Voice and telephony glossary

0

0300 and 0303 numbers (shared-cost numbers (0300 and 0303))

0300 and 0303 numbers are Brazilian nationwide shared-cost access codes, where the cost of the call is split between the caller and the number's subscriber, under Anatel's numbering plan.

0800 number (toll-free number (0800))

An 0800 number is a non-geographic access code where the call is free for the caller and the number's subscriber pays for it, under Anatel's numbering plan.

4

4004, 4003 and 3003 numbers (single national number (4004, 4003, 3003))

4004, 4003 and 3003 numbers are eight-digit fixed-line access codes that Brazilian carriers sell as a single nationwide number, dialed without an area code in the locations the carrier serves.

D

DID (DID (Direct Inward Dialing))

DID (Direct Inward Dialing, called DDR in Brazil) is the service that maps public numbers to the extensions of a PBX, so outside callers reach an extension directly without going through an operator or an IVR.

DTMF (DTMF (Dual-Tone Multi-Frequency))

DTMF (Dual-Tone Multi-Frequency) is the system that represents each phone key as the sum of two frequencies, used for dialing and for choosing options in IVRs.

E

E1 trunk

An E1 trunk is a 2.048 Mbit/s digital circuit between the carrier and the PBX that carries up to 30 simultaneous voice channels, with ISDN or R2 signaling.

Early media (early media (SIP 183 Session Progress))

Early media is audio sent to the caller before the call is answered, usually after a SIP 183 Session Progress response, such as ringback tone or a carrier message.

I

IP PBX

An IP PBX is a company's private telephone exchange running over an IP network, connecting extensions, trunks and IVR and routing internal and external calls with SIP and RTP.

IVR (IVR (Interactive Voice Response))

An IVR (Interactive Voice Response, called URA in Brazil) is the system that answers calls without a human agent, plays recorded messages and routes callers based on DTMF keypresses or speech.

J

Jitter (jitter)

Jitter is the variation in the arrival interval of a voice call's packets, which forces the receiver to hold packets in a buffer before playing the audio.

M

Maintenance window (maintenance window)

A maintenance window is a scheduled, agreed period in which a service may be unavailable for planned work, without failures during that interval being treated as incidents.

MOS (MOS (Mean Opinion Score))

MOS (Mean Opinion Score) is the 1-to-5 score, defined by ITU-T P.800, that expresses the voice quality perceived by someone listening to a call.

MOS-LQO (MOS-LQO (Mean Opinion Score, Listening Quality Objective))

MOS-LQO is the 1-to-5 score, defined in ITU-T P.800.1, that an algorithm calculates by comparing the audio received on a call with the reference audio that should have arrived.

MTTR and MTTA (MTTR (Mean Time to Repair) and MTTA (Mean Time to Acknowledge))

MTTR and MTTA are per-incident time averages: MTTA measures how long the team takes to acknowledge the alert, and MTTR how long it takes to restore the service.

P

Packet loss (packet loss)

Packet loss is the fraction of a VoIP call's packets that never reach the destination or arrive too late to be played, causing gaps in the audio.

POLQA and PESQ (POLQA (Perceptual Objective Listening Quality Analysis) and PESQ (Perceptual Evaluation of Speech Quality))

POLQA (ITU-T P.863) and PESQ (ITU-T P.862) are ITU standards that calculate a MOS-LQO score by comparing the audio received on a call with the reference audio.

R

RTP (RTP (Real-time Transport Protocol))

RTP (Real-time Transport Protocol), defined in RFC 3550, is the protocol that carries a VoIP call's audio in numbered, timestamped packets.

S

SBC (SBC (Session Border Controller))

An SBC (Session Border Controller) is the device or software at the edge of a VoIP network that controls, protects and adapts the SIP signaling and audio going in and out.

SIP (SIP (Session Initiation Protocol))

SIP (Session Initiation Protocol), defined in RFC 3261, is the signaling protocol that sets up, modifies and ends voice calls over IP networks.

SIP ALG (SIP ALG (Application Layer Gateway))

SIP ALG (Application Layer Gateway) is a feature of routers and firewalls that rewrites addresses inside SIP messages to get through NAT, often breaking the audio on calls.

SIP registration (SIP registration (REGISTER))

SIP registration is the process in which a phone, PBX or trunk sends a REGISTER message to the registrar to report its current address and be able to receive calls for its account.

SIP trunk

A SIP trunk is the voice connection between a company's PBX and the carrier over an IP network, with SIP signaling and audio in RTP, in place of physical lines or E1.

SLA (Service Level Agreement)

An SLA (Service Level Agreement) is the contractual commitment that defines the minimum level of a service, such as 99.9% availability, and how it is measured and calculated.

STIR/SHAKEN (STIR/SHAKEN (Secure Telephone Identity Revisited / Signature-based Handling of Asserted information using toKENs))

STIR/SHAKEN is the set of standards that digitally signs the calling number of a SIP call, so the terminating network can verify whether the caller ID is legitimate.

V

ViSQOL (ViSQOL (Virtual Speech Quality Objective Listener))

ViSQOL (Virtual Speech Quality Objective Listener) is an open-source algorithm from Google Research that compares received audio with its reference and returns a 1-to-5 MOS-LQO score.

Voice codecs (voice codecs (G.711, G.729, Opus))

Voice codecs are the algorithms that encode speech into digital data for transmission on a call and decode it at the destination, such as G.711, G.729 and Opus.