Glossary
Short definitions for common Voice, Network, and IT terms
What Is SIP? Session Initiation Protocol Guide
SIP (Session Initiation Protocol, RFC 3261) is the application-layer signalling protocol that creates, modifies and terminates voice, video and other real-time sessions over IP. This reference explain…
glossaryWhat Is PBX?
PBX is the internal telephone switching system in an organization, managing extensions, external trunks, IVR, voicemail, and other telephony features.…
glossaryWhat Is IP PBX?
IP PBX is a PBX that communicates over IP networks rather than TDM and copper, primarily using SIP. It enables full VoIP feature sets.…
glossaryWhat Is Cloud PBX?
Cloud PBX is a phone system that runs on a provider's cloud servers, accessed by customers over the internet on a monthly subscription, without on-site hardware.…
glossaryWhat Is Hosted PBX?
Hosted PBX is a PBX maintained and delivered as a subscription service. Customers do not own the hardware. The term is often used interchangeably with Cloud PBX.…
glossaryWhat Is SBC?
An SBC is a border device between two voice networks on IP. It is used to connect customer PBX to carriers, enable Microsoft Teams Direct Routing, and bridge SIP Trunks.…
glossaryWhat Is DID?
DID is a carrier service that gives organizations multiple external numbers, each tied to an internal extension so callers can dial through directly without a receptionist.…
glossaryWhat Is BYOC?
BYOC lets customers connect their own carrier to Cloud PBX, Microsoft Teams, or UC platforms instead of purchasing voice plans bundled by the platform vendor.…
glossaryWhat Is G.711?
G.711 is the ITU-T standard audio codec for telephony, using 64 Kbps per direction at 8 KHz narrowband quality. It is the baseline codec supported by every VoIP / ISDN / PSTN system.…
glossaryWhat Is G.722?
G.722 is an audio codec providing HD voice (16 KHz wideband) at 64 Kbps, same bandwidth as G.711 but noticeably clearer and more natural sound. It is the most common HD codec on IP phones.…
glossaryWhat Is G.729?
G.729 compresses voice down to just 8 Kbps per direction, the lowest of the standard codecs. Ideal for bandwidth-limited WAN links.…
glossaryWhat Is Opus?
Opus is an open royalty-free audio codec from IETF that adapts bitrate from 6 to 510 Kbps in real time. It outperforms other codecs at any equivalent bitrate and dominates WebRTC use cases.…
glossaryWhat Is SRTP?
SRTP is the encrypted version of RTP, used in VoIP to prevent on-path eavesdropping. Pair it with TLS, which encrypts the SIP signaling.…
glossaryWhat Is VLAN?
VLAN is a Layer 2 segmentation technique that divides one switch into multiple isolated networks, with separate broadcast domains. Used for security and traffic management.…
glossaryWhat Is SD-WAN?
SD-WAN is the modern WAN paradigm: software centrally manages multiple links, selecting the best path per application automatically. It replaces or augments traditional MPLS.…
glossaryWhat Is QoS?
QoS prioritizes traffic on a network so latency-sensitive apps (voice, video) get bandwidth before tolerant apps (email, file transfer).…
pillarWhat Is NAT? How It Works and How It Is Classified
NAT (Network Address Translation) rewrites the IP addresses of packets as they pass through a router, firewall, or NAT gateway. The translation may affect source data, destination data, or both, and s…
glossaryWhat Is FXS?
An FXS port on a gateway or PBX supplies -48V DC power, dial tone, and ring voltage to analog devices like a carrier line would. Used to connect old analog phones, faxes, and door phones.…
glossaryWhat Is FXO?
An FXO port acts as a phone endpoint, receiving power and ring from a carrier. Used to bring legacy analog trunks from NT or True into an IP PBX.…
glossaryWhat Is DECT?
DECT is a wireless phone standard on dedicated 1.9 GHz spectrum in Thailand. It covers wider area more reliably than WiFi VoIP. Common in factories, warehouses, hospitals, and hotels.…