Planning and Troubleshooting VoIP Performance

Nick Kephart, Director of Product Marketing Joao Antunes, Software Engineer

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We are building a performance management platform architected for the cloud era We make monitoring complex enterprise networks easy and enable you to find and solve problems regardless of where they occur

About ThousandEyes

Founded in 2010 by UCLA PhDs and backed by:

What We Do Our Background

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•  VoIP – Voice over IP –  Set of protocols designed to deliver voice communication over the

Internet –  H.323, MGCP, SIP, H.248 RTP, RTCP, SRTP, SDP, IAX, etc.

•  Two phases: –  Signaling (e.g., SIP) –  Audio transport (e.g., RTP) with audio frames (e.g., G.711)

What Is VoIP

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Corporate Network A

Corporate Network B

Making a VoIP Call

RTP

SIP SIP

SIP

Internet

SIP gateway

SIP gateway

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•  Audio frames are encapsulated in RTP packets •  RTP packets are encapsulated in UDP packets •  UDP packets are encapsulated in IP packets

Inside a Voice Packet

Frame 1 RTP header Frame 2 IP

header UDP header

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•  Audio codec –  Generated traffic vs audio compression/quality

•  QoS – IP Differentiated Services (DSCP) –  Traffic shaping, firewall and LB configuration –  3 bits for class: Best effort, Assured Forwarding, Expedited

Forwarding, Voice Admit

•  De-jitter buffer –  Network jitter vs call latency

•  Network –  Latency, loss, etc.

Key VoIP Metrics and Concepts

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Time

Packet delay (from sender to receiver)

Latency

Packet 1 Packet 2 Packet 4 Sent at

Packet 1 Packet 2 Packet 4 Received at Packet 3

Packet 3

Latency Latency Latency Latency

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Time

Variation of the latency

Jitter

Packet 1 Packet 2 Packet 4 Sent at

Packet 1 Packet 2 Packet 4 Received at Packet 3

Packet 3

Min Latency Max Latency

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Time

99.9th percentile of the packet delay variation

Packet Delay Variation

Packet 1 Packet 2 Packet 4 Sent at

Packet 1 Packet 2 Packet 4 Received at Packet 3

Packet 3

Played at

Delayed playback

Min Latency Max Latency PDV = max latency – min latency De-jitter buffer should be able to accommodate PDV.

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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:

Mean Opinion Score (MOS)

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)

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VoIP Metrics

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

Source

Destination

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•  Regional expansion •  New offices, call centers

and locations •  Network topology and

routing •  Capacity and utilization

Key Use Cases

•  Latency, jitter and loss •  Infrastructure faults •  Routing issues •  QoS and DSCP values

Pre-Deployment Post-Deployment

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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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How ThousandEyes Voice Tests Work

Enterprise

Enterprise Agent Cloud Agent (at dozens of global POPs)

Active Tests (create a test in each direction)

ThousandEyes SaaS Platform

Branch office

Branch office

Enterprise Agent

External caller

Agent required on both ends

At least one Enterprise

Agent required

Demo

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Add a New Voice Test

Select voice test

Select target

Select source

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Add a New Voice Test: Advanced Settings

On Advanced Settings

Choose codec, DSCP and de-jitter

buffer

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Voice Metrics

Target Agent

Source Agents

Up to 30 day timeline

Metric selector

Metric averages

Jump to other views

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Voice Metrics: Three Agents Experiencing Loss

Source Agents

Packet loss

Lower MOS scores

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Path Visualization: Common Problem Node

Details for problem node

Target Agent

Three agents with issues

Issue summary

View the Live Demo http://vimeo.com/thousandeyes/

voip-monitoring