data that has a destination address and other information attached to it. In a packet switched network such as the internet, all data is first divided into packets, and then sent over the network to the destination address. When the packets arrive at the destination, they are collected and reassembled back into the original block of data.
Packet Loss	Packet loss occurs when one or more packets of data fail to reach their destination (when they are lost by a network device), resulting in a metallic sound or conversation dropouts. It can be caused by network congestion (overload), distance, and/or poor line quality. It starts to become audible then the loss rate approaches 30 packets/second.  Eliminating packet loss on a network is crucial for quality VoIP communication.  As a guide, VoIP communication will generally provide acceptable voice quality on networks with up to 0.75% packet loss.
PBX (Private Branch Exchange)	A PBX is a system that automates the process of switching (or routing) calls to destinations within a business. For example, when you call a company’s main number, the call goes to a PBX instead of directly to a desk phone. The PBX then forwards (or switches) the call to the correct desk phone within the company.  PBXs allow one phone number (or DID) to be shared among multiple in-house phones. All calls made to and from a company’s phones are processed by the PBX.  In addition to routing, PBXs provide a range of telephony features such as auto attendant, voice mail, call recording, extension-to-extension calls and transfers, call hold, conference calls, activity reports, presence status, etc.The most common types of PBXs are:Premise (Prem) based PBX – a device located on the company premises that has at least one one external port (e.g., PRI) that connects to a phone service provider (carrier). The in-house phones are connected directly to the premise based PBX.IP PBX – an on-premise device that has at least one one external port (e.g., PRI) that connects to a phone service provider (carrier) and at least one internal connection to a LAN (local area network).  Only IP phones can be used with an IP PBX, and they connect to the LAN versus connecting directly to the PBX.  IP PBXs typically provide more features than standard premise based PBXs.Hosted PBX – A “virtual” (software only) version of an IP PBX.  It’s located on a remote VoIP provider’s server and accessed via the internet, so there is no on-premise device.  Only IP phones can be used with a Hosted PBX, and they connect directly to the LAN and use the company internet connection to reach the PBX.  The Hosted PBX is managed through a software interface from a computer on the LAN.
Post Dial Delay (PDD)	PDD is the interval between dialing the last digit of the called number and hearing the ringback tone.
POTS (Plain Old Telephone Service)	POTS refers to traditional phone service such as a basic household phone line. Service is provided by a local phone company and travels over a local connection (local loop) to and from the PSTN.The terms analog, landline, POTS, and PSTN are often used interchangeably.
Premise based PBX	Short for Premise Based Private Branch Exchange, this is an onsite, hardware based telephony system that automates the calling processes for a business. It provides centralized management of phones, extensions, numbers, voicemail, etc.; and it also allows a single number (DID) from the phone company to be shared between multiple in-house phones.  All calls made to and from a company’s in-house phones are first sent to the PBX, which then sends the call to its destination.There are two types of premise based PBXs: Standard and IP.  Both have at least one one external port (e.g., PRI) that connects to a phone service provider (carrier).  The difference is in the in-house phone connections. Standard PBX has multiple ports, and on-site phones are connected directly to these ports.IP PBX has one (or more) ethernet ports and is connected directly to the company LAN.  On-site devices must be IP phones, which also connect to the company LAN. IP PBXs have more features than standard PBXs.
Presence Status	Presence status, or just presence for short, is the ability to see a colleague’s presence status – whether they’re in the office or away from their desk.  This can be signified, for example, by a green light appearing beside a user’s name in a chat bar when they are online, or a red light when they are busy.
Primary Rate Interface (PRI)	PRI provides a ways of connecting your Prem based PBX to the PSTN.  It is a refers to a telecommunications standard that specifies a dedicated physical connection to the PSTN that only transmits voice.  That physical line provides 23 voice channels (allowing your business to conduct up to 23 voice calls simultaneously) per line.
PSTN (Public Switched Telephone Network)	The PSTN is the worldwide telecommunications infrastructure. It’s a collection of interconnected, regional networks through which two devices, located anywhere in the world, can be connected for the purpose of transmitting audio.The most visible PSTN components are the telephone wires along roads which terminate at buildings (also called the local loop or last mile). At the other end, these wires connect to phone company switching stations, which are connected to each other through a series of high-capacity, fiber optic backbone lines. This allows a direct circuit (or 1:1 connection) to be established between any two devices on the PSTN.The PSTN was originally built with analog technology, but is now mostly digital. The terms POTS, analog, and landline are still used to describe a standard (local loop) connection to the PSTN such as a traditional household phone line.
Quality of service (QoS)	QoS is a measure of the overall performance of the network. It takes into consideration factors such as error rates, bandwidth, throughput, transmission delay, availability, and jitter.
Real-Time Transport Protocol (RTP)	Also called RTTP.  RTP  is the actual voice traffic that is transmitted once the call is set up between the two endpoints. Many calls can be going on at the same time from one site to another so a wide range of ports are typically available for RTP traffic so that each call can be designated a unique port. While a smaller subset of ports is typically needed, UDP port range 10000 – 20000 are earmarked for RTP traffic and can be used when establishing rules to allow and prioritize VoIP traffic as needed.  Quite simply, the RTP is an Internet protocol standard that specifies a way for programs to manage the real-time transmission of multimedia data over either unicast or multicast network services in two steps:The voice data stream is divided into 20 ms packets (50 packets/second).The packets are compressed using a compression/decompression algorithm (codec).
RespOrg	RespOrg stands for Responsible Organization,” the process of transferring “Toll Free” telephone numbers from one carrier to another.  A RespOrg form is required in order to submit a request to port a customer’s toll free number(s).
Ring Groups	Enable multiple extensions to be joined as a group, and then route calls sequentially or simultaneously to that group.
Router	A router is a device that routes packets between various IP subnets/networks. A router makes the decision to route traffic based on its routing table which is populated either via static routes which are provided manually during configuration or via routing protocols which automatically exchange route information between routers based on network configuration and status. Most routers that will be deployed by your customers will likely use static route and will simply route between the ISP on the WAN side of the router and the customer’s internal network on the LAN side of the router.

S

Term	Definition
Screen Pop	A small pop-up window that displays inbound Caller ID information on the user’s screen.
Security Gateway	This enables secure termination for multiple VoIP handsets, IP phones and softphones by providing such things as Encryption, Dynamic Session Security and Bandwidth management for users.
Service Level Agreement (SLA)	The SLA is the part of a service contract that defines the level of service for voice quality.  SLAs typically includes call completion rate, PDD, and some measure of voice quality.
Service Provider	A company that purchases telecommunications services from a Telecommunications Service Provider (TSP) and resells it as their own (standalone or integrated) product offering.  SPs are commonly known as Managed Service Providers (MSPs), ITSPs and Resellers.  CoreDial Partners belong to this category.
Session Border Controller (SBC)	SBCs are network devices used to register, setup, control, and tear down VoIP multimedia communications sessions. An SBC ensures that only approved traffic passes into the heart of your business. An SBC also hides your internal network and your users’ IP addresses from the outside world, providing additional security protection.
Shared Line Appearance (SLA)	This is a telephony feature that makes two or more phones behave as though they are linked, in the sense that a call to one number causes all phones to ring.  A Busy Lamp Field (BLF) appears on the user’s phone when one or more of the lines are in use.
Short Message Service (SMS)	SMS is a text messaging service component of communication systems.
SIP (Session Initiation Protocol)	SIP is one of the VoIP protocols.  The role of SIP (in a VoIP call) is to establish a connection between the sender and receiver.  This is a fixed route that simulates a circuit switched connection because it can only be used by the sender and receiver.  Once the SIP connection is in place, the other VoIP protocols will use it to send audio between both parties. When the call is over, SIP disconnects the session. A SIP connection is often referred to as a virtual path.Analogy: Think of SIP as a highway road crew – they make sure the ramps are opened and the highway is in good repair, so vehicles can enter, drive, and exit. They are not concerned with the types of vehicles or their contents. Likewise, SIP creates a pathway (connection) for calls but does not interact with the contents (voice packets) of the call.
SIP Trunking	SIP trunking is an economical alternative for delivering phone service to any type of premise based PBX. Traditionally, a high capacity line (or trunk) such as a T1 connected the PBX to a carrier for PSTN access.  In a SIP trunk implementation, the PBX is connected to a VoIP provider through an internet connection.  The VoIP provider (CoreDial) manages the call delivery between the PSTN and the PBX.SIP trunks do not require any changes to the on-site phones or phone-to-PBX connections. Users can continue to make and receive calls the same way, regardless of the trunk connection.
Softphone	A softphone is voice software that emulates a VoIP telephone on a computer, smartphone, or tablet.
Softswitch	This is a central device in a telephone network that connects calls from one phone line to another through software running on a computer system.  It is predominantly used to control connections at a junction point between circuit and packet networks.
SSL (Secure Sockets Layer) Certificate	SSL Certificates are small data files that digitally bind a cryptographic key to an organization’s details.  SSL Certificates secure a website that transmits personal information.  SSL creates an encrypted connection between an organization’s web server and a user’s web browser to allow a secure connection for any personal or confidential information to be transmitted.
Switch	Switches are generally layer 2 devices that are deployed in Local Area Network (LAN) environments and used to aggregate device connections on the LAN. Switches can be used to segment Ethernet networks and connect a large number of devices on a Layer 2 network to a layer 3 network which is typically represented by the use of a router. While some higher end switches can act as a router, most switches you will work with do not make IP routing decisions or perform firewall functions such as filtering and NAT.

T-U

Term	Definition
T.38 Fax	This is a form of sending fax messages over an IP network from a traditional analog physical fax device. An ATA (Analog Telephone Adapter) is required to connect one or more standard analog phones to a digital and/or non-standard telephone system such as a VoIP network. The fax is transmitted in real-time.  In order to sell the T.38 fax service three things are required: a T.30 Fax Extension, T.38 Fax Call Path and a T.38 Fax DID Number.Alert: While CoreDial uses the T.38 standard for faxing and adheres to the FCC guidelines regarding its use, T.38 fax services are provided on a best-effort basis and cannot be guaranteed on all end to end connections.
T1 Line	This is a fiber optic line that that is point-to-point from your business network to the telephone company’s central office. It is a high-performance line and can carry about 60 times more data than a regular residential modem.
TDM (time division multiplexing) PBX	TDM PBXs do not have any network connections. Analog or digital end user devices connect directly to the PBX, and an analog or PRI trunk interface connects to the PSTN. They are also called Legacy, Traditional, Non-IP, or Analog/Digital PBXs.  TDM PBXs require a VoIP Gateway to connect to a SIP trunk.  VoIP Gateway is a common term and can have multiple meanings.  CoreDial’s definition is: a device (on the premise LAN) that translates IP packets into a format readable by a PBX.  VoIP Gateways are also called Media, SIP, and SIP-ISDN Gateways.
Tentative Order Completion (TOC) Date	The TOC date is the date that you are requesting from the local carrier for transferring the customers existing phone number to another carrier.
Termination	Termination refers to outbound calls or minutes from the PSTN.
Tiers (networks)	There is no authority that defines the tiers of networks participating in the internet however the following are common descriptions.Tier 1 is a network that can reach every other network on the internet without needing to purchase IP transit or paying settlements.Tier 2 is commonly described as a network that peers with some networks, but still purchases IP transit or pays settlements to reach at least some portion of the internet.Tier 3 would be a network that solely purchases transit from other networks in order to reach the internet.
Trunk	A trunk refers to a group of individual phone lines that is consolidated into a single, high capacity line.  The name is literally based on “tree trunk” because it resembles to the way that tree trunks consolidate pathways between the branches and roots.Telephony trunks convert the analog signals from individual phone lines into digital signals that can be transmitted more efficiently over long distances. They are also referred to as Trunk Lines.
Unattended Transfer	Transfer a call to another extension, group, or phone number without announcing the party being transferred.
Unified Communications (UC)	UC is the seamless integration of voice, presence, chat, video, data, applications, and other technologies to improve communications, processes, and business productivity.
User Datagram Protocol (UDP)	UDP is a transport protocol.  Once assembled, RTP packets are transported with UDP instead of TCP.  UDP is optimal for real-time, streaming applications, where interruptions to data flow are more detrimental than occasional data loss. UDP is connectionless meaning that packets are continually sent without delivery confirmation. Lost packets are not recovered or retransmitted and intermittent packet loss is not audible because each packet contains only 1/50th of a second of sound.TCP is better suited for user applications where data accuracy is more important than delivery speed. TCP is connection-oriented meaning that delivery confirmation is required for packets; lost packets are recovered and re-transmitted; and the order of packets is guaranteed.

V-Z

Term	Definition
Virtual LANs (VLANs)	VLANs are used to “logically separate” devices and departments on the same Ethernet wire. VLANs are becoming increasingly more common as switch technology advances. With traditional LAN switching, a physical switch could only belong to one LAN segment. With VLANs, a physical switch can contain multiple logical LAN segments.  Each VLAN has its own broadcast domain and isolates network traffic the same way that physical switches do.
VLAN Switch	A physical switch that can emulate two (or more) physical switches. Each “emulated” switch behaves as an independent LAN segment (or virtual LAN). VLAN switches include access ports and trunk ports.  Access ports connect to end user devices. Trunk ports connect to other switches or routers.  Some VLAN switches have integrated (Layer 3) routers.  These routers are designed for VLAN to VLAN connections, and should not be used for VoIP internet access.
VLAN trunks	The connections between trunk ports.  VLAN trunks are not related to SIP trunks or PSTN trunks.
Virtual Call Path	Virtual Call Paths deliver calls to and from the PSTN to either hosted endpoints or SIP Trunks.  They are shared by all endpoints on a Hosted PBX, regardless of location.  Customers can purchase any number of call paths to suit their needs and are limited only by budget constraints and bandwidth consumption (typically 80 Kbps per call).  On the CoreDial platform, call paths are defined as “concurrent calls”, meaning, the number of call paths required by a customer equals the number of concurrent calls that customer expects to have at any given time.
Virtual Private Network (VPN)	A VPN is a private network extended across a public network, i.e. the Internet. It enables the sharing of data across public networks (such as WiFi hotspots) in a typically more secure and functional way.  Privacy is increased with a VPN because the user’s initial IP address is replaced with one from the VPN provider.
VoIP (Voice over Internet Protocol)	VoIP refers to the process of making telephone calls over the internet and is an alternative to using traditional phone service (or the PSTN).  VoIP is often referred to as Internet Telephony or IP Telephony.Just like computers using the web, VoIP requires that phones have a direct internet connection. However, VoIP uses a different set of protocols to transmit voice than computers use to transmit data. Even though VoIP is generally referred to as a single entity, it’s actually a collection of several protocols: SIP, RTP, UDP, and IP.VoIP calls made through the internet must be able to reach non-VoIP numbers on the PSTN and vice versa. This function is provided by a VoIP Service Provider. VoIP calls first go to the Service Provider’s site and are then routed to the correct destination (on the PSTN or internet).
WebRTC	WebRTC stands for Web Real Time Communications. It’s an emerging standard that that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs.
Zero Out	Allows a caller to leave the queue by pressing “0” to choose other call routing options, such as voicemail.