VoIP Performance Management

Joao Antunes Software Engineer

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Corporate Network A

Corporate Network B

VoIP Performance Management

RTP

SIP SIP

SIP

Internet

SIP gateway SIP gateway

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Corporate Network A

Corporate Network B

VoIP Performance Management

Internet

Agent

RTP

Agent

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•  Key use cases –  Planning VoIP deployment (capacity, scaling, traffic management) –  Troubleshooting existing infrastructure

•  Each test measures metrics in a single direction –  Displays loss, latency, discards, packet delay and QoS metrics –  To simulate a voice call, a test in each direction is desirable

•  VoIP requires an agent on both ends (UDP+RTP-based) –  At least one agent must be an Enterprise Agent –  Enterprise Agents ideally located in branch offices

VoIP Performance Management

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VoIP Performance Management

Average of packet delays

99.9th percentile of packet delay variation

Packets dropped by the de-jitter buffer

Packets dropped by the network

MOS Score (1-5)

Audio codec used

Quality of Service requested (IP layer)

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VoIP Performance Management

Average of packet delays

99.9th percentile of packet delay variation

Packets dropped by the de-jitter buffer

Packets dropped by the network

MOS Score (1-5)

Audio codec used

Quality of Service obtained (IP layer)

Average of packet delays

99.9th percentile of packet delay variation

Packets dropped by the de-jitter buffer

Packets dropped by the network

MOS Score (1-5)

Audio codec used

Quality of Service requested (IP layer)

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VoIP Performance Management

Jump to… •  Path Visualization •  BGB Route Visualization

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•  Mean Opinion Score (MOS) E-Model (ITU-T Recommendation G.107, 1998-2014) Based on a mathematical model in which the individual transmission parameters are transformed into different individual "impairment factors” such as codec characteristics, delay, loss ratio, discard ratio, etc., to obtain a quality metric called R factor:

VoIP Performance Management

Basic signal-to-noise ratio

Delay impairment

Equipment impairment

Advantage factor

(expectation)

•  Network latency •  De-jitter buffer size

•  Ie (codec) •  Packet loss robustness (codec) •  Packet loss probability

•  Network latency

Simultaneous impairment

4 Rec. ITU-T G.107 (12/2011)

There are three different parameters associated with transmission time. The absolute delay Ta represents the total one-way delay between the send side and receive side and is used to estimate the impairment due to excessive delay. The parameter mean one-way delay T represents the delay between the receive side (in talking state) and the point in a connection where a signal coupling occurs as a source of echo. The round-trip delay Tr only represents the delay in a 4-wire loop, where the "double reflected" signal will cause impairments due to listener echo.

7.1 Calculation of the transmission rating factor, R According to the equipment impairment factor method, the fundamental principle of the E-model is based on a concept given in the description of the OPINE model (see [b-ITU-T P-Sup.3]).

Psychological factors on the psychological scale are additive.

The result of any calculation with the E-model in a first step is a transmission rating factor R, which combines all transmission parameters relevant for the considered connection. This rating factor R is composed of:

AIe-effIdIsRoR +!!!= (7-1)

Ro represents in principle the basic signal-to-noise ratio, including noise sources such as circuit noise and room noise. Factor Is is a combination of all impairments which occur more or less simultaneously with the voice signal. Factor Id represents the impairments caused by delay and the effective equipment impairment factor Ie-eff represents impairments caused by low bit-rate codecs. It also includes impairment due to randomly distributed pack losses. The advantage factor A allows for compensation of impairment factors when the user benefits from other types of access to the user. The term Ro and the Is and Id values are subdivided into further specific impairment values. The following clauses give the equations used in the E-model.

7.2 Basic signal-to-noise ratio, Ro The basic signal-to-noise ratio Ro is defined by:

( )NoSLRRo +!= 5.115 (7-2)

The term No [in dBm0p] is the power addition of different noise sources:

»»¼

º

««¬

ª+++= 10101010 10101010log10

NfoNorNosNc

No (7-3)

Nc [in dBm0p] is the sum of all circuit noise powers, all referred to the 0 dBr point.

Nos [in dBm0p] is the equivalent circuit noise at the 0 dBr point, caused by the room noise Ps at the send side:

( )214004.0100 !!!+!!!= DsOLRPsDsSLRPsNos (7-4)

where OLR = SLR + RLR. In the same way, the room noise Pr at the receive side is transferred into an equivalent circuit noise Nor [in dBm0p] at the 0 dBr point.

2)35(008.0121 !++!= PrePreRLRNor (7-5)

The term Pre [in dBm0p] is the "effective room noise" caused by the enhancement of Pr by the listener's sidetone path:

»»¼

º

««¬

ª++= 10

)–10(

101log10LSTR

PrPre (7-6)

It’s time to see the entire picture.

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•  Audio codec –  Generated traffic vs audio compression/quality

•  DSCP –  Traffic shaping, firewall and LB configuration

•  De-jitter buffer –  Network jitter vs call latency

•  Network –  Capacity, latency, loss, etc.

VoIP Performance Management