| 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. |