VoIP Glossary

Term

Definition

VoIP

Voice over Internet Protocol. A technology that enables voice communication over the internet.

SIP

Session Initiation Protocol. A protocol used to establish, modify, and terminate real-time sessions that involve video, voice, messaging, and other communications applications and services between two or more endpoints on IP networks.

PBX

Private Branch Exchange. A telephone system used within an organization that allows for internal communication and external communication through a Public Switched Telephone Network (PSTN).

ATA

Analog Telephone Adapter. A device that allows analog telephones to connect to a digital VoIP network.

Softphone

A software-based phone that allows users to make and receive calls over the internet using a computer or mobile device.

Codec

A method of encoding and decoding voice data into digital signals for transmission over a network.

QoS

Quality of Service. A set of technologies and techniques used to ensure that VoIP traffic is given priority on a network and that voice quality is not degraded by other network traffic.

Latency

The time delay between the transmission of a voice packet and its arrival at the destination.

Jitter

The variation in delay between voice packets arriving at their destination.

Bandwidth

The amount of data that can be transmitted over a network in a given amount of time, usually measured in kilobits per second (Kbps) or megabits per second (Mbps).

Firewall

A security system that controls access to a network by analyzing incoming and outgoing network traffic and allowing or blocking specific traffic based on a set of security rules.

VPN

Virtual Private Network. A network that allows users to securely connect to a remote network over the internet.

NAT

Network Address Translation. A method of remapping one IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device.

Echo cancellation

A technique used to eliminate echo in VoIP calls caused by the delay between the transmission and reception of voice packets.

DTMF

Dual-Tone Multi-Frequency. A method of sending signals over a VoIP network to control functions such as call routing or voicemail access.

PSTN

Public Switched Telephone Network. The traditional phone system that uses physical copper wires to transmit analog voice signals between two points.

IP phone

A telephone that uses VoIP technology to make and receive calls over the internet.

Hosted VoIP

A type of VoIP service where the VoIP provider hosts the PBX system in their data center and provides the service to customers over the internet.

On-premises VoIP

A type of VoIP system where the PBX system is hosted on-site within an organization’s own network.

Unified Communications

A suite of communication tools and services that integrate voice, video, messaging, and collaboration tools into a single platform.

SIP trunking

A method of connecting an organization’s PBX to a VoIP service provider’s network over the internet, allowing for both internal and external communication.

DID

Direct Inward Dialing. A service that allows individuals within an organization to have their own phone number, which can be dialed directly from an external phone.

Auto-attendant

A feature of a VoIP system that answers incoming calls and provides a menu of options for the caller to choose from.

Call forwarding

A feature that allows calls to be redirected to another phone number, such as a mobile phone, if the primary phone is unavailable.

Conference calling

A feature that allows multiple users to participate in a call simultaneously.

Voicemail

A service that allows callers to leave a message when the called party is unavailable.

Virtual number

A phone number that is not associated with a physical phone line, but instead routes calls to another phone number.

Porting

The process of transferring an existing phone number from one service provider to another.

E911

Enhanced 911. A service that automatically transmits a caller’s location information to emergency services when 911 is dialed.

RTP

Real-time Transport Protocol. A protocol used to transport audio and video data over IP networks.

SRTP

Secure Real-time Transport Protocol. A protocol that provides encryption, authentication, and integrity for VoIP data.

NAT traversal

A method of allowing VoIP traffic to traverse through NAT devices, which may change the source and destination IP addresses of VoIP packets.

Jitter buffer

A buffer used to temporarily store and smooth out voice packets that arrive at irregular intervals to prevent voice quality degradation.

T.38

A protocol used to transmit faxes over VoIP networks.

Echo

A phenomenon where a person hears their own voice repeated back to them during a phone call, caused by a delay in the transmission of audio data.

Latency compensation

A technique used to compensate for the delay (latency) caused by VoIP transmission to maintain real-time conversation.

SIP Proxy Server

A server that acts as an intermediary between two SIP endpoints, relaying SIP messages between them.

Softswitch

A software-based PBX that performs call control, routing, and billing functions for VoIP networks.

Quality of Experience (QoE)

A measure of overall user satisfaction with the quality of a VoIP call, taking into account factors such as voice quality, delay, and jitter.

Call detail record (CDR)

A record that contains information about a VoIP call, including the calling and called parties, the time and date of the call, and its duration.

International Termination

The process of routing calls from one country to another, typically involving multiple carriers.

Session Border Controller (SBC)

A device that controls the signaling and media streams in a VoIP network, providing security and connectivity between different network segments.

WebRTC

Web Real-Time Communication. A technology that enables real-time communication, such as voice and video calling, directly from web browsers without the need for plugins or additional software.

DNS

Domain Name System. A system that translates domain names into IP addresses, allowing devices to locate and communicate with each other over the internet.

SIP trunk

A virtual phone line that connects a company’s PBX to the internet, allowing for voice and data communication.

VPN tunneling

A method of creating a secure, encrypted connection between two points over the internet, allowing for secure VoIP communication.

Codec negotiation

The process of selecting a codec that both endpoints support for voice communication.

Packet loss

The loss of data packets during transmission, which can result in degraded voice quality.

DSCP

Differentiated Services Code Point. A field in the IP header that is used to prioritize and differentiate types of traffic.

Bandwidth shaping

The process of managing network traffic to ensure that VoIP traffic is given priority and other traffic is limited, preventing congestion.

QoS policies

A set of rules that dictate how network traffic is managed and prioritized, ensuring that VoIP traffic is given priority over other types of traffic.

Fax over IP (FoIP)

The transmission of fax data over a VoIP network.

PBX system

Private Branch Exchange. A private telephone network used within an organization that allows for internal communication and external communication through a Public Switched Telephone Network (PSTN).

VoIP gateway

A device that converts analog voice signals from a traditional phone system into digital signals for transmission over a VoIP network.

Redundancy

The use of backup systems and failover mechanisms to ensure that a VoIP network remains operational in the event of a failure.

SIP registrar

A server that maintains a database of SIP user information, allowing for authentication and registration of SIP endpoints.

SIP trunk provider

A company that provides SIP trunking services, allowing organizations to connect their PBX system to the internet for voice and data communication.

Session border controller (SBC)

A device that provides security and connectivity between different network segments, controlling the signaling and media streams in a VoIP network.

SIP Proxy Server

A server that acts as an intermediary between two SIP endpoints, relaying SIP messages between them.