Important Terminologies

Contents

Overview

While working with VQManager, you will come across many terminologies, predominantly industry-standard terms. This section provides information on some of the important terminologies.

Terminologies

Terminology

Definition

Session Initiation Protocol [SIP]

SIP is a signaling protocol for initiating an interactive user session such as video, voice, instant messaging, online games, and virtual reality. SIP is used for modifying and terminating the user sessions too

CISCO® Skinny Protocol

 

The Skinny Client Control Protocol (SCCP) is a signaling protocol between Cisco® CallManagers and Cisco® IP phones

H.223 protocol

An umbrella standard defined by the ITU-T, under which a number of other protocols fit, supporting call setup and disconnect, audio encoding/decoding, video encoding/decoding. These protocols include the H.225, H.245 protocols, plus the IETF's Real-Time Transport Protocol (RTP), and others.

H.225 protocol

Call signalling protocols and media stream packetization for packet-based multimedia communication systems 

H.245 protocol

Call control protocol for multimedia communication

RTP/RTCP

The Real-time Transport Protocol (RTP) represents the standardized packet format for delivering audio and video over the Internet. Real-time Transport Control Protocol (RTCP) provides the control information for RTP flow

CDR

Call Detail Record is a data record that contains information related to a telephone call, such as the origination and destination addresses of the call, the time the call started and ended, the duration of the call, the time of day the call was made, etc.

SNMP

Simple Network Management Protocol; a standard for gathering statistical data about network traffic and the behavior of network components; SNMP uses management information bases (MIBs), which define what information is available from any manageable network device.

NAT

Network Address Translation is a process of re-writing the internal IP address or network address into a single globally unique IP address

STUN

Simple Transversal of UDP over Nat is a network protocol which helps many types of software and hardware receive UDP data properly through home broadband routers that use network address translation (NAT)

Latency or Delay

The amount of time taken by a Packet to travel from source to destination. Together, Latency and Bandwidth define the speed and capacity of a network

Jitter

The variation in the amount of latency among the packets being received

Packet Loss

Voice packets lost during the call

Mean Opinion Score [MOS]

MOS measures subjective call quality, ranges from 1 for unacceptable to 5 for excellent.

[VoIP calls often are in the 3.5 to 4.2 range]

ACD(Average Call Duration)

The average duration of all calls made by the endpoint

Average Length of Call (ALOC)

The ratio between the sum of the time duration of all successful calls and the number of all successful calls

Answer Seizure Ratio (ASR)

The ratio between the number of successfully answered calls to the total number of calls attempted is known as Answer Seizure Ratio.

ASR = successful calls / attempted calls *100

 

The statistic pertaining to ASR is important as it may relate to a non-productive link. It helps in identifying the problem areas in your system

Average Answer Delay

The difference in time duration between successful initiation of call handling by the device and the time taken by the person on the other end to actually attend the call (i.e. delay in answering the call)

e-model

The ETSI e-model as defined in ITU-T G.107 is a planning tool that assigns a certain equipment impairment factor Ie to each piece of equipment in the transmission chain. These Ie values are then summed up and combined with several other parameters to give the final R factor or R rating.

R factor

Similar to MOS, both are derived from the same formula and follow the same trend range from

ranges from 1-100, 100 is good and 1 poor

CODECS

CODECS are used to convert an analog voice signal to digitally encoded version. CODECS vary in the sound quality, the bandwidth required, the computational requirements etc., Each service, program, phone, gateway etc., typically supports several different CODECS and when talking to each other, negotiate which CODEC they will use.

 

 

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