EXFO EXpert VoIP Test Tools User Manual

EXpert VoIP Test Tools

User Guide

Copyright © 2014 EXFO Inc. All rights reserved. No part of this publication may be reproduced, stored in a retrieval system or transmitted in any form, be it electronically, mechanically, or by any other means such as photocopying, recording or otherwise, without the prior written permission of EXFO Inc. (EXFO).

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Units of measurement in this publication conform to SI standards and practices.

October 16, 2014

Version number: 2.0.0.0

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Licence Agreement and Warranty

IMPORTANT: CAREFULLY READ THE FOLLOWING LICENSE AGREEMENT BEFORE OPENING THIS INSTALLATION PACKAGE. BY OPENING THIS PACKAGE AND USING THE SOFTWARE WHETHER INCORPORATED OR NOT IN AN EXFO INC. ("EXFO") PRODUCT, YOU INDICATE YOUR ACCEPTANCE TO BE BOUND BY THE TERMS AND CONDITIONS OF THIS AGREEMENT. IF YOU DO NOT ACCEPT THE TERMS AND CONDITIONS OF THIS LICENSE AGREEMENT, DO NOT OPEN THIS PACKAGE AND PROMPTLY RETURN THE PRODUCT OR SOFTWARE WITH YOUR PROOF OF PAYMENT, WHEREUPON YOUR MONEY WILL BE REFUNDED.

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THE PRODUCT AND THE SOFTWARE MUST BE USED ONLY FOR YOUR INTERNAL BUSINESS OPERATION AND ITS INTENDED APPLICATION. YOU MAY NOT COPY OR USE THE SOURCE CODE OR THE SOFTWARE COMPONENTS TO PRODUCE OTHER SOFTWARE DEVELOPMENT TOOLS FOR DISTRIBUTION AND RESALE WITHOUT EXPRESS WRITTEN PERMISSION FROM EXFO. EXFO RETAINS ALL RIGHTS TO THE SOURCE CODE, THE SOFTWARE COMPONENTS AND ALL MODIFICATIONS THEREOF. YOU SHALL RESPECT AND COMPLY WITH ANY OF THE PROVISIONS LISTED BELOW WHICH MAY ALSO BENEFIT ANY GIVEN THIRD PARTY BENEFICIARY AS DEFINED HEREIN.

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102 EXpert VoIP Test Tools

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Contents

1 Introducing the EXpert VoIP Test Tools ....................................................... 1

Conventions ............................................................................................................................2

2 Getting Started ............................................................................................ 3

Signal Connection ...................................................................................................................3

Laser Safety Information .........................................................................................................3

Starting the EXpert VoIP Test Tools Application .......................................................................3

3 Using the Graphical User Interface ............................................................. 5

Main Application Window .....................................................................................................5

Arrow Buttons .......................................................................................................................8

Keyboard Usage .....................................................................................................................9

4 Selecting and Starting a Test ..................................................................... 11

RTP .......................................................................................................................................12

SIP ........................................................................................................................................13

SCCP .....................................................................................................................................15

H.323 ....................................................................................................................................17

H.248/Megaco ......................................................................................................................18

5 Test Setup ................................................................................................... 19

Interface .............................................................................................................................20

Configuration (RTP) .............................................................................................................23

Configuration (SIP) ..............................................................................................................26

Configuration (SCCP) ...........................................................................................................32

Configuration (H.323) ..........................................................................................................35

Configuration (H.248/Megaco) ............................................................................................39

Controller Setup ..................................................................................................................42

Responder Setup .................................................................................................................45

Parameters (SCCP) ...............................................................................................................48

Parameters (H.323) ..............................................................................................................49

Parameters (H.248/Megaco) .................................................................................................51

Threshold .............................................................................................................................52

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6 Test Results ..................................................................................................53

Call Details ...........................................................................................................................54

Call Manager Statistics (H.248/Megaco) ..............................................................................56

Call Manager Statistics (H.323) ............................................................................................58

Call Manager Statistics (SCCP) .............................................................................................60

Call Quality Details ...............................................................................................................62

Controller Statistics ...............................................................................................................65

Responder Statistics ..............................................................................................................68

Summary (H.248/Megaco) ...................................................................................................71

Summary (H.323) .................................................................................................................73

Summary (RTP) ....................................................................................................................76

Summary (SCCP) ..................................................................................................................79

Summary (SIP) .....................................................................................................................81

7 Test Control .................................................................................................85

Start/Stop Button ..................................................................................................................85

Load/Save ...........................................................................................................................86

Save/Load - Record Audio Tab ..............................................................................................87

Save/Load - Packet Capture ..................................................................................................87

Report .................................................................................................................................88

Report - Open Tab ...............................................................................................................89

8 Troubleshooting ..........................................................................................91

Solving Common Problems ...................................................................................................91

Contacting the Technical Support Group ..............................................................................92

A Glossary .......................................................................................................93

Acronym List .........................................................................................................................93

Index .................................................................................................................97

Copyright Information ........................................................................................................101

2 EXpert VoIP Test Tools

1 Introducing the EXpert VoIP

Test Tools

The EXpert VoIP Test Tools provides Ethernet testing capability on the EXFO’s platform. It runs as a separate software application on the platform and operates independently of the modules installed on the platform. The EXpert VoIP Test Tools application generates a single VoIP call from the platform to another EXFO’s platform supporting the application or any IP phone. The call can either be "live", “stored audio” or “synthesized media”. Several signaling technologies are supported including SIP, SCCP, H.323, and H.248 to support the majority of VoIP environments currently used in the industry. Resulting quality metrics include MOS, R-factor as well as numerous RTP loss metrics.

The EXpert VoIP Test Tools application runs on the following EXFO’s platforms: FTB-1, FTB-2, FTB-2 Pro, and FTB-200v2.

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Introducing the EXpert VoIP Test Tools

Conventions

Before using the product described in this guide, you should understand the following conventions:

Indicates a potentially hazardous situation which, if not avoided, could result in death or serious injury. Do not proceed unless you understand and meet the required conditions.

Indicates a potentially hazardous situation which, if not avoided, may result in minor or moderate injury. Do not proceed unless you understand and meet the required conditions.

Indicates a potentially hazardous situation which, if not avoided, may result in component damage. Do not proceed unless you understand and meet the required conditions.

WARNING

CAUTION

IMPORTANT

Refers to information about this product you should not overlook.

2 EXpert VoIP Test Tools

2 Getting Started

Signal Connection

The EXpert VoIP Test Tools uses the 10/100/1000 Mbit/s Ethernet RJ45 port of the EXFO’s platform.

Connect the 10/100/1000 Mbit/s electrical signal to be tested to the RJ45 port of the EXFO’s platform.

Note: Refer to the EXFO’s platform User Guide for more information.

Laser Safety Information

The EXpert VoIP Test Tools software is not provided with any hardware components. However, it may be used with your platform or modules which may contain laser components. Refer to the user guides of your platform or modules for further laser safety details and instructions.

Starting the EXpert VoIP Test Tools Application

The EXpert VoIP Test Tools application is pre-installed on the EXFO’s platform. If the EXpert VoIP Test Tools is not already installed, refer to the EXFO’s platform User Guide for more information on how to install the application.

Note: The application can be used in trial mode if you do not have a valid license.

You can follow the on-screen instructions to activate the trial. If you wish to renew the trial period or purchase a valid license, then refer to MyEXFO. For information on how to install and activate software options, refer to the EXFO’s platform User Guide.

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Getting Started

Starting the EXpert VoIP Test Tools Application

To start the EXpert IP Test Tool application:

1. FTB-1: From Mini ToolBox, tap on the Te st To ol s tab.

FTB-2 and FTB-2 Pro: From Tool Bo x X, tap the Te s t Too ls button. FTB-200v2: From Compact ToolBox, tap on the Tes t Too l s tab.

2. Select the E Xpe rt VoI P Tes t Tool s then tap on Start. At startup, the application always loads the previous configuration.

4 EXpert VoIP Test Tools

3 Using the Graphical User

Main Window

Title Bar

Te s t Co n tr o l

Main Menu

Application Buttons

Status Bar

Test Results

Interface

This chapter describes the graphical user interface of the EXpert VoIP Test Tools application.

Main Application Window

The following main application window is displayed when the EXpert VoIP Test Tools application is started.

Main Window

The main window allows to setup a test and to view the test status and results.

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Using the Graphical User Interface

Main Application Window

Status Bar

The status bar displays the following information.

Icon and/or text Description

Test name Test name of the selected test.

Link arrow Green arrow: Ethernet Link up

Red arrow: Ethernet Link down.

Interface speed 10 Mbit/s, 100 Mbit/s, or 1000 Mbit/s.

Test status Current test status.

Title Bar

The title bar displays the software application name and the battery level indicator.

Test Control

Note: Refer to Test C ont ro l on page 85 for more information.

Main Menu

The EXpert VoIP Test Tools main menu allows the selection of a test tool. The selected test tool is the one with a check mark to its left. Refer to Selecting and Starting a Test on page 11.

Test Results

Note: Refer to Test R esu l ts on page 53 for more information.

6 EXpert VoIP Test Tools

Using the Graphical User Interface

Main Application Window

Application Buttons

 About (i) mainly displays the product version details and technical

support information.

Software Options button displays the list of software options enabled. For information on how to install and activate software options, refer to the EXFO’s platform User Guide.

Option Name Description

EXPERTVOIP Enable "VoIP call test tool" application

EXPERTSIP SIP call signaling support for EXpert VoIP Test Tools

EXPERTSCCP SCCP call signaling support for EXpert VoIP Test

Too ls

EXPERTH.323 H.323 call signaling support for EXpert VoIP Test

Too ls

EXPERTH.248 H.248/Megaco call signaling support for EXpert

VoI P Test Too ls

Licence Agreement button displays the details of the product licence agreement.

 Help (?) displays the help information related to the content of the

active main window. It is also possible to navigate through the remainder of the help information.

 Exit (x) closes the application.

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Using the Graphical User Interface

Moves to the top of the list.

Moves one page up.

Moves one line up.

Moves one line down.

Moves one page down.

Moves to the end of the list.

Arrow Buttons

8 EXpert VoIP Test Tools

Using the Graphical User Interface

Keyboard Usage

The GUI pops up different keyboards to modify data. Following are the usual keyboard keys:

 Left arrow moves the cursor one position to the left.

 Right arrow moves the cursor one position to the right.

 Up arrow increases the byte value by one.

 Down arrow decreases the byte value by one.

 Del deletes the value at the cursor position.

 Back deletes the value preceding the cursor position.

 OK completes data entry.

 Cancel closes the keyboard and discards the keyboard entry.

 Previous... or Presets allows the selection of previously configured

values. This button is only available for certain fields like IP Address, MAC Address, etc.

Note: For certain text fields, the GUI pops up or uses the platform’s on-screen

keyboard. Refer to the platform user guide for more information on how to used it.

For full keyboard, the Back, Del, Shift, and Space bar keys have the same functionality as a regular PC keyboard.

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4 Selecting and Starting a Test

A test can be created either by selecting the test from the Tes t Menu or by loading a previously saved configuration (refer to Load/Save on page 86 for more information).

The EXpert VoIP Test Tools main menu offers the following test tools:

Test Tools

RTP on page 12

SIP on page 13

SCCP on page 15

H.323 on page 17

H.248/Megaco on page 18

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Selecting and Starting a Test

RTP

The RTP (Real Time Protocol) network test is a peer-to-peer test that measures the VoIP related parameters by streaming RTP packets between two endpoints.

The RTP network test simulates VoIP traffic by streaming RTP packets between a Controller test set, which initiates the packet stream, and a Responder endpoint, which initiates its own stream to the Controller test set.

The Controller sends an audio packet stream to the Responder. The audio stream is based on the type of traffic parameter (and, optionally, test length parameter). Port selection is determined by the setting of the UDP Port Parameters.

The Responder initiates a packet stream to the Controller. The Controller receives the packets that are sent from the Responder.

To select, configure, and start a RTP Test:

1. From the Main Menu, tap RTP.

2. From the Interface tab, configure the test interface parameters. See

Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step.

3. From the Configuration tab, configure the RTP test. See Configuration (RTP) on page 23.

4. From the Threshold tab, enable and configure the threshold values. See Threshold on page 52.

5. Press Start to start the test.

6. For results, refer to Summary (RTP) on page 76, Controller Statistics on

page 65, and Responder Statistics on page 68.

7. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.

12 EXpert VoIP Test Tools

Selecting and Starting a Test

SIP

The SIP (Session Initiation Protocol) test measures the performance of Voice over IP (VoIP) ser vices within a network. The test uses the SIP for call signaling and the RTP for digitally transporting the encoded audio.

The SIP signaling portion of the test can be transported over UDP/TCP. SIP calls transported over UDP/TCP can be routed through a proxy server. The test's RTP functionality conforms to RFC 3550.

The SIP test measures call statistics for a SIP call and initiates a call to or answers a call from a given endpoint. It reports call setup and media path statistics between the test set and a SIP endpoint or another EXpert VoIP SIP enabled test set.

The SIP test operates in two modes.

 Network Active Test

 Service Active Test

The SIP network active test is performed, if EXFO devices (i.e. EXpert IP Test Tools devices) are selected on both ends, that is, on the Controller side and the endpoint. The SIP network active test operates peer-to-peer in UDP communication mode along with third-party SIP proxy servers to set up calls between two endpoints, a Controller, which initiates the call, and a Responder, which is responsible for completing the call and, optionally, echoing media packets sent from the Controller. For the SIP network active test in TCP communication mode third-party SIP proxy servers are mandatory. A SIP test cycle consists of the initiation and termination of an audio session (simulated audio data) between two endpoints for which it reports call setup and audio stream statistics.

SIP

The SIP test measures the call statistics. It initiates and answers a SIP call from a given endpoint. It reports call setup and media path statistics between the test set and a SIP endpoint or another EXpert VoIP SIP enabled test set.

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Selecting and Starting a Test

SIP

To select, configure, and start a SIP Test:

1. From the Main Menu, tap SIP.

2. From the Interface tab, configure the test interface parameters. See

Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step.

3. From the Configuration tab, configure the SIP test. See Configuration (SIP) on page 26.

4. From the Controller Setup tab, configure the controller settings. See

Controller Setup on page 42.

5. From the Responder Setup tab, configure the responder settings. See

Responder Setup on page 45.

6. From the Threshold tab, enable and configure the threshold values. See Threshold on page 52.

7. Press Start to start the test.

8. For results, refer to Summary (SIP) on page 81, Call Details on page 54,

and Controller Statistics on page 65.

9. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.

14 EXpert VoIP Test Tools

Selecting and Starting a Test

SCCP

The SCCP (Skinny Client Control Protocol) service active test measures the performance of calls between two endpoints and a SCCP peer endpoint made through the Cisco Call Manager.

In the SCCP test, the Controller acts as a calling endpoint (Caller), and the Responder acts as an answering endpoint (Callee). Both endpoints must be registered in the CallManager. Devices registered in the CallManager are assigned a telephone number associated with the MAC address of the device.

Note: Each endpoint executing a test against a given CallManager must have a

unique MAC address-phone number pair for that CallManager.

Note: The Caller and the Callee must be manually registered with the

CallManager with respect to their phone type before you configure the test. For executing the call successfully, the Callee’s phone number must be a valid number from the CallManager’s registration tables, and it should be associated with the Responder’s MAC address.

To select, configure, and start a SCCP Test:

1. From the Main Menu, tap SCCP.

2. From the Interface tab, configure the test interface parameters. See

Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step.

3. From the Configuration tab, configure the SCCP test. See Configuration (SCCP) on page 32.

4. From the Parameters tab, configure the call, phone, and media parameters. See Parameters (SCCP) on page 48.

5. From the Threshold tab, enable and configure the threshold values. See Threshold on page 52.

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Selecting and Starting a Test

SCCP

6. Press Start to start the test.

7. For results, refer to Summary (SCCP) on page 79, Call Quality Details

on page 62, and Call Manager Statistics (SCCP) on page 60.

8. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.

16 EXpert VoIP Test Tools

Selecting and Starting a Test

H.323

The H.323 test is a set-to-service test that measures call statistics for an H.323 call. It either makes a H.323 call to a given destination (directly to an endpoint, or through the gatekeepers and/or gateways that you specify) or answers a call made to it from an endpoint, and reports call setup and media path statistics. The test assumes that you know the destination (specified using an IP address, telephone number, or H.323 ID), and that the destination will answer the call and will not hang up during the configured call duration. The H.323 test supports versions 1 through 4 of the H.323 protocol.

To select, configure, and start a H.323 Test:

1. From the Main Menu, tap H.323.

2. From the Interface tab, configure the test interface parameters. See

Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step.

3. From the Configuration tab, configure the H.323 test. See

Configuration (H.323) on page 35.

H.323

4. From the Parameters tab, configure the local, and media parameters. See Parameters (H.323) on page 49.

5. From the Threshold tab, enable and configure the threshold values. See Threshold on page 52.

6. Press Start to start the test.

7. For results, refer to Summary (H.323) on page 73, Call Quality Details

on page 62, and Call Manager Statistics (H.323) on page 58.

8. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.

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Selecting and Starting a Test

H.248/Megaco

The H.248, the Media Gateway Control Protocol, is designed for distributed media gateway systems in which one or more centralized gateway controllers (Call Agents) control one or more media gateways.

The VoIP H.248 test measures the performance of the services of H.248 based VoIP networks. It either initiates a call to or answers a call from other endpoints based on a telephone number. Performance is measured for both the call signaling path (H.248 signaling) and the media path (RTP/RTCP). Each set of measurements is performed during a test cycle. A test cycle consists of the initiation and termination of a single, bi-directional RTP audio session between the test set and a third-party device.

To select, configure, and start a H.248/Megaco Test:

1. From the Main Menu, tap H.248/Megaco.

2. From the Interface tab, configure the test interface parameters. See

Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step.

3. From the Configuration tab, configure the H.248/Megaco test. See Configuration (H.248/Megaco) on page 39.

4. From the Parameters tab, configure the local parameters. See Parameters (H.248/Megaco) on page 51.

5. From the Threshold tab, enable and configure the threshold values. See Threshold on page 52.

6. Press Start to start the test.

7. For results, refer to Summary (H.248/Megaco) on page 71, Call Quality

Details on page 62, and Call Manager Statistics (H.248/Megaco) on page 56.

8. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.

18 EXpert VoIP Test Tools

5Test Setup

The test setup offers the following tabs depending on the test:

Tabs

RTP SIP SCCP H.323 H.248

Interface XXXXX20

Configuration 23 26 32 35 39 <---

Controller Setup X 42

Responder Setup X 45

Parameters 48 49 51 <---

Threshold XXXXX52

Tes t

Page

EXpert IP 19

Test Setup

Interface

Note: The interface configuration parameters apply globally to all tests.

From the Main Menu, tap on a test and the Interface tab.

Test Int erface

 Tes t Int er fa ce displays the network adapter(s) found on the EXFO’s

platform. The first network adapter is selected by default.

 Speed/Duplex selects the network adapter speed and duplex mode:

Auto Negotiation (default), 10 Mbit/s Half Duplex, 10 Mbit/s Full Duplex, 100 Mbit/s Half Duplex, 100 Mbit/s Full Duplex, 1000 Mbit/s Full Duplex. Changing this

setting restarts the network adapter, a user confirmation is required.

Obtain IP from DHCP

Obtain IP from DHCP check box when selected (default) allows to dynamically obtain an IP address from a DHCP (Dynamic Host Configuration Protocol) server for the Ethernet port.

VLAN Tagging

Allows to enable or disable VLAN Tagging. VLAN Tagging check box must be selected (default) for all test tools at the exception of the VLAN Scan test tool for which VLAN Tagging must be disabled.

Note: VLAN Tagging enable/disable will reboot the device. A warning pop-up

message is displayed. Press Yes to reboot the platform. The EXpert VoIP Test Tools application has to be manually restarted after the platform’s reboot.

20 EXpert VoIP Test Tools

Test Setup

Interface

Network Configuration

Note: Network configuration is only possible when the Obtain IP from DHCP

check box is cleared. Otherwise, when the Obtain IP from DHCP check box is selected, all parameters are not configurable and set to the values obtained through DHCP process.

 IP Address allows to enter the IP address for the Ethernet port: 0.0.0.1

to 223.255.255.255.

 Subnet Mask allows to enter the Subnet Mask for the Ethernet port:

0.0.0.1 to 255.255.255.255.

 Default Gateway allows to enter a default Gateway address: 0.0.0.01

to 255.255.255.255.

 DNS Server allows to enter the DNS Server address: 0.0.0.0 to

223.255.255.255.

 Refresh, available when the Obtain IP from DHCP check box is

selected, refreshes the network configuration parameters.

 Apply allows to apply the network configuration settings.

EXpert IP 21

Test Setup

Interface

Ethernet Statistics

The following Ethernet Statistics are displayed.

 Link Status displays the status of the link: Connected or

Disconnected.

 Link Speed (Mbps) displays the speed of the Ethernet connection.

 Domain Name displays the name of the connected domain.

 Frame TX displays the total Ethernet frames transmitted.

 Frame RX displays the total Ethernet frames received.

 Duplex displays the Duplex mode of the link: half duplex, full duplex,

or auto.

 Errored Frames displays the number of Ethernet frames received with

error.

 Dropped Frames displays the number of Ethernet frames dropped.

 Bytes TX displays the total data bytes transmitted.

 Bytes RX displays the total data bytes received.

 MAC Address displays the MAC Address of the EXFO’s platform.

Note: Bytes TX and Bytes RX indicate the total amount of data that has passed

through the Ethernet interface either way. EXpert VoIP Test Tools update the Ethernet statistics every second.

22 EXpert VoIP Test Tools

Test Setup

Configuration (RTP)

From the Main Menu, tap RTP and the Configuration tab.

Call Mode

Select the call mode:

 Controller establishes a control communications channel with the

Responder. It also sends the configuration details to the Responder. By default, the Controller is selected.

 Responder (default) is the endpoint, which initiates its own stream to

the Controller test set. When Responder is selected, all other configuration parameters are disabled.

Test Length

Select the test length:

 Tes t Dur at ion (se c ) is the length of time the test will run in seconds:

1 to 600 seconds (default is 10 seconds).

 Number of Packets is the number of RTP packets to exchange

between the controller and the responder: 2 to 65535 (default is 500).

Call Information

 Media Type

 Live Call (default) specifies that actual call is established between

the Controller and the Responder.

 Synthesized Media specifies that EXpert IP Test Tools application

automatically generates a randomly RTP data.

EXpert IP 23

Test Setup

Configuration (RTP)

 Codec represents the audio Codec (packet interval, payload size, and

 IP Media DSCP

Destination IP address

 IP Address

IP bandwidth) of the call simulated by the Controller.

For Live Call media type: G.711 μlaw, and G.711 aLaw (default).

For Synthesized Media type: G.711 μlaw, G.711 aLaw (default), G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.

(IP Media Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end. The accepted range is from 0 (default) to 63.

 Manually Configure IP allows to configure the destination IP

address manually: 0.0.0.1 to 223.255.255.255.

 Auto Discovery List allows to select the IP address from a list of IP

addresses discovered by the application.

Utilities

 Traceroute check box when selected (default) calculates the

performance of the network path between two endpoints of a running test every time the test is run.

 Capture check box when selected (cleared by default) captures the

RTP packet stream for the running test. Once test is completed, the capture file is available and can be opened in the Wireshark application from the Load/Save button; refer to Save/Load - Packet Capture on page 87 for more information.

1. Not supported on FTB-2 and FTB-2 Pro.

24 EXpert VoIP Test Tools

Test Setup

Configuration (RTP)

 Record check box when selected (cleared by default) allows

recording network VoIP traffic. The recorded file can be listened from the Load/Save option; refer to Save/Load - Record Audio Tab on page 87 for more information. Record is only available for Live Call media type.

Capture Filters

Available when the Utilities - Capture check box is selected.

 IP Address check box when selected (cleared by default) represents

the Peer IP Address that acts as a filter that will capture packets having the source or destination IP address matching the specified Peer IP address. The accepted range is from 0.0.0.1 to 223.225.225.225.

 File Size represents the maximum size of the file to capture either in

megabytes (1 to 50 MB (default is )) or in seconds (1 (default) to 3600 seconds).

EXpert IP 25

Test Setup

Configuration (SIP)

From the Main Menu, tap SIP and the Configuration tab.

Peer Device

 EXFO Device: Select EXFO Device option on both the endpoints (i.e.

 Other Device (default): Select the Other Device option to run the

Call Mode

 Controller when selected (default) establishes a control

Controller side and the Responder side) to run the SIP Network Active Tes t.

service active test. In service active test, the test set can either act as controller or responder with the other end being the SIP endpoint. By default, Other Device is selected.

communications channel with the Responder and also sends the configuration details to the Responder.

 Responder: If selected all the other configurations are disabled (only

for Network Active test). In the Service Active test, the configurations are available at responder side also. The Responder is the endpoint, which initiates its own stream to the Controller test set.

Communication Channel

 TCP selects the Transmission Control Protocol (TCP) for the

transmission of the data on the network.

 UDP selects the User Datagram Protocol (UDP) for the transmission of

the data on the network.

Note: Communication channel (TCP/UDP) is used for SIP signaling packets only.

Voice data will pass through UDP only even though TCP is selected. SIP calls over TCP must be routed through a proxy server.

26 EXpert VoIP Test Tools

Test Setup

Configuration (SIP)

Test Length

 Tes t Dur at ion (sec) is the length of time for which the test will run in

seconds: 1 to 600 seconds (default is 10 seconds). Tes t Len gth is not available when Stored Audio File media type is selected.

 Number of Packets is the number of RTP packets to exchange

between the Controller and the Responder: 2 to 65535 (default is 500).

Call Information

 Media Type

 Live Call (default) specifies that actual call is established between

the Controller and the Responder.

 Stored Audio File specifies that any stored media file on the drive

is transmitted from Controller to Responder and the results are recorded.

 Synthesized Media specifies that the EXpert IP Test Tools

application automatically generates a randomly RTP data.

 Codec selects the audio Codec (packet interval, payload size, and IP

bandwidth) of the call simulated by the Controller.

For Live Call media type: G.711 μlaw (default), and G.711 aLaw.

For Audio File media type: G.711 μlaw (default), G.711 aLaw, G729, G.729A, G.729B, and G.729AB.

For Synthesized Media type: G.711 μlaw (default), G.711 aLaw, G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.

EXpert IP 27

Test Setup

Configuration (SIP)

 Call Type selects the type of the call. For Peer Device set as EXFO

 PRACK (Provisional Response ACKnowledgement) method is used to

Device, only SIP Call is available. In Network Active test, the supported call type is SIP Call.

 SIP Call attempts to establish a single call with a single audio

session.

 SIP Options Request attempts a single “options” request method

with the specified endpoint. Use this mode to test the availability of the SIP endpoint without making a complete call.

 SIP Proxy Registration attempts a single proxy “register” request

method with the specified registrar domain and proxy server.

 SIP Call After Proxy Registration acts as a SIP Call, but includes

data acquired during registration with a specified registrar domain and proxy server.

acknowledge receipt of reliably transported provisional responses (1xx). The reliability of 2xx, 3xx, 4xx, 5xx, and 6xx responses to INVITE requests are achieved using the ACK method. However, in cases where a provisional response such as 180 ringing is critical in determining the call state, it may be necessary for the receipt of a provisional response to be confirmed. The PRACK method applies to all provisional responses except for the 100 trying response which is never reliably transported.

28 EXpert VoIP Test Tools

Test Setup

Configuration (SIP)

For Controller Call Mode:

 Disabled (default) doesn‘t use the PRACK method.

 Supported does not insist on usage of reliable provisional

responses. However, the UAC supports the reliable provisional responses in case where the UAS needs to send one. The UAC creates a request by inserting a Supported header field into the request.

 Required insists on reliable delivery of provisional responses. The

UAC creates a request by inserting a Require header field into the request.

For Responder Call Mode:

 PRACK check box when selected (default) allows to respond to

Provisional Response ACKnowledgement by providing reliable provisional response messages.

 SIP DSCP

(SIP Differentiated Services Code Point) describe a set of QoS (Quality of service) capabilities. The accepted range is from 0 (default) to 63.

 IP Media DSCP

(IP Media Differentiated Services Code Point)

describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end. The accepted range is from 0 (default) to 63.

1. Not supported on FTB-2 and FTB-2 Pro.

EXpert IP 29

Test Setup

Configuration (SIP)

Destination IP address

Only available when the Peer Device is set as EXFO Device.

 Manually Configure IP allows to configure the destination IP address

 Auto Discovery List allows to select the IP address from a list of IP

 IP address allows to either enter the IP address manually (from 0.0.0.1

Utilities

 Capture check box when selected (cleared by default) captures the

 Record check box when selected (cleared by default) allows

manually.

addresses discovered by the EXpert VoIP Test Tools.

to 223.255.255.255) or from a list.

RTP packet stream for the running test. Once test is completed, the capture file is available and can be opened in the Wireshark application from the Load/Save option; refer to Pac k et Ca pt ure on page 87 for more information.

recording network VoIP traffic. The recorded file can be listened from the Load/Save option; refer to Record Audio Tab on page 87 for more information. Record is only available for Live Call media type.

Capture Filters

Available when the Utilities - Capture check box is selected.

 IP Address check box when selected (cleared by default) uses the

Peer IP Address that acts as a filter that will capture packets having the source or destination IP address matching the specified Peer IP address. The accepted range is from 0.0.0.1 to 223.225.225.225.

 File Size represents the maximum size of the file to capture either in

megabytes (1 to 50 MB (default is )) or in seconds (1 (default) to 3600 seconds).

30 EXpert VoIP Test Tools

Test Setup

Configuration (SCCP)

From the Main Menu, tap SCCP and the Configuration tab.

Call Mode

 Make Call: The Caller establishes a control communications channel

with the Receiver. By default, Make Call is selected.

 Answer Call: It is the endpoint, which initiates its own stream to the

Caller test set.

Test Length

Select the test length. Not available when Stored Audio File is selected as the media type.

 Tes t Dur at ion (se c ) is the length of time the test will run in seconds:

1 to 600 seconds (default is 10 seconds).

 Number of Packets is the number of RTP packets to exchange

between the controller and the responder: 2 to 65535 (default is 500).

Call Information

 Media Type

 Live Call (default) specifies that actual call is established between

the Controller and the Responder.

 Stored Audio File specifies that any stored media file on the drive

is transmitted from Controller to Responder and the results are recorded.

 Synthesized Media specifies that the EXpert IP Test Tools

application automatically generates a randomly RTP data.

EXpert IP 31

Test Setup

Configuration (SCCP)

 Codec selects the audio Codec (packet interval, payload size, and IP

 SCCP DSCP

Destination Information

Destination Phone Number is the phone number associated with the MAC address for the Callee as specified in the CallManager registration tables. This number is dialed by the Caller at the start of each test. The accepted range is from 1 to 9999999999. Only available when Make Call is selected as the call mode.

bandwidth) of the call simulated by the Caller.

For Live Call media type: G.711 μlaw (default), and G.711 aLaw.

For Audio File media type: G.711 μlaw (default), G.711 aLaw, G729, G.729A, G.729B, and G.729AB.

For Synthesized Media type: G.711 μlaw (default), G.711 aLaw, G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.

(SCCP Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end. The accepted range is from 0 (default) to 63.

Note: The destination phone number must be assigned by the CallManager

administrator in order to ensure compliance with the network standard.

1. Not supported on FTB-2 and FTB-2 Pro.

32 EXpert VoIP Test Tools

Test Setup

Configuration (SCCP)

Utilities

 Traceroute check box when selected (default) calculates the

performance of the network path between two endpoints of a running test every time the test is run.

 Capture check box when selected (cleared by default) captures the

 Record check box when selected (cleared by default) allows

Capture Filters

Available when the Utilities - Capture check box is selected.

 IP Address check box when selected (cleared by default) represents

 File Size represents the maximum size of the file to capture either in

megabytes (1 to 50 MB (default is )) or in seconds (1 (default) to 3600 seconds).

EXpert IP 33

Test Setup

Configuration (H.323)

From the Main Menu, tap H.323 and the Configuration tab.

Call Mode

 Make Call: The Caller establishes a control communications channel

with the Receiver. By default, Make Call is selected.

 Answer Call: It is the endpoint, which initiates its own stream to the

Caller test set.

Call Information

 Media Type

 Live Call (default) specifies that actual call is established between

 Synthesized Media specifies that the EXpert IP Test Tools

the Caller and the Callee.

application automatically generates a randomly RTP data.

 Codec selects the audio Codec (packet interval, payload size, and IP

bandwidth) of the call simulated by the Caller.

For Live Call media type: G.711 μlaw (default), and G.711 aLaw.

For Synthesized Media type: G.711 μlaw (default), G.711 aLaw, G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.

Note: If the call endpoint does not support the specified codec, the call fails.

34 EXpert VoIP Test Tools

Test Setup

Configuration (H.323)

DSCP

IP Media DSCP (IP Media Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end. The accepted range is from 0 (default) to 63.

Call Duration

Call Duration (sec) is the length of time for the H.323 session. The accepted range is from 1 to 3600 seconds (default is 10). Only available when Make Call is selected as call mode.

Destination Information

 Destination Phone specifies the destination of the H.323 call. Select

the destination phone and its value (a maximum 64 characters are allowed). If gatekeeper details are present, select H.323 ID or E.164 ID otherwise the call will fail.

 H.323 ID specifies the call destination's H.323 ID address. The

H.323 ID is a dialing attribute of an H.323 endpoint. For example, if the H.323 endpoint is a softphone, then the H.323 Id will be H.323 extension or H.323 phone number’ of the H.323 softphone.

 E.164 specifies the destination phone number. The E.164 is an

International Public Telecommunications Numbering Plan. The E.164 ID is a dialing attribute of E.164 or an E.164 extension of H.323 endpoint.

1. Not supported on FTB-2 and FTB-2 Pro.

EXpert IP 35

Test Setup

Configuration (H.323)

 Transport Address specifies the call destination’s transport

Utilities

 Traceroute check box when selected (default) calculates the

performance of the network path between two endpoints of a running test every time the test is run.

 Capture check box when selected (cleared by default) captures the

RTP packet stream for the running test. Once test is completed, the capture file is available and can be opened in the Wireshark application from the Load/Save option; refer to Save/Load - Packet Capture on page 87 for more information.

address. Enter the IP address with Q.931 TCP server port number or DNS name with Q.931 TCP server port number of the called H.323 terminal. The TCP server port number is optional. The format used for destination address transport is <IP address or DNS name>[:port]. The default port number is 1720. For example

192.168.1.150 is equivalent to 192.168.1.150:1720, and yourname.com is equivalent to yourname.com:1720.

 Record check box when selected (cleared by default) allows

36 EXpert VoIP Test Tools

Test Setup

Configuration (H.323)

Capture Filters

Available when the Utilities - Capture check box is selected.

 IP Address check box when selected (cleared by default) represents

 File Size represents the maximum size of the file to capture either in

megabytes (1 to 50 MB (default is )) or in seconds (1 (default) to 3600 seconds).

EXpert IP 37

Test Setup

Configuration (H.248/Megaco)

From the Main Menu, tap H.248/Megaco and the Configuration tab.

Call Mode

 Make Call (default): The Caller establishes a control communications

channel with the Receiver. By default, Make Call is selected.

 Answer Call: It is the endpoint, which initiates its own stream to the

Caller test set.

Test Length

Select the test length.

 Media Duration (sec) is the length of time the test will run in seconds:

1 to 600 seconds (default is 10 seconds).

 Number of Packets is the number of RTP packets to exchange

between the controller and the responder: 2 to 65535 (default is 500).

Call Information

 Media Type

 Live Call (default) specifies that actual call is established between

the Caller and the Callee.

 Synthesized Media specifies that the EXpert IP Test Tools

application automatically generates a randomly RTP data.

 Codec selects the audio Codec (packet interval, payload size, and IP

bandwidth) used to transmit audio data within the RTP stream.

For Live Call media type: G.711 μlaw (default), and G.711 aLaw.

For Synthesized Media type: G.711 μlaw (default), G.711 aLaw, G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.

38 EXpert VoIP Test Tools

Test Setup

Configuration (H.248/Megaco)

 H.248 DSCP

(H.248 Differentiated Services Code Point) describe a set of QoS (Quality of service) capabilities. The accepted range is from 0 (default) to 63.

 IP Media DSCP

Destination Information

Destination Phone is the phone number of the Callee: 1 to

99999999999999. Only available when Make Call is selected as the call

mode.

Utilities

 Capture check box when selected (cleared by default) captures the

 Record check box when selected (cleared by default) allows

1. Not supported on FTB-2 and FTB-2 Pro.

EXpert IP 39

Test Setup

Configuration (H.248/Megaco)

Capture Filters

Available when the Utilities - Capture check box is selected.

 IP Address check box when selected (cleared by default) represents

 File Size represents the maximum size of the file to capture either in

megabytes (1 to 50 MB (default is )) or in seconds (1 (default) to 3600 seconds).

40 EXpert VoIP Test Tools

Test Setup

Controller Setup

Note: Only available when Controller is selected as the call mode (see Call Mode

on page 26).

From the Main Menu, tap SIP and the Controller Setup tab.

Controller Parameters

 User Name is the name used for the SIP URL associated with the

Controller. A maximum 50 characters are allowed.

 Signaling Port is the signaling port of the Controller: 1000 to 65535

(default is 5060).

 Domain Name is the DNS name of the Controller, specified in either

DNS name format or IP address format. A maximum 60 characters are allowed. If the DNS domain or IP address is specified, this value is used as the host portion of the caller’s SIP URL, otherwise the IP address of the caller is used.

 Media Mode defines the flow direction of the RTP packets (media).

 Send and Receive (default) specifies that the RTP packets flows in

both directions. For example, if Controller is set in Send and receive media mode then it will send RTP Packets to Responder and also it will receive the RTP packets from Responder.

 Receive Only specifies that the RTP packets flows only in one

direction, i.e. from Responder to Controller.

EXpert IP 41

Test Setup

Controller Setup

Proxy Settings

 Force Proxy

If the Force Proxy check box is selected, the test routes all requests associated with each call through the specified proxy server. The test ignores any Record-Route or contact information contained in the final response to the initial “SIP invite” request method.

If the Force Proxy check box is cleared (default), the test routes subsequent requests in one of the following ways.

 Using the Record-Route path, if specified

 Directly to the peer Responder via the contact information

 Proxy IP Address/DNS: Port is the proxy IP address or DNS. IP address

can be in a dotted decimal notation (for example, 192.168.34.5) or in a DNS name (for example, cm.acme.com) format. A maximum 60 characters are allowed.

 Authentication User Name is the user name for the authentication for

the Controller, if the proxy server or registration server requires authentication. A maximum 50 characters are allowed.

 Authentication Password is the password for the authentication for

the Controller, if the proxy server or registration server requires authentication. A maximum 64 characters are allowed.

42 EXpert VoIP Test Tools

Test Setup

Controller Setup

Registration Parameters

Note: Available when Other Device is selected as Peer Device (see Peer Device

on page 26).

 Specify Registrar specifies SIP Registrar or Registration-Server. When

the Specify Registrar check box is selected (cleared by default), SIP registration requests are accepted and processed by the specified SIP Registrar.

 Registrar IP Address/DNS: Port: Option for the Controller to attempt

to register itself every 60 minutes within the specified registrar domain using the specified proxy server. If selected Call Type under Configuration tab is SIP Call After Proxy Registration or SIP Proxy Registration, the Controller attempts to register itself within the specified registrar domain using the specified proxy server. If this parameter is not provided, it defaults to the Proxy IP address. A maximum 60 characters are allowed.

EXpert IP 43

Test Setup

Responder Setup

Note: Not available when both EXFO Device (see Peer Device on page 26) and

Responder (see Call Mode on page 26) are selected.

From the Main Menu, tap SIP and the Responder Setup tab.

Responder Parameters

 User Name is the name of the Responder, which is used for the SIP

URL associated with the Responder. A maximum of 50 characters are allowed.

 Domain Name is the DNS name used as the host portion of the

Responder’s SIP URL. A maximum of 60 characters are allowed.

 Signaling Port is the signaling port of the Responder: 1000 to 65535

(default is 5060).

 Media Mode defines the flow direction of the RTP packets (media).

 Send and Receive (default) specifies that the RTP packets flows in

both directions. For example, if Controller is set in Send and receive media mode then it will send RTP Packets to Responder and also it will receive the RTP packets from Responder.

 Receive Only specifies that the RTP packets flows only in one

direction, i.e. from Responder to Controller.

44 EXpert VoIP Test Tools

Test Setup

Responder Setup

Proxy Settings

 Force Proxy

If the Force Proxy check box is selected, the test routes all requests associated with each call through the specified proxy server. The Responder ignores any Record-Route or contact information contained in the final response to the initial “SIP invite” request method.

If the Force Proxy check box is cleared (default), the test routes subsequent requests in one of the following ways.

 Using the Record-Route path, if specified

 Directly to the peer endpoint via the contact information

 Proxy IP Address/DNS: Port is the proxy IP address or DNS. IP address

can be in a dotted decimal notation (for example, 192.168.34.5) or in a DNS name (for example, cm.acme.com) format. A maximum 60 characters are allowed.

 Authentication User Name is the user name for the authentication for

the Controller, if the proxy server or registration server requires authentication. A maximum 50 characters are allowed.

 Authentication Password is the password for the authentication for

the Responder, if the proxy server or registration server requires authentication. A maximum 64 characters are allowed.

EXpert IP 45

Test Setup

Responder Setup

Registration Parameters

Note: Available when Other Device is selected as Peer Device (see Peer Device

on page 26).

 Register check box when selected (cleared by default) specifies that

the SIP-Responder will register with a SIP Proxy (or SIP Registrar -if specified) during every test run.

 Specify Registrar specifies SIP Registrar or Registration-Server. When

the Specify Registrar check box is selected (cleared by default), SIP registration requests are accepted and processed by the specified SIP Registrar.

 Registrar IP Address/ DNS: Port: Enter the IP Address to register. If

enabled, the Responder attempts to register itself within the specified registrar domain using the specified proxy server. If disabled, the Responder does not attempt to register itself within the specified registrar domain using the specified proxy server. When the Registrar IP Address/ DNS: Port field is selected for editing using virtual keyboard, the Presets button appears on the screen. Press the Presets button to display the list of previously configured values. Select the desired value from the list, if any. Maximum 60 characters are allowed.

 Registration Interval (Sec): Enter the proxy registration refresh

interval in seconds. When the Registration Interval field is selected for editing using virtual keyboard, the Presets button appears on the screen. Press the Presets button to display the list of previously configured values. Select the desired value from the list, if any. The accepted range is from 1 to 65527 seconds. The default value is 3600 seconds.

46 EXpert VoIP Test Tools

Test Setup

Parameters (SCCP)

From the Main Menu, tap SCCP and the Parameters tab.

Call Parameters

 Call Manager IP Address/DNS allows to enter the IP Address or DNS of

the Call Manager. IP address can be in a dotted decimal notation (for example, 192.168.34.5) or in a DNS name (for example, cm.acme.com) format. A maximum of 56 characters are allowed.

 Local MAC Address allows to enter the MAC Address associated with

the Callee as specified in the CallManager registration tables. This need not be the Callee’s actual MAC address, nor the address of an actual network device. However, it should have a valid MAC Address format. The accepted range is from 00:00:00:00:00:00 to FF:FF:FF:FF:FF:FF. The MAC Address must be unique across all tests using the same CallManager.

Phone Type

Phone Type is the Phone Type that the answering endpoint will emulate: 12 S (default), 12 SP, 12 SP+, 30 SP+, 30 VIP, Cisco IP Phone 7910, Cisco IP Phone 7935, Cisco IP Phone 7940, and Cisco IP Phone 7960.

Media Parameters

Media Mode defines the flow direction of the RTP packets (media).

Send and Receive specifies that the RTP packets will flow in both

directions. For example, if Controller is set in Send and receive media mode then it will send (will send RTP as Audio Codec is selected) RTP Packets to Responder and also it will receive the RTP packets from Responder.

EXpert IP 47

Test Setup

Parameters (H.323)

From the Main Menu, tap H.323 and the Parameters tab.

Local Parameters

 Gatekeeper IP Address/DNS allows to enter the IP address, or DNS

 Gatekeeper Port allows to enter the port number of the Gatekeeper:

 Local Phone allows to select the Local Phone number to use when

name, or optionally, the UDP RAS port number of the H.323 Gatekeeper. A maximum of 56 characters are allowed.

1000 to 65535 (default is 1719).

registering with the Gatekeeper then enter the phone number of the selected phone type. A maximum of 64 characters are allowed.

 H.323 ID: Select the Local H.323 ID address to use when

registering with the Gatekeeper. The H.323 ID is a dialing attribute of an H.323 endpoint. For example, if the H.323 endpoint is a softphone, then the H.323 Id will be H.323 extension or H.323 phone number’ of the H.323 softphone.

 E.164 ID: Select the Local E.164 ID address to use when registering

with the Gatekeeper. The E.164 is an International Public Telecommunications Numbering Plan. The E.164 ID is a dialing attribute of E.164 or an E.164 extension of H.323 endpoint.

Note: Gatekeeper details are mandatory to make a call with E.164 ID or H.323 ID.

48 EXpert VoIP Test Tools

Test Setup

Parameters (H.323)

Media Parameters

 Media Mode defines the flow direction of the RTP packets (media).

 Send and Receive (default) specifies that the RTP packets flows in

both directions. For example, if Controller is set in Send and receive media mode then it will send RTP Packets to Responder and also it will receive the RTP packets from Responder.

 Receive Only specifies that the RTP packets flows only in one

direction, i.e. from Responder to Controller or Controller to Responder.

EXpert IP 49

Test Setup

Parameters (H.248/Megaco)

From the Main Menu, tap H.248/Megaco and the Parameters tab.

Local Parameters

 H.248 End Point Name allows to enter the H.248 End Point Name. The

End Point Name should be registered with the end point location (i.e. IP address) and the phone number present in the Call Agent or Media Gateway Controller. For example, if the End Point Name is registered with the abc/1 and abd/1, then you can enter abc and abd in the H.248 End Point Name field. A maximum of 64 characters are allowed.

 H.248 Gateway IP Address/DNS allows to enter the H.248 Gateway IP

Address/DNS name of the Media Gateway Controller or the Call Agent. The Media Gateway Controller should be configured with the end point name, end point location, and the end point phone number. A maximum of 60 characters are allowed.

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Test Setup

Threshold

The threshold is used to specify the Pass and Fail values for the test.

From the Main Menu, tap any test and the Threshold tab.

VoIP Pass/Fail Thresholds

 Select All enables all the threshold parameters.

 Maximum Jitter (ms): Enter the maximum Jitter value: 0 (default) to

125 milliseconds. The test fails if the average jitter is greater than the configured value.

 Maximum Latency (ms): Enter the Maximum Latency value:

0 (default) to 600 milliseconds. The test fails if the average latency is greater than the configured value.

 Maximum Packet Loss (%): Enter the maximum Packet Loss

percentage value: 0 (default) to 100 %. The test fails if the average packet loss (%) is greater than the configured value.

Threshold

 Minimum Conversational MOS: Enter the minimum Conversational

MOS value: 0 (default) to 5. The Conversational Voice Quality Mean Opinion Score (MOS) for the RTP stream going from the Controller to the Responder, if the thresholds setting are at Controller then the RTP stream will flow from Responder to Controller, and if the thresholds setting are at Responder then the RTP stream will flow from Controller to Responder. The test fails if the average MOS is less than the configured value.

 Minimum User R-Factor: Enter the minimum User R-Factor value:

0 (default) to 100. The Conversational Voice Quality R factor for the RTP stream going from the Controller to the Responder, if the thresholds setting are at Controller then the RTP stream will flow from Responder to Controller, and if the thresholds setting are at Responder then the RTP stream will flow from Controller to Responder. The test fails if the average R-Factor is less than the configured value.

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6Test Results

The rest Results menu offers the following Result Summary tabs depending on the test:

Tabs

RTP SIP SCCP H.323 H.248

Call Details X 54

Call Manager Statistics 60 58 56 <---

Call Quality Details X X X 62

Controller Statistics X X 65

Responder Statistics X X 68

Summary 76 81 79 73 71 <---

Tes t

Page

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Test Results

Call Details

From the Test s menu, tap Results and the Call Details tab.

Test Results

 Endpoint IP Address displays the IP Address of the endpoint device.

 Peer URL displays the URL of the endpoint device.

 Response is the method and response string of the last SIP message

received by the test.

 Response Code is the SIP Response code that the destination returned

during the last SIP transaction.

Controller Results

 No of Packets Sent indicates the number of packets sent to the

responder.

 Round Trip Latency indicates respectively the length of time in

milliseconds for the minimum, maximum, and average round-trip latency.

Call Setup and Termination Time (ms)

 Option Response Time (ms) indicates the time in milliseconds to

complete the “Sip options” request method.

Note: The Option Response Time result is valid only, if SIP Options Request is

selected under Call Type in Call Information in the Configuration tab.

 Initial Response Time (ms) indicates the time measured in

milliseconds from the initial SIP request message sent by the test until the first response is received.

 Call Termination Time (ms) indicates the length of time in

milliseconds to complete the call termination.

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Call Details

 Registration Time (ms) indicates the time in milliseconds to complete

a SIP “register request” method.

 Authentication Time (ms) indicates the time in milliseconds to

complete the proxy authentication during call initiation.

Note: The Authentication Time result is valid only, if SIP Proxy Registration Only

or SIP Call After Proxy Registration is selected under Call Type in Call Information in the Configuration tab.

 Post Dial Delay (ms) indicates the time measured in milliseconds

from the transmission of the “SIP invite” request method to the reception of first provisional or final response message from the Responder (180 Ringing, for example).

 Call Setup Time (ms) indicates the length of time in milliseconds to

complete the call initiation from the Controller to the Responder (from the reception of “invite” request method to the reception of “ack” response method).

 Post Pickup Delay (ms) indicates the post pickup delay measured by

the test Responder in milliseconds. This is defined as the difference between the transmission of the SIP OK message and the reception of the first media (RTP) packet from the Controller.

 Media Start Time (ms) indicates the time in milliseconds from dialing

to the reception of the first media packet from the called party.

 Call Terminated By indicates whether the call was terminated by the

Controller or the endpoint.

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Test Results

Call Manager Statistics (H.248/Megaco)

Displays the performance statistics of the Call Manager.

From the Test s menu, tap Results and the Call Manager Statistics tab.

Result Summary Metrics

 Peer Phone Number is the phone number used by the endpoint in

Caller mode.

 Peer IP Address is the IP address of the peer endpoint.

Note: RTCP (Real-time Transport Control Protocol) provides out of band statistics

and control information for an RTP flow.

 Peer RTP Port is the UDP port number used by the peer endpoint for

media packet reception. The corresponding UDP port number for RTCP is 1 greater than this value.

 Local RTP Port is the UDP port number used by the test for media

packet reception. The corresponding UDP port number for RTCP is 1 greater than this value.

 Local Name is the endpoint name used in the Media Gateway

Controller.

Call Manager Performance Statistics

 Off Hook Time (ms) indicates the time in milliseconds between the

transmission of the Notify (Off Hook) message and the reception of the CreateConnection (dialtone) message from the Media Gateway Controller.

 Post Dial Delay (ms) indictes the length of time in milliseconds from

the start of the call to either the ringing or connection of the called terminal, whatever is earlier.

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Test Results

Call Manager Statistics (H.248/Megaco)

 Post Pickup Delay (ms) is the post pickup delay measured by the

Caller in milliseconds. This value is the difference between the transmission of the Off Hook Notification message by the test and the reception of the first media (RTP) packet from the endpoint. A negative value indicates that the media arrived early (audio clipping).

 Total Dial Delay (ms) is the time in milliseconds to complete the call

initiation as seen by the Caller. This is defined as the difference between the transmission of the Off Hook Notification message by the test and the reception of the first On Hook Notification Request from the Call Agent.

 Media Start Time (ms) is the time in milliseconds from the dialing to

the reception of the first media packet from the Callee.

 Call Termination Time (ms) is the time in milliseconds to complete

the call initiation as seen by the Caller. This is defined as the difference between the transmission of the response to the Call Agent's Delete Connection command and the reception of the next Off Hook Notification Request from the Call Agent.

 Number of Notified Entities is the total number of notified entities

(Call Agents) that the test endpoint is connected to during the test.

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Test Results

Call Manager Statistics (H.323)

Displays the performance statistics of the Call Manager.

From the Test s menu, tap Results and the Call Manager Statistics tab.

General Results

 Endpoint Display Information is provided by the peer endpoint for

the call.

 Endpoint Phone Number is the called number used for this call.

 Receiving Audio Codec is the Audio Codec used for the inbound audio

channel from the H.323 endpoint to the Caller.

 Sending Audio Codec is the Audio Codec used for the outbound audio

channel from the Caller to the H.323 endpoint.

 Registration Failure Reason is the error message received from the

CallManager, if the registration is failed.

Call Manager Performance Statistics

 Post Dial Delay (ms) is the length of time in milliseconds from the start

of the call to either the ringing or connection of the called terminal, whatever is earlier.

 Media Setup Delay (ms) is the length of time in milliseconds for the

media to setup after the call is answered.

 Total Call Setup Delay (ms) is the aggregate amount of time it takes to

setup the call. This statistic is a summation of the following delay components: Gatekeeper Discovery Delay, Gatekeeper Registration Delay, Gatekeeper Admission Delay, Post-Dial Delay, Ring Duration, and Media Setup Delay. This result is measured with microsecond accuracy.

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Test Results

Call Manager Statistics (H.323)

 Ring Duration (ms) is the length of time in milliseconds for a called

terminal to answer the call after it rings.

 Media Duration (sec) is the length of time in seconds for the duration

of the test.

Gatekeeper Results

 Gatekeeper Address displays the Gatekeeper DNS or IP Address

through which the call happens.

 Discovery Delay (ms) is the length of time in milliseconds for the

Gatekeeper Discovery Delay.

 Registration Delay (ms) is the length of time in milliseconds for the

Gatekeeper Registration Delay.

 Admission Delay (ms) is the length of time in milliseconds for the

Gatekeeper call Admission Delay.

 Gatekeeper RAS Port is the RAS port number of the Gatekeeper that

was used.

 Local RAS Port is the RAS port number of the local endpoint.

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Test Results

Call Manager Statistics (SCCP)

Displays the performance statistics of the Call Manager.

From the Test s menu, tap Results and the Call Manager Statistics tab.

Call Statistics

 Assigned Phone Number is the phone number assigned to the

endpoint (caller) by the Call Manager during registration.

 Peer Phone Number is the phone number used by the endpoint in

Caller mode to reach the peer endpoint.

 Peer Display Information is the peer endpoint’s display information.

 Call Manager IP Address/DNS is the IP/DNS of Cisco call manager

which establish the SCCP call signaling.

 Registration Failure Reason is the error message received from the

CallManager if the registration is failed.

Call Manager Performance (ms)

 Registration Time is the time required to transmit the registration

request to the CallManager and receive the registration acknowledgement from the Call Manager.

 Unregistration Time is the time required to transmit the unregistration

request to the CallManager and receive the unregistration acknowledgement from the CallManager.

 TCP Connect Time is the time required to establish the connection.

 TCP Disconnect Time is the time required to close the connection.

 Off Hook Time is the time in milliseconds between the transmission of

the Off Hook Message to the CallManager and the reception of the Call State (off-hook) message.

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Test Results

Call Manager Statistics (SCCP)

 Post Dial Delay is the time in milliseconds between the transmission

of the last dialed digit to the CallManager and the reception of the Alerting tone message.

 Post Pickup Delay is the time in milliseconds between the reception of

the SCCP Open Receive Channel “Ack” message and the reception of the first media RTP packet.

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Test Results

Call Quality Details

From the Test s menu, tap Results and the Call Quality Details tab.

MOS / R-Factor

 EXFO Conversational MOS is the Voice Quality Conversational Mean

 User R-Factor is the Voice Quality R-factor for the incoming RTP

RTP Statistics

 Packet Count is the number of good packets received from endpoint.

 Lost Packets is the Number of lost packets in the incoming RTP

 Late Packets is the number of RTP packets that arrived with a higher

Opinion Score (MOS) score for the incoming RTP stream.

stream.

jitter than the acceptable jitter input parameter value. The test discards all late packets.

 Early Packets is the number of packets received and dropped by the

local endpoint that were too early and could not be buffered within the jitter buffer.

 Total Lost Packets is the total number of RTP packets that were early,

lost and late. This value is calculated using the jitter buffer length input parameter value.

 Total Lost Packets (%) is the percentage of total lost packets from the

number of correct packets and total lost packets. i.e total lost packets/(packet count + total lost packets)* 100.

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Test Results

Call Quality Details

 Out-of-Order Packets is the number of RTP packets that were

received out of order. This value is calculated using the jitter buffer length input parameter value.

 Duplicate Packets is the number of duplicate packets received and

dropped by the local endpoint.

 No. of Loss Periods is the count for the number of periods of lost RTP

packets. One period of lost packets equals one or more consecutive packets that were lost during the transmission of the packet stream.

 Loss Period Size (Min) is the size of the smallest period of lost RTP

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result would return 1.

 Loss Period Size (Max) is the size of the largest period of lost RTP

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result would return 5.

 Loss Period Size (Avg) is the average size of periods of lost RTP

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result would return 3.

 Loss Type is the type of packet loss (random or burst) that occurred in

the media stream to the Callee. “None” is displayed, when there are no lost packets.

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Test Results

Call Quality Details

Degradation Factors

 Codec (%) is the percentage of MOS score degradation that was due to

 Latency (%) is the percentage of MOS score degradation that was due

 Packet Loss (%) is the percentage of MOS score degradation that was

Jitter (ms)

Length of time in milliseconds of the minimum, maximum, and average jitter of the incoming RTP packets.

the codec type in the incoming RTP stream.

to the delay in the incoming and outgoing RTP stream.

due to the packet loss in the incoming RTP stream.

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Test Results

Controller Statistics

Displays the result information for packets that travelled from the responder to the controller.

Note: The Controller Statistics tab is only available when Responder is selected

under the Call Mode from the Configuration tab.

From the Test s menu, tap Results and the Controller Statistics tab.

MOS / R-factor

 EXFO Conversational MOS is the Voice Quality Conversational Mean

Opinion Score (MOS) score for the RTP stream from the Responder to the Controller.

 User R-Factor is the Voice Quality R-factor for the RTP stream from the

Responder to the Controller.

RTP Statistics

 Packet Count is the number of good packets received at Controller.

 Lost Packets is the number of lost RTP packets going from the

Responder to the Controller.

 Late Packets is the number of RTP packets that arrived with a higher

jitter than the acceptable jitter input parameter value. The test discards all late packets.

 Early Packets is the number of packets received and dropped by the

Controller that were too early and could not be buffered within the jitter buffer.

 Total Lost Packets is the total number of RTP packets that were either

lost, early, or late. This value is calculated using the jitter buffer length input parameter value.

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Test Results

Controller Statistics

 Total Lost Packets (%) is the percentage of RTP packets that were

 Out-of-Order Packets is the number of RTP packets that were

 Duplicate Packets is the number of duplicate test packets received

 No. of Loss Periods is the count for the number of periods of lost RTP

 Loss Period Size (Min) is the size of the smallest period of lost RTP

 Loss Period Size (Max) is the size of the largest period of lost RTP

either lost or late.

received out of order. This value is calculated using the jitter buffer length input parameter value.

and dropped by the Controller.

packets. One period of lost packets equals one or more consecutive packets that were lost during the transmission of the packet stream.

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result will return 1.

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result will return 5.

 Loss Period Size (Avg) is the average size of periods of lost RTP

packets. For example, if three periods of loss packets were detected, the first with 3 packets, the second with 5 packets, and the third with 1 packet, this result will return 3.

 Loss Type is the type of packet loss (random or burst) that occurred in

the media stream to the Controller. “None” is displayed, when there are no lost packets.

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Test Results

Controller Statistics

Degradation Factors

 Codec (%) is the percentage of MOS score degradation that was due to

the codec type in the RTP stream from the Responder to the Controller.

 Latency (%) is the percentage of MOS score degradation that was due

to the delay in the RTP stream from the Responder to the Controller.

 Packet Loss (%) is the percentage of MOS score degradation that was

due to the packet loss in the RTP stream from the Responder to the Controller.

Jitter (ms)

Length of time in milliseconds of the minimum, maximum, and average jitter of the RTP packets traveling from the responder to the controller.

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Test Results

Responder Statistics

Displays the result information for packets that travelled from the controller to the responder.

From the Test s menu, tap Results and the Responder Statistics tab.

MOS / R-factor

 EXFO Conversion MOS is the Voice Quality Conversational Mean

 User R-Factor is the Voice Quality R-factor for the RTP stream from the

RTP Statistics

 Packet Count is the number of good packets received at Responder.

 Lost Packets is the number of lost RTP packets going from the

Opinion Score (MOS) score for the RTP stream from the Controller to Responder.

Controller to Responder.

 Late Packets is the number of RTP packets that arrived with a higher

jitter than the acceptable Jitter input parameter value. The test discards all late packets.

 Early Packets is the number of packets received and dropped by the

Responder that were too early and could not be buffered within the jitter buffer.

 Total Lost Packets is the total number of RTP packets that were either

lost, early, or late. This value is calculated using the jitter buffer length input parameter value.

 Total Lost Packets (%) is the percentage of RTP packets that were

either lost or late.

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Test Results

Responder Statistics

 Out-of-Order Packets is the number of RTP packets that were

received out of order. This value is calculated using the jitter buffer length input parameter value.

 Duplicate Packets is the number of duplicate test packets received

and dropped by the Responder.