VoIP Glossary

VOIP Glossary

If you are new to this IP system and cloud telephony, this is all you need to know to help your business grow better

A

Audio Teleconferencing or Audio Conferencing

Today’s PC-based systems allow for the use of conference bridges. You can set up, attend, and manage your own conferences over any touch-tone telephone by installing a conference server on your voice networks. Users can also schedule meetings using desktop software from their e-mail systems or a web browser.

Audio Response Unit (ARU)

A computer telephony system that uses voice storage and forwarding technology. ARUs can be passive or interactive. Passive ARUs simply play messages, whereas interactive ARUs play messages based on caller input.

Audio Menu

A verbal option provided by a phone recording. Audio selection menus are common in automated attendant, interactive voice response, and fax-on-demand systems. They are caller input prompts. Audio menus can direct you to speak commands or press keys on a touch-tone keypad.

Asynchronous Communication

A form of data transmission in which bits are transmitted without the use of a synchronising clock signal. Instead, each character is delivered with a start and stop bit surrounding it to indicate the beginning and end of the information. This is in contrast to synchronous communication, when data is delivered in blocks utilising a synchronising clock.

C

Conference Bridge

A device that allows multiple parties to communicate over the phone. Conference bridges can be manned by a proctor or operator, or they can be supervised. Some PBXs (Private Branch Exchange) include standalone conference bridges as well as conference bridge functions These systems include circuitry that sums and balances the energy (noise) on each channel so that everyone can hear each other. More sophisticated conference bridges can “idle” the transmit side of non-speaking parties’ channels.

D

Digital Subscriber Line (DSL)

A high-speed digital switched service that connects subscriber CPE (customer premises equipment) to the CO via existing copper pairs (central office). DSL handles more data downstream (data flowing towards the subscriber) than upstream (data flowing away from the subscriber) (flowing towards the network).

E

E-1

The designation for the 2.048 Mbps ITU standard for Europe’s 30-channel digital telephone service. It is the European version of T-1 (DS-1). The bandwidth is divided into 2 signaling channels (channels 15 and 31 starting from 0) and 30 bearer channels (voice channels). A&B bit signaling (robbed bit signaling) is not used here. E-1 uses one of the control channels for signaling and the other for clock synchronization.

F

Full Duplex

Full duplex refers to the ability of both ends of a communication to send and receive information at the same time without degrading the quality of the content.

Fax Server

A fax machine that is computer-based. Fax servers are “shared use” devices that are typically installed on a local area network (LAN). Clients on the LAN can access the fax server from their PCs in the same way that they access a network-based (shared) printer. Users can generate faxes at their workstations and have them “printed” to the fax server for transmission. Fax servers, similarly, can route incoming faxes to printers, file server directories, or individual users. Fax servers eliminate the need for users to print documents, transport them to the fax machine, and then wait for them to be transmitted after creating a cover page.

Frame Relay

Frame Relay is a packet switching method used in data communications that uses available bandwidth only when it is required. With proper network management, this fast packet switching method is efficient enough to transmit voice communications.

I

Interactive Voice Response (IVR)

IVR is a horizontal application in computer telephony that allows users to get computer-based information over the phone instead of utilising a computer. After listening to an audio menu, an IVR platform uses computer telephony components to translate callers’ touch-tones or voice commands into computer questions. “Please enter your account number using the touch-tones on your telephone,” for example. The IVR platform then “fetch” these inquiries from the host computer. In some instances, the data is stored on the same platform (self-hosted). The data is transformed into voice commands that are delivered to the caller over the phone.

Internet Service Provider (ISP)

A company that provides Internet access via subscription. Individuals or businesses can become subscribers. According to Jack Rickard, publisher of Boardwatch Magazine, ISPs operate on the Internet’s fourth or lowest level. Regional providers aggregate traffic from lower-order ISPs to the second, backbone level at the third level. The NAP (Network Access Point) is the highest level in North America, serving as peer-to-peer interconnection points for the largest backbones. There are three “official” NAPs: one in San Francisco, one in Chicago, and one in Pennsauken, New Jersey. ISPs use Internet routers, servers, and Rrack-mounted modems to provide a variety of services such as website hosting, FTP service, e-mail accounts, unified messaging, audio and video broadcasting, and, in some cases, Internet telephony and fax gateway services.

Internet Telephony

Any method of transmitting human voice over the Internet in real-time or near real-time. There are several parts: 1) A multimedia-equipped PC with special client software will digitise your voice on the client side. This can be accomplished using a voice modem or another method of voice encoding. 2) A direct or dial-up Internet connection allows your voice to be transmitted in packet form to its destination; 3) Connection with the far side is accomplished through IP address search, common servers, or beacons to identify the called party (and “ring” that person’s phone); 4) A similar arrangement on the far end completes the call and allows both parties to speak. PSTN/Internet gateways enable regular phone callers to make phone-to-Internet-to-phone connections. There are both PC-to-phone and phone-to-PC connections.

IP PBX

A VoIP telephone switching system is an Internet Protocol private branch exchange (IPPBX). An IP PBX can switch between VoIP and traditional phone lines, as well as between two traditional phone users. An IP PBX, as opposed to a traditional PBX, eliminates the need for separate voice and data networks.

IP telephony

Internet Protocol telephony is a system in which voice information is transmitted over packet-switched IP networks rather than traditional, dedicated circuit-switched connections. Voice, fax, and video data are transmitted as data packets over packet-switched connections that avoid the tolls of traditional phone lines. VoIP is the standardised version of IP telephony, and the two terms are used interchangeably.

M

Messaging

Any method of storing and forwarding messages in computer telephony. Fax mail, voice mail, and broadcast messaging are all examples of this. The most popular voice solution is this horizontal application. Messaging systems allow for the storage and transmission of “nonreal time” communication. A recorded voice message, for example, can be stored for later playback either locally or remotely, and a fax can be received and stored before being re-transmitted to the intended recipient. Messages can vary in content and media type, with the difference being that they are recorded or saved for later retrieval.

Modem

A modem is a device that converts digital signals to analogue signals and vice versa. Modems are used to send data signals (digital) over an analogue telephone network. A modem converts binary signals into tones that can be transmitted over a phone line. The modem’s demodulator converts the tones to binary code at the other end.

P

Public Switched Telephone Network (PSTN)

The PSTN is the world’s combined public circuit-switched telephone networks. Originally comprised of analogue telephone systems, the PSTN is now almost entirely digital and includes mobile telephones in addition to fixed telephones.

R

Real-Time

Real-time communications occur when perceptible delays between the sender and receiver are minimal and easily tolerated. Regular phone calls are made in real time. Point-to-point fax transmissions are nearly instantaneous. Voice messaging does not operate in real time.

S

Store and Forward

The process of archiving a message or transmission for later playback or transmission. In contrast to real-time communication, store and forward is the foundation of all messaging systems, such as email, fax-on-demand, unified messaging, and so on. The term “store and forward” in data communications refers to the temporary buffering of packets or other data strings.

Speech Recognition

Speech recognition is a technology that allows callers to speak commands to control applications.

SIP

The Session Initiation Protocol is used to initiate and terminate voice, video, and messaging communications between two or more IP network endpoints. SIP determines the location of endpoints, whether those endpoints are available, and the capabilities of each endpoint. SIP does not provide the actual means of communication between endpoints; rather, it facilitates communication sessions.

Service Provider

A company that provides services to Internet, telephone, and mobile phone users.

T

Telephony

The process of converting or transmitting voice or other signals over a long distance and then re-converting them to audible sound at the other end is referred to as telephony.

Transmission Control Protocol (TCP)

The ARPAnet transport layer protocol, which includes layers 4 and 5 of the OSI model. TCP manages TCP/IP sequential data exchange for remote hosts in a peer-to-peer network.

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User Login

Instantaneously receive user login credentials via email once the extension has been deployed by the adminstrator.

Log in conveniently via any web browser for access to the user portal.

Integrations

Integrate various CRM software for better customer service and satisfaction.

Enable transcription of audio messages into text with integration of third-party transcription service.

Enable API for integration with various third-party integrations for advanced performance such as call center and hotel reception services.

System

Provision email SMTP and template settings for event notifications to be sent out.

Manage storage allocation and duration of various types of data being stored on the PBX.

Determine the types of event notifications to be sent out, and the parties receiving them.

IVR

Add various audio voice prompts to be played according to different call scenarios.

Add voice prompts on multiple levels for more effecient and precise inbound call routing.

Voicemail

Customize default voicemail settings for each user extension.

Add various voicemail messages for allocation to different call scenarios and users.

Provision specific rules to how voicemail messages are played over the PBX system for better caller experience.

Reports and CDR

All call log activities are stored, with the option of having audio conversations recorded by default or for specific extensions.

Use intuitive visual call reports, to check various call statistics over extensions, trunks, queues, ring groups, and more.

Call reports can be viewed and downloaded directly from the PBX, or scheduled to be sent via email to respective parties.

Security

Restrict a general group or specific IP addresses to prevent unauthorized access to your PBX.

Limit the number of otbound calls users are allowed to make within a time period to protect against toll fraud.

Restrict certain country codes for outbound calls either company-wide or on certain extensions.

User Login

Determine extension ranges for different functions.

Define use of different voice prompts depending on call scenarios or intended function.

Provision basic and advanced SIP settings to be used on the PBX.

Recordings

Record all calls made on specified trunks, including inbound and outbound calls.

Recordings

Record all calls made on specified extensions, conferences, or call queues.

Paging and Intercom

Create different types of paging and intercom broadcast groups for selected extensions.

Set a time schedule to automatically start your broadcast.

Maintenance

Check for and download firmware upgrades to ensure that your PBX is always up-to-date.

Perform back-ups to prevent data and settings from getting lost.

Perform troubleshooting on your own to debug.

Conference

Pre-set conference settings and dial-in passwords for security, and allow internal users to send conference invites to external parties.

Speed Dial

Create multiple speed dial numbers to respective external numbers frequently used for outgoing calls.

Speed dial is activated when the set prefix (default or customized) is dialed.

Ring Group

Create multiple ring groups and determine custom ring strategies and failover settings for each.

Call Queue

Create different call queues with specific ring strategies and failover settings.

Static agents permanently answer calls in queue, while dynamic agents can log in and out. Allocate different extensions to each group.

Determine what actions are to be allowed for Mangers or Agents to manage on the Queue Panel.

Call Control

Have calls received on specific SIP trunks routed to designated extensions or destinations within the PBX.

Set-up dial patterns so that outbound calls are routed to their intended destinations.

Determine how calls are routed and received at different times of a day.

Contacts

Store, retrieve, and share company contact details easily on your PBX.

Organize company contacts into dedicated phonebooks and determine accessibility rights to extension groups for each.

Bulk import a list of contacts instead of having to key details in individually.

Role and Client Permission

Pre-defined user roles within the PBX are conveniently available for allocation, or create your own set of permissions.

Restrict users from viewing specific extensions, departments, or extension groups with Extension Visibility.

Allow or resrict specific users from managing or viewing certain company contacts

Auto Provision

Pre-defined setting parameters specific to respective IP phone models are conveniently available for easy set-up.

IP Phones can be quickly provisioned to your Yeastar PBX with just the pre-defined template and device MAC address.

Trunks

Register multiple SIP trunks on a single PBX as needed.

Different types of trunks can be created, Caller ID can be customized.

Extensions

View, deploy, add, and remove user extensions within your PBX system.

Configure default call and user settings for individual extensions.

Customize allocation of user extensions to specific ring groups.

Portal Login Page

Receive your administrator log-in credentials and set-up guide via email.

Log-in on any web-browser to get started with PBX set-up.

Installation Wizard will guide you through basic configurations.

User Login

Access all company contacts made available to you by the administrator.

Create your own personal contacts list for easy reference and dialing.

Bulk import contact details for convenient data entry.

User Login

Access and manage user settings for softphone application and voicemail.

Manage call routing and presence status for your individual extension.

Retrieve or download your personal call logs and recordings.

User Login

Receive your log-in credentials and set-up guide via email, sent by your admin.

Log-in on any web-browser to get started with PBX set-up.

Configure contacts and personal settings from the web portal.

Auto Provision

Add, remove, or modify IP phone settings from your web interface.

Upgrade IP phone firmware conveniently from your PBX.

Upload various default setting parameters specific to respective IP phone models for greater convenience during set-up.

Contacts

Store, retrieve, and share company contact details easily on your PBX.

Have detailed contact information for your vendors and clients conveniently stored .

Bulk add a list of contacts instead of having to key details in individually.

PBX Monitor

Monitor the status extensions, whether they aren idle, in use, or unavailable.

Know whether your SIP Trunk running properly, or experiencing problems.

Monitor the number of call conferences currently running on your PBX, and their individual durations.

Maintenance

Check for and download firmware upgrades to ensure that your PBX is always up-to-date.

Perform back-ups to prevent data and settings from getting lost.

Perform troubleshooting on your own to debug.

Call Detailed Report

All call log activities are stored, with the option of having audio conversations recorded by default or for specific extensions.

Filter by date, time, or extension number to retrieve specific call logs or recordings

Playback call recordings directly from the PBX or download to store externally.

Event Center

Determine what events and logs are to be recorded and/or have notifications sent.

Specify particular notifications to be sent to respective authorized personnel.

User Permission

Set up vaious groups of users for allocation to certain sets of access rules.

Determine what features a certain group of extensions can or cannot access.

Determine if users can access, play-back, or download call logs and recordings from the pbx.

Security

Restrict a general group or specific IP addresses to prevent unauthorized access to your PBX.

Determine specific the range of SIP ports for connection between your PBX and endpoints.

Restrict country codes for outbound calls either company-wide or on certain extensions.

Voice Prompt

Upload and designate customised audio recordings at multiple levels of a call journey.

Upload recorded music to fill the silence that would be heard by callers on hold.

Upload customised voice prompts to create a professional call experience.

General

Customize general call preferences either company-wide or for individual extensions.

Set-up feature dial codes for convenient user navigation to specific call features.

Determine how voicemails are retrieved, and also upload custom voicemail recordings.

Call Features

Add multiple automated responses to be played at different phases of a call journey.

Designate specific extensions to ring concurrently based on an external caller's dial pattern.

Customize messages and waiting times to keep external callers engaged when put on hold.

Call Control

Have calls received on specific SIP trunks routed to designated extensions within the PBX.

Set-up dial patterns so that outbound calls are routed to their intended destinations.

Determine how calls are routed and received at different times of a day.

Trunks

Register multiple SIP trunks on a single PBX as needed.

Provision SIP trunk settings and dedicate them to particular extensions.

Bulk add DDIs on each SIP trunk, for allocation to an individual or groups of extensions.

Extensions

View, deploy, add, and remove user extensions within your PBX system.

Allocate and deploy specific extension numbers to respective users via email.

Set-up default ring and presence status settings for individual users.

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Deal One View

View cases created for a Deal in One View.

View cases created for a Deal in One View.

View cases created for a Deal in One View.