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. |
