The SIP & VoIP glossary

Every term that comes up when you add mobile, push and Teams calling to a phone system, explained in plain English. Bookmark it.

SIP & signalling

SIP: Session Initiation Protocol, the signalling protocol that sets up, changes and ends voice and video calls over IP.
SDP: Session Description Protocol, carried inside SIP to negotiate media, codecs, IP addresses and ports for a call.
REGISTER: The SIP request a phone sends to tell the network where it can be reached, refreshed periodically so it stays online.
INVITE: The SIP request that starts a call between two endpoints.
Registrar: The server that accepts REGISTER requests and tracks where each user is currently reachable.
SIP proxy: A server that routes SIP requests between endpoints without terminating the call itself.
B2BUA: Back-to-back user agent, an element that terminates and re-originates both call legs, giving it full control. An SBC is a B2BUA.
Digest authentication: The challenge-response scheme (username, password, nonce) SIP uses to authenticate registrations and calls.

RTP: Real-time Transport Protocol, which carries the actual audio and video packets of a call.
SRTP: The encrypted form of RTP, securing the media stream so calls cannot be intercepted.
Codec: The algorithm that encodes and decodes audio, such as Opus or G.711.
G.711: The classic uncompressed PCM codec (a-law / µ-law), high quality at about 64 kbps.
Opus: A modern, adaptive wideband codec used for HD VoIP and WebRTC.
DTMF (RFC 2833): Touch-tone digits, carried as RTP events per RFC 2833 / 4733 rather than as audio, so they survive compression.
Jitter buffer: A small buffer that smooths out variation in packet arrival times to keep audio clean.
MOS: Mean Opinion Score, a 1 to 5 measure of perceived call quality.
ASR: Answer-Seizure Ratio, the share of call attempts that are answered. A key measure of route and carrier quality.
ACD: Average Call Duration, the mean length of answered calls. Read alongside ASR to judge traffic and route health.
RTT: Round-Trip Time, the latency for a packet to reach the far end and return. High RTT degrades call quality.

NAT: Network Address Translation, the reason devices behind routers and firewalls are hard to reach directly.
STUN: A method for a device to discover its public IP and port when it sits behind NAT.
TURN: A relay that carries media when a direct path cannot be established, for example behind strict corporate firewalls.
ICE: The framework that tries multiple paths (host, STUN, TURN) to find the best media route between two endpoints.
Anycast: Routing that sends traffic to the nearest of many identical nodes by network distance.
GeoDNS: DNS that returns the nearest or healthiest server based on where the client is.

PBX: Private Branch Exchange, the phone system that routes calls inside an organization.
IP PBX: A PBX that runs over IP and SIP, such as Asterisk, FreeSWITCH or 3CX.
SIP trunk: A SIP connection between a PBX and a carrier that carries calls to and from the PSTN.
DID: Direct Inward Dialing, an external phone number that routes to a specific extension or user.
E.164: The international phone-number format, for example +15125550123.
PSTN: The Public Switched Telephone Network, the traditional global phone network.
MWI: Message Waiting Indicator, the signal that lights up "you have voicemail".

Softphone: A software phone, an app that makes and receives SIP calls.
VoIP push: A notification over APNs or FCM that wakes a softphone for an incoming call without keeping a socket open.
APNs: Apple Push Notification service, used to wake iOS apps.
FCM: Firebase Cloud Messaging, the push service for Android.
PushKit: Apple’s framework for VoIP pushes that can wake an app specifically to receive a call.
CallKit: Apple’s framework that renders VoIP calls with the native iOS call screen, on the lock screen and over Bluetooth.
ConnectionService: Android’s equivalent of CallKit, integrating VoIP calls into the system dialer.
BYOD: Bring Your Own Device, registering existing SIP phones and clients to a service.

SBC: Session Border Controller, a back-to-back user agent at the network edge that secures, normalizes and controls SIP and RTP.
TLS / WSS: Encryption for SIP signalling (TLS) and SIP over WebSocket (WSS), used by browser and app clients.
mTLS: Mutual TLS, where both sides present certificates. Microsoft Teams Direct Routing uses it.
STIR/SHAKEN: The framework that cryptographically signs and verifies caller ID to fight spoofing and robocalls.
Toll fraud: Abuse where attackers place expensive calls on a compromised account, the number-one worry for ITSPs.

Teams Phone: Microsoft’s calling capability and licence for Microsoft Teams.
Direct Routing: Connecting Teams to any carrier through a certified SBC, the bring-your-own-carrier path.
Operator Connect: A Microsoft program where a listed operator runs the SBC and you assign numbers in the admin center.
Calling Plans: Microsoft acting as your PSTN carrier directly, bundling numbers and minutes as a licence.

Registration bridge: Spanvox’s model: a server-side upstream registration to your PBX plus a push-driven downstream registration from the app, so no trunk or dialplan change is needed.
BYOC: Bring Your Own Carrier, keeping your existing PSTN provider and rates.
Media proxy: A managed relay that anchors media to traverse NAT and keep call quality high.