Glossary

VoIP glossary

Plain-English definitions of the acronyms and jargon you'll meet when choosing or running a VoIP phone system.

ATA (Analogue Telephone Adapter): A small box that connects an ordinary analogue phone, fax or alarm to a VoIP service. Many NBN routers have one built in as a "phone" port.
Auto attendant: The automated greeting and menu that answers calls ("press 1 for sales"). Increasingly replaced or supplemented by speech-driven AI routing.
Codec: The algorithm that compresses voice into data and back. Common examples are G.711, G.722, G.729 and Opus. See our codec guide.
Channel: Capacity for one simultaneous call on a SIP trunk. Ten channels means up to ten calls at once.
CCO (Copper Continuity Obligation): Telstra's obligation to keep providing eligible copper voice services in areas outside the NBN fixed-line footprint.
DID (Direct Inward Dialling): A phone number that routes straight to a specific extension, group or service on your phone system.
Direct Routing: Connecting Microsoft Teams Phone to your own SIP trunk through a certified session border controller.
DSCP: A marking in each IP packet that tells routers and switches how to prioritise it. Voice audio is usually marked EF (46).
EF (Expedited Forwarding): The DSCP class (value 46) used for real-time voice packets so they jump the queue.
HD voice / wideband audio: Voice carried with roughly twice the frequency range of a traditional phone call (up to about 7 kHz), using codecs like G.722, Opus or AMR-WB. Sounds noticeably clearer.
Hunt group / ring group: A group of extensions that rings together or in sequence when a single number is called.
IVR (Interactive Voice Response): An automated phone system that interacts with callers by keypad or voice, from simple menus to AI voice agents.
Jitter: Variation in the time it takes packets to arrive. High jitter causes robotic or warbling audio. Under 20 ms is good.
Jitter buffer: A small delay at the receiving end that collects packets and plays them out evenly to smooth over jitter.
Latency: The delay for audio to travel from speaker to listener. Under 150 ms one way feels natural; above 300 ms people talk over each other.
LLDP-MED: A network protocol that lets switches tell phones which VLAN and QoS settings to use automatically.
MOS (Mean Opinion Score): A 1–5 rating of perceived call quality. Above 4.0 is good; many systems estimate it from jitter, latency and loss.
NAT (Network Address Translation): How routers share one public IP address across many devices. A common cause of VoIP problems such as one-way audio when combined with SIP ALG.
Operator Connect: A Microsoft Teams Phone option where an approved carrier supplies numbers and calling directly into Teams, without customer-run equipment.
Opus: A modern, open, adaptive codec used by WebRTC and many softphones. Handles poor networks well.
Packet loss: Packets that never arrive. Above about 1–2% causes audible dropouts.
PBX (Private Branch Exchange): The phone system that manages extensions, transfers, voicemail and routing for an organisation. Can be on-site, self-hosted or cloud-hosted.
PoE (Power over Ethernet): Supplying power to desk phones through the network cable from the switch.
Porting: Moving an existing phone number from one provider to another.
PSTN: The Public Switched Telephone Network: the traditional circuit-switched phone network, increasingly replaced by all-IP networks.
QoS (Quality of Service): Router and switch settings that prioritise voice traffic over other data so calls stay clear when the connection is busy.
RTP / SRTP: Real-time Transport Protocol, which carries the audio of a VoIP call. SRTP is the encrypted version.
SBC (Session Border Controller): A device or service that sits at the edge of a VoIP network to secure, route and normalise SIP traffic.
SIP (Session Initiation Protocol): The signalling protocol most VoIP uses to set up, modify and end calls.
SIP ALG: A router feature that rewrites SIP messages to help with NAT. Frequently causes problems and is usually best turned off.
SIP trunk: A service connecting your own PBX to the public phone network over SIP. See SIP trunking explained.
Softphone: A phone app on a computer or mobile that registers to a phone system like a desk phone.
SQM (Smart Queue Management): A router feature that manages upload and download queues to keep latency low for real-time traffic such as voice.
STIR/SHAKEN: A framework used in North America to digitally sign calls so carriers can verify the caller ID is genuine.
TLS: Encryption for SIP signalling, usually on port 5061.
Toll fraud: Criminals using a compromised phone system to make expensive calls at the owner's cost. See the security checklist.
Transcoding: Converting audio from one codec to another mid-call. Adds delay and slightly reduces quality.
UCaaS (Unified Communications as a Service): Cloud platforms that combine phone, video meetings, messaging and often contact centre features.
VLAN: A virtual network on shared switches. A dedicated voice VLAN keeps phone traffic separate and easier to prioritise.
VoLTE / VoWiFi: Voice over LTE and Wi-Fi calling: the mobile networks' own VoIP technologies, now standard on Australian mobile networks.
WebRTC: Technology built into web browsers for real-time voice and video, used by many browser-based softphones and contact centres.