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Agilent Confidential Rev 1.0 2012-03-05

Testing Voice over LTE (VoLTE) Phones

with the Agilent E6966A IMS-SIP Network

Emulator Application white paper

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Contents

Introduction This Application White Paper offers guidance for LTE mobile phone developers, test engineers and

test lab personnel using the Agilent E6966A IMS-SIP Network Emulator Software. While many

concepts described in the document will apply to all VoLTE devices, examples have been written

around settings needed for LTE Band 13 VoLTE 2012 devices.

The document is structured as follows:

I. Verifying Agilent E6966A installation

a. Software, Licenses, Example and Configuration Files

b. Verifying configuration with IPv4 Calls from Agilent client to Agilent client

c. Verifying configuration with IPv6 Calls from Agilent client to Agilent client

d. Loading complimentary applications

e. Enabling IPv6 on Windows XP PCs

f. Audio Paths

g. Windows volume settings

h. Troubleshooting

II. Making voice calls with VoLTE phones

a. IPv6 voice call with a VoLTE enabled UE

b. Call Waiting

c. Making end to end calls with two VoLTE phones

d. WiFi Connection

III. Voice quality testing with HATS

a. Connecting to external audio analyzers and HATS systems

b. Audio interface reference levels

c. Calibration procedure

d. Delay budgets

e. Other Audio Test Configurations

IV.3rd party software and hardware

a. Router Configuration

b. Recommended PC Line in/out interface

c. SIM selection

d. WireShark

e. Obtaining and Preparing Audio Files

f. Adding delay and Jitter

g. Kapanga Softphone

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I. Verifying Agilent E6966A installation

Software, Licenses, Example

and Configuration Files

Links for software downloads, installation instructions and license redemption are available by

navigating from;

www.agilent.com/find/E6966A_software and directly from

Agilent E6966A IMS-SIP Network Emulator Software Downloads site.

Updated versions of this document, together with configuration and example files referenced are

available from a password protected portion of the site linked above, or can be downloaded

directly from the link below;

VoLTE Sample and Configuration Files for Agilent E6966A (zip)

Testing Voice over LTE (VoLTE) Phones with the Agilent E6966A IMS-SIP Network

Emulator (pdf)

Verifying configuration with

IPv4 Calls from Agilent client to

Agilent client

Connect PC Ethernet to a local router, PXT or another PC to activate Ethernet adaptor

Set IP IPv4 address on PC to 192.168.1.230

For Windows 7; from start menu, enter “View Network Connections” into the Search

programs and files entry box and press enter

Or Alternatively, for Windows XP; Open the control panel and select “Network Connections”.

Or, if available, Start>Connect to>Show all network connections

Double click on the Local Area Connection icon, select properties, then Internet Protocol

Version 4 (TCP/IPv4), then properties

Select “Use the following IP address” and enter 192.168.1.230, subnet 255.255.255.0

Select “Use the following DNS server address” and leave entry fields blank

Select OK, Close, Close, X, to close all IP network adaptor setting windows

Open server application Start>All Programs>Agilent E6966A>IMS-SIP server>IMS-SIP

server emulator

Review Error log (tab at bottom of application) – should be no errors – refer to

troubleshooting if errors appear

Verify WCF port 8240 and Server port 5060 are shown in bottom of screen – if not refer to

troubleshooting

If “-----has not been activated” appears on the top frame of the server window refer to

troubleshooting

If necessary, close and re-open the application until it runs without errors appearing in the

Error Log

On the Register tab set Realm/Domain to test.3gpp.com

On the Configuration tab set the Link-Local Network Interface to the Ethernet connection that

has been activated with the address 192.168.1.230

In the lower pane select the green arrow button to enable logging

Connect headphones and microphone to PC before opening client application

Open the client application Start>All Programs>Agilent E6966A>IMS-SIP client>IMS-SIP

client emulator

Review Error log (tab at bottom of application) – should be no errors – refer to

troubleshooting if errors appear

Select Register button (upper Left)

Verify that upper right status changes from Not Registered to Registered, if registration fails,

refer to troubleshooting section

On the server application, the Agilent client should now appear as the first entry in the list of

Registered Users

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On the client, select Configuration tab, then File locations, press the button next to “Test file

name location” under Audio Files. Select the file “ImsSipClientTest”, the path to this file

should have been set during the installation process but may vary depending on the PC

configuration. The file can usually be located at C:\Documents and Settings\All

Users\Documents\Agilent\IMS-SIP Client\ImsSipClientTest.wav or similar

On the client select Configuration tab then Audio

Select the Speaker device to correspond to headphones from the Speaker drop down menu

Similarly, select the Microphone device from the drop down menu

Press Play Local to play the ImsSipClientTest wav file to verify headphone connection. The

contents of the ImsSipClientTest.wav should be heard. If not see troubleshooting section

Adjust volume controls – see also notes on Windows Volume Settings

Press Stop Local then press Start Loop Test. Speak into microphone and adjust volume

and/or change Microphone selection to hear voice echoed back in headphones. Press Stop

Loop Test when done.

Select client application Start>All Programs>Agilent E6966A>IMS-SIP client>IMS-SIP client

emulator to open a second instance of the IMS-SIP client

On the 2nd client select the SIP connection tab, change the contact IP port to a different value

from the 1st client. E.g. if the first client has 5080 for Contact IP port, enter 6000 for the 2nd

client Contact IP port.

On the 2nd client verify that the User name is different from the 1st client, and change if

necessary so that 1st and 2nd client have different user names.

Enter other settings to be the same on the 2nd client as the 1st client instance

Select Register on the 2nd client, verify that its state (upper right) changes to Registered and

that it appears as the second entry in the Registered Users list on the server application

On the 1st client VoIP tab Call location, enter the User Name of the 2nd client and press Dial

The 2nd client should show Incoming Call, select Answer on the 2nd client

Call should be connected between two clients

Audio results in this configuration may vary from Windows XP to Windows 7 and on PC audio

configurations. In most setups, it should be possible to select Enable received audio loopback

on the VoIP tab of the 2nd client and hear voice echoed back.

Audio analyzer and logging should be active on both clients, logging should be active on

server to become familiar with these and other client and server settings and features

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Verifying configuration with

IPv6 Calls from Agilent client to

Agilent client

IPv6 is enabled by default on Windows 7 PCs, for Windows XP, see Enabling IPv6 on

Windows XP PCs

Repeat procedure above, but with the following changes

In Windows Double click on the Local Area Connection icon, select properties, then Internet

Protocol Version 6 (TCP/IPv6), then properties, Use the following IPv6 address and enter

2001::210:1ff:fe23:4567, Subnet prefix length 64. All other settings can be blank

Open the server as above

Open the 1st client and for both SIP server IP address and Contact IP address enter

2001::210:1ff:fe23:4567

Open the 2nd client and for both SIP server IP address and Contact IP address enter

2001::210:1ff:fe23:4567

Both clients should be able to register and make calls as above

Loading complimentary

applications

Refer to the section on 3rd party applications for more information. It is recommended that

WireShark is downloaded an installed as a companion to the Agilent IMS-SIP Server and Client.

WireShark is the industry standard IP and SIP protocol logging and analysis tool. It is freely

available (GPL licensed) for download, installation and use. The Agilent IMS-SIP server and Client

include SIP logging. WireShark complements this capability by providing more detailed and

independent SIP logs, and adds the capability to decode aspects of RTP packets helpful for

debugging interoperability and setting issues.

Sample Speech files can be downloaded from a number of internet sources linked in later sections.

These include reference recording of Harvard Sentences available from the ITU and other sources.

Sample rates can be adjusted to 8 or 16kHz to suit the Agilent IMS-SIP client using a number of

tools including Audacity.

Audacity can be used to edit wav files and create test signals, for example sine-wave tones and

noise. Audacity is freely available (GPL licensed) for download, installation and use.

Many delay and jitter emulators can be used successfully with the Agilent IMS-SIP server and

client. ZTI NetDisturb is a low cost tool that can be co-resident with the Agilent applications and

seems to work well. It can be used to add delay and jitter to any Ethernet link in the system.

In addition to the Agilent IMS-SIP client, it is possible to use the Agilent IMS-SIP server with a

variety of other, non-Agilent, SIP clients. The Kapanga client from Ecotronics Ventures LLC is a low

cost softphone client that has the necessary features to connect to Voice over LTE phones. Its

feature set complements the Agilent IMS-SIP client. Kapanga is based on a different SIP stack,

media stack and codec to the Agilent IMS-SIP client so can be helpful as a tool for interoperability

testing. A configuration file for Kapanga can be downloaded in the zip file linked above. With

these settings, Kapanga operates effectively with the Agilent IMS-SIP Server and Client.

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Enabling IPv6 on Windows XP

PCs

Microsoft guidance is available from;

http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en-

us/sag_ip_v6_pro_inst.mspx?mfr=true

Windows XP settings are summarized below;

Open Network Connections

Right-click any local area connection, and then click Properties.

Click Install.

In the Select Network Component Type dialog box, click Protocol, and then click Add.

In the Select Network Protocol dialog box, click Microsoft TCP/IP version 6, and then click

OK.

Click Close to save changes to your network connection.

Audio Paths The Agilent IMS-SIP client includes a variety of audio paths to enable simple voice echo testing,

playing and recording wav files. The application also includes facilities to play wav files and

loopback signals to enable gain calibration.

Encoding

Decoding

IMS-SIP client

RTP

RTP

Audio in

Audio out

Ethernet

Normal call path

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Windows volume settings When adjusting volume settings to achieve calibrated Line in/out audio levels, some care is needed

to achieve expected results.

When operating with Windows XP, the Agilent IMS-SIP client speaker volume slider controls the

Windows XP master speaker volume for the selected device. Windows sets volume from master

volume x wave volume. When setting calibrated audio levels, set Windows XP wave volume slider

to maximum, and then make calibration volume adjustments using the slider in the Agilent IMS-SIP

client.

When operating with Windows 7, the Agilent IMS-SIP client speaker volume slider controls the

Windows 7 application volume for the selected device. Windows sets volume from master volume

x application volume. When setting calibrated audio levels, set Windows 7 master volume slider to

maximum, and then make calibration volume adjustments using the slider in the Agilent IMS-SIP

client.

If two or more instances of the Agilent IMS-SIP client are running and more than one client

instance is configured with the same Speaker and Microphone device, both instances may affect

the same Windows XP volume controls. When making calibrated audio measurements, take care

to have just one client instance running, or select different audio devices for use with each client

instance.

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Troubleshooting Agilent IMS-SIP client doesn‟t register with server, or not able to connect a voice call

Verify that Contact IP port for each instance of IMS-SIP Client running in the PC is different

and that there are no other applications using the selected ports. For example if 1st instance

is set to have a Contact IP port of 5080, select 6000 for 2nd instance

Verify that IMS-SIP server Configuration>Link Local Network Interface is set to an active

Ethernet adaptor with a relevant IP address, for example 192.168.1.230

When making calls without a fully qualifies URI (e.g. dialing +12222222222 rather than

+12222222222@test.3gpp.com), verify that the realm/domain is set in the IMS-SIP server to

match the setting in each IMS-SIP client.

When using Agilent IMS-SIP client to make calls to an Agilent IMS-SIP client in another PC, audio

flows in just one direction.

Check and confirm the correct setting of the contact IP address in each client. The contact IP

address should be set to match the IP address of the PC hosting the client.

Server error messages

“Start Transport: Error Opening Server Socket” ;Port clash with other SIP applications. Close

all applications and open server before running other SIP applications, check other

applications to verify that port 5060 is available for server

WCF port not showing 8240

Application needs to be run on PC with administrator privileges

Potential port clash with other applications. Re-install server on a clean PC

Server port not showing 5060

Port clash with other SIP applications. Close all applications and open server before running

other SIP applications, check other applications to verify that port 5060 is available for server

Resetting client settings to default values

When the client opens it creates a file “ImsSipClientSettings” located at

C:\Users\Public\Documents\Agilent\IMS-SIP Client\ or similar path depending on your PC

configuration.

Close the client application. Delete the “ImsSipClientSettings” file. Re-open the client

application. The client will now open with default settings and re-create a new version of

the “ImsSipClientSettings” file.

IMS-SIP server or client opens with message at top of window “Has not been activated”

Licenses have not been properly installed. Open Agilent License Manager Application (Blue

key icon in Windows System Tray) and verify that E6966A 1FP has been installed to activate

both server and client, or E6966A 2FP has been installed to activate only the client. Review

installation instructions linked above.

No sound from headphones or microphone not working;

Agilent IMS-SIP client application needs to be opened after new audio devices (e.g.

headphones) have been added to PC. The application queries Windows audio devices when

it opens. Close the IMS-SIP client and re-open to enable the application to query and

become synchronized with any changes to Windows audio device availability

Try using headphone and microphone with another Windows audio application for example

Sound Recorder. Windows may not be configured properly to operate with headphones;

drivers may not have been loaded or activated.

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II. Making voice calls with VoLTE phones

IPv6 voice call with a VoLTE

enabled UE

Connecting the system;

Windows 7 PCs can be connected directly to the Agilent E6621A PXT using Ethernet cable

Alternatively, Windows 7 or Windows XP PCs with IPv6 enabled can be connected to the

PXT via the LAN side of an IPv6 router. In this configuration the Cisco RVS4000 IPv6 router is

recommended but other IPv6 routers are likely to be suitable. See router configuration.

Settings files referenced in this section can be downloaded in the zip file linked above. These

include;

Batch (.bat) file to configure Windows 7 for router advertisements

Scenario files for Agilent E6621A PXT LTE Test Set

Recall register files for Agilent E6621A PXT LTE Test Set

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client

Windows 7 PC

Ethernet

RF

.bat file, IPv6

router adv*

* Batch file can be used to configure Windows 7 PC

to transmit IPv6 Router Advertisements. Alternatively

an IPv6 router can be added to the connection

between a Windows 7 or XP PC and the PXT.

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Router Advertisements

In setups including a router, the advertisement messages needed by the UE under test will be

generated by the router

In direct PC to PXT connection setups only, no router will be present to generate the needed

router advertisement messages. In this configuration, run the IPv6_Win_7_Router_Adv.bat

batch file located in the zip file linked above to configure a Windows 7 PC to provide the

needed router advertisement messages. Note with some Windows 7 installations it will be

necessary to run the batch file as administrator. Right click and select Run as Administrator.

Other PC Ethernet settings should be made as described in IPv6 calling section above. Eg. PC

IP address 2001::210:1ff:fe23:4567

Configure the Agilent PXT

PXT firmware 6.3.1.0 or above is recommended.

Settings will vary depending on band and other UE characteristics. Typical configuration for

a band 13 UE with a Gemalto W004v2 SIM are described (see SIM selection)

Two example scenario files „VoLTE_Band_13_IPv6_5x_1.LBMF‟ and

„VoLTE_Band_13_IPv6_8x_2.LBMF‟, and a recall register file „Band_13_VoLTE.xml‟ are

provided in the zip file linked above . Copy these files using a USB memory stick to the PXT

Place the scenario files (LBMF files) in the PXT Windows system at D:\Program

Files\Agilent\E6621A\LTE-Scenario,

Place the register file (xml file) at D:\Program Files\Agilent\E6621A\State

Set the PXT address 192.168.1.60 (IP Config>Network Setup>IP setup)

Set the PXT Gateway to 192.168.1.230 for direct connection to PC or 192.168.1.1 for router

connection

On the PXT, select Recall>Recall State, then select Band_13_VoLTE.xml

EPC settings (use a mouse to access EPC settings from E-EPC icon on PXT Windows Taskbar

IPv4 range 192.168.1.51 to 192.168.1.59

IPv6 range 021f:29ff:fe7c:8f51 to 021f:29ff:fe7c:8f59

Non local prefix 2001::

Gateway 192.168.1.230 for direct connection to PC or 192.168.1.1 for router

connection

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Return to the PXT application

Load and run scenario on the PXT by selecting Mode Setup>Call Scenario, then Scroll to

VoLTE_Band_13_IPv6_5x_1.LBMF, Press Load

Run the scenario by selecting; Mode> Emulator Mode>Run

Verify connections

On the PC hosting the IMS-SIP server and client, select start>Run and type cmd then enter to

open up a Windows command shell

Type ping 192.169.1.60 and look for replies to confirm that that PXT and the server have IP

connectivity

Open applications

On the PC hosting the IMS-SIP server, open the server and then open IMS-SIP client

On the client in the SIP connection tab for username enter +12222222222 or similar

Verify that IP address matches IP addresses are set as below to match server PC addresses

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Select „Register‟ to register the IMS-SIP client with the IMS-SIP server

On Configuration tab set codecs as below for operation with VoLTE 2012 UE (or as required

by UE)

AMR and/or AMR-WB codec selected

For both AMR and AMR-WB, select Octet aligned with value 1, select Include

Mode Set and Select 12.2kbit/s for AMR and 12.65kbit/s for AMR-WB

Configure VoLTE UE

Load suitable SIM for example Gemalto W004v2 (see SIM selection)

Configure the UE to have SigComp disabled

The P-CSCF (IMS-SIP server address) is signalled to the UE from the PXT scenario file during

registration and link setup. Some UE will have a separate menu for P-CSCF which needs to

be also set to 2001::210:1ff:fe23:4567 to match the IPv6 address of the IMS-SIP server

Connect the UE to the PXT with a suitable RF cable

Power-on the UE, after some seconds, the UE should attach to the PXT. The PXT will show

CON

Shortly after, the UE should register with the IMS-SIP server and appear on the Registered

Users list under the entry for the Agilent IMS-SIP client.

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Make a call

Calls should be possible from the UE to the Agilent IMS-SIP client, for example by dialling

+12222222222 on the UE and selecting an IMS call, or by entering the UE phone number in

the Agilent IMS-SIP Call Location field and pressing Dial.

The UE phone number is derived from settings on the SIM and can be seen in the Address of

Record column of the IMS-SIP server Registered Users. An unmodified Gemalto W004v2

SIM will typically be configured for +11234567890

Troubleshooting

In the event of a failure to establish a SIP call connection, or a failure to get an intelligible

voice connection, review logs on the Agilent IMS-SIP server, IMS-SIP client and by running

Wireshark on the server PC to investigate and SIP messaging and RTP packet configurations.

Next steps

Call Waiting

For parametric audio quality measurements see Connecting to external audio analyzers and

HATS systems

To make end to end UE to UE calls see Making end to end calls between VoLTE UEs

To add impairments see Adding delay and Jitter

Call Waiting With two or more instances of the Agilent IMS-SIP client open, it‟s possible to place calls from

each of them to a VoLTE UE and test call waiting and call reject scenarios.

With settings as described in IPv6 voice call with a VoLTE enabled UE, open a second instance of

the Agilent IMS-SIP client.

On the 2nd client select the SIP connection tab, change the contact IP port to a different value

from the 1st client. E.g. if the first client has 5080 for Contact IP port, enter 6000 for the 2nd

client Contact IP port.

On the 2nd client change User name to be different from the 1st client. E.g. of the first client is

set to +12222222222, second client could be +13333333333

Enter other settings to be the same on the 2nd client as the 1st client instance

Select Register on the 2nd client; verify that its state (upper right) changes to Registered and

that it appears as an additional entry in the Registered Users list on the server application.

It should be possible to place calls from either client instance to the VoLTE UE and from the

VoLTE UE to each of the two client instances

To test call waiting, place a call from the first client instance to the VoLTE UE, once

answered, place an additional call from the second client instance to the VoLTE UE. The UE

should then be able to answer or reject the second call and/or flip between each call, placing

each of the clients on hold in turn.

Using a similar procedure to select unique port numbers and user names (phone number)

additional clients can be opened to verify that the VoLTE UE is able to reject additional calls.

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Making end to end calls with

two VoLTE phones

Using two PXTs and the Agilent IMS-SIP server connected to an IPv6 router, end to end calls can

be made between VoLTE UEs.

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client(s)

Windows XP or 7 PC

Ethernet

RF

Optional 2nd

Agilent PXT

VoLTE UE

RF

IPv6 Router

Optional WAN

connection for web browsing

Optional

Delay/Jitter added to any linkDelay/Jitter tool,

e.g. ZTI NetDisturb

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Configure the IMS-SIP server and one of the PXTs and one of the VoLTE UEs as described in IPv6

voice call with a VoLTE enabled UE

A router will be needed to connect the two PXTs with the IMS-SIP server PC. See also router

configuration. An Ethernet switch or hub will not enable this configuration, a router is required.

Configure the second PXT with the same settings as the first with the following differences;

PXT 1 address setup

IP address 192.168.1.60

Gateway 192.168.1.1

EPC

IPv4 range 192.168.1.51 to 192.168.1.59

IPv6 range 021f:29ff:fe7c:8f51 to 021f:29ff:fe7c:8f59

Non local prefix 2001::

Gateway 192.168.1.1

Load and run scenario file VoLTE_Band_13_IPv6_5x_1.LBMF

PXT 2 address setup

IP address 192.168.1.61

Gateway 192.168.1.1

EPC

IPv4 range 192.168.1.100 to 192.168.1.109

IPv6 range 021f:29ff:fe7c:8f81 to 021f:29ff:fe7c:8f89

Non local prefix 2001::

Gateway 192.168.1.1

Load and run scenario file VoLTE_Band_13_IPv6_8x_2.LBMF

VoLTE UE 1 and 2 should have the same settings but be loaded with different SIMs. Since the

phone number of the UE is derived from the SIM, one of the UEs will need to have a SIM which

has been modified to contain a different phone number.

With suitable RF cables, connect one UE to each PXT. Power-on the UEs, each will attach to a PXT

and register with the IMS server. Calls can be placed from UE to UE and from each UE to and from

Agilent IMS-SIP client(s) running on the server or other PCs connected to the LAN side of the

router.

With suitable VoLTE UEs, video calls can also be connected.

Next steps

To add impairments see Adding delay and Jitter

Call Waiting

For parametric audio quality measurements see Connecting to external audio analyzers and

HATS systems

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WiFi Connection

With an appropriate IPv4 or IPv6 WiFi router/access point connected to the Agilent IMS-SIP server

and a WiFi enabled phone, VoIP calls can be made.

Agilent IMS-SIP server & client(s)

Windows XP or 7 PCEthernet

WiFi

Optional WAN

connection for web browsing

WiFi AP

WiFi

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III. Voice quality testing with HATS

Connecting to external audio

analyzers and HATS systems

Using a PC with a sound card supporting line in and out levels, the Agilent IMS-SIP client can be

connected to an external audio analyzer to perform PESQ and other audio quality measurements.

Depending on the capabilities of the audio analyzer, these tests can include cabled audio

connections to the UE under test or acoustic coupling via HATS (Head and Torso Simulator).

The Brüel & Kjær (B&K) ZE 0948 USB Audio Interface is recommended as a suitable PC sound card

to interface the Agilent IMS-SIP client to external audio analyzers.

The B&K ZE 0948 Audio Interface is connected by USB to the PC hosting the Agilent IMS-SIP

client. Drivers self install. When the Agilent IMS-SIP client application is opened the Speakers

(USB Audio CODEC) and the Microphone (USB Audio CODEC) can be selected from the Speaker

and Microphone lists in the Audio section of the Configuration tab.

The B&K ZE 0948 is currently only supported by the Agilent IMS-SIP Client with Windows 7

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client

Windows 7 PCB&K ZE 0948

Ethernet

USB

RF

Audio Line in/out

HATS

PESQ Audio Analyzer

.bat file, IPv6

router adv*

* Batch file can be used to configure Windows 7 PC

to transmit IPv6 Router Advertisements. Alternatively

an IPv6 router can be added to the connection

between a Windows 7 or XP PC and the PXT.

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Audio interface reference

levels

The Agilent IMS-SIP client includes features to align analogue line in and out voltage levels with

required levels of codec full scale.

Calibration wav files are available in the zip file linked above with signal amplitudes

referenced to codec full scale to calibrate line out voltages

A loop back function can be enabled to calibrate line in voltages

An example calibration level can be performed based on industry conventions defined in 3GPP

26.132 section 5.2.1.

The 0dBr reference point is defined as;

Analogue domain; 0dBm sine wave in a 600 ohm system (interpreted as 775mV rms, or

1.09V peak)

Digital domain; 3.14dB below codec maximum full load (interpreted as 0.696 or 69.6% full

scale PCM wav file)

Analogue domain

V2 / R = P (0.775 Volts)2 / (600 ohms) = 0.001 Watts

10 x Log(P x 1000) = 0 dBm Log (0.001 Watts x 1000) = 0 dBm

Digital domain

dB relative to codec full scale dBFS = 20 x Log(0.696) = -3.14 dB or 3.14 dB below codec

clipping point

Input and output gain levels can be adjusted to equate the level of a 70% wav file (approximately

0.696) to an analogue signal level of 775mV rms.

Encoding

Decoding

IMS-SIP client

RTP

RTP

Audio in

Audio out

Ethernet

Sound card Line In/Out

Normal call path

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Calibration procedure Example levels referenced to calibration with B&K ZE 0948 USB Audio Interface, similar procedure

can be followed with other audio interfaces.

Install B&K ZE 0948 USB Audio Interface with a Windows 7 PC

Open the IMS-SIP client and register the client with IMS-SIP server

On the Configuration tab, select Audio, then for Speakers and Microphone select; Speakers

(USB Audio CODEC) and the Microphone (USB Audio CODEC)

In Windows 7, verify that master volume is set to maximum. see notes on Windows volume

settings

In Agilent IMS-SIP client press Reset Levels (default settings are optimized to be close for ZE

0948)

Select File Locations from the Configuration tab. Under Audio Files, Test File Name location,

navigate to „70 percent -3.1dB 1kHz Sine 16k.wav‟. The file can be copied to the PC from

the zip file linked above.

Select Configuration, Audio, then press the Play Local button

Measure the left channel line out voltage from B&K ZE 0948 USB Audio Interface using

audio analyzer or voltmeter

Verify that line out voltage is 775mV rms or 1.09V peak. Adjust Speaker Volume setting on

the Agilent IMS-SIP client Configuration tab if needed. The volume slider can be moved with

PC mouse, fine adjustments can be made with mouse clicks on either side of the slider

control.

Select to Stop Local once adjustment is complete

Connect a 1kHz sine wave from an external signal generator set to 775mV rms or 1.09V peak

into the B&K ZE 0948 USB Audio Interface Line in

Select Start Loop Test

Verify that the Line out voltage is 775mV rms or 1.09V peak. Adjust Microphone Volume

slider on IMS-SIP client if necessary to optimize level.

Select Stop Loop Test once adjustment is complete.

Wave files of other than 70% and reference voltages of other than 775mV rms can be used to

achieve a variety of alignment levels between the digital and analogue domains. While 775mV

mapping to 70% is a common signal level, 70% wav file mapping to 1Vrms (approximately 2.2dB

lower codec drive level) is also a frequently used operating point.

3.14dB corresponds to a 69.66% wav file. Lower level uncertainties will be possible using 100%

wav file and calibrating to alternate voltage levels.

Volume setting resolution limitations within the Agilent IMS-SIP client and Windows mean that to

achieve the best possible calibration it is necessary to make addition level offset fine adjustments in

the external audio generator and analyzer. These additional fine adjustments can be used to

compensate for Windows level setting resolution limitations.

Wave file Vrms Vpeak

70% ~0.775V ~1.095V

100% 1.112V 1.572V

0dBm 600 ohm matched

to 3.14dB down from

codec full load

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Agilent Confidential

Delay budgets Nominal delays have been measured as follows;

Analogue audio line in to B&K ZE 0948 to Agilent IMS-SIP voice codec; 24ms

AMR or AMR-WB voice codec encoding delay; 32ms

Ethernet RTP transmission delay from PC to PC via IPv6 router; 1ms

Ethernet to LTE RF transmission delay via IPv6 router and Agilent PXT; 11 ms

AMR or AMR-WB voice codec decoding delay; 32ms

Agilent IMS-SIP voice codec to analogue line out via B&K ZE 0948; 24ms

Delays have been measured with Agilent E6621A PXT revision 6.3.1.0 and Cisco RVS4000 router.

Delays were confirmed to be consistent with two Windows 7 (32 bit) PCs; Core i5 2.53GHz and

Core 2 Due P8600 2.4GHz. Delays may vary slightly with PCs with significantly higher or lower

performance.

Delay variation for an end to end call between two Agilent IMS-SIP clients varied nominally by less

than +/-10 ms. Larger delay variations can be seen during initial call setup and when PCs are

being used to run other CPU intensive applications. Jitter buffer effects will also increase delay if

IP path has high levels of jitter.

Based on the nominal results above, measurements confirm the following system level delays;

End to end phone to phone calls. LTE RF interface of PXT1, RTP stream via IPv6 router to

PXT2 RF interface; 11+11=22ms

Agilent IMS-SIP client calling Agilent IMS-SIP client in another PC, connected via IPv6 router

with B&K ZE 0948 connected to each client. Delay from audio line in of first client to audio

line out of second client; 24+32+1+32+24=113ms

Agilent IMS-SIP client calling Agilent IMS-SIP client on another PC, connected via IPv6

router. First client has B&K ZE 0948 line in and out, second client configured for voice echo.

Delay from line in to line out on first client; 24+32+1+32+32+1+24=178ms

Based on these measurements, it is possible to compute system delays as follows;

VoLTE phone connected to PXT; delay from RF interface to B&K ZE 0948 line out via Agilent

IMS-SIP client in PC connected to PXT via IPv6 router; 11+32+24=67ms

B&K ZE 0948 line in via Agilent IMS-SIP client in PC connected to PXT via IPv6 router to RF

interface connected to VoLTE phone; 24+32+11=67ms

VoLTE phone RF interface echo path via PXT, IPv6 router, PC with Agilent IMS-SIP client set

to voice echo; 11+32+32+11=86ms

Other Audio Test

Configurations

The Agilent IMS-SIP Client and Server can be used with Agilent E6621A PXT and/or WiFi

connections to phones with a variety of other audio test configurations. Examples are shown

below

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & clientB&K ZE 0948

Ethernet

USB

RF

Audio Line in/out

PESQ Audio Analyzer

Headphone/mic jack

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client

Ethernet

RF

PESQ Audio Analyzer

Headphone/mic jack

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Agilent Confidential Rev 1.0 2012-03-05

IV. 3rd party software and hardware

The 3rd party hardware and software products listed in this section have been found to work

effectively with the Agilent E6966A IMS-SIP Server and Client, and the Agilent E6621A PXT LTE

Test Set. 3rd party products are listed as examples and it is likely that other similar products will

also work effectively.

Agilent is not able to endorse the stability or performance of 3rd party products, other than to cite

that in initial interoperability testing, they have functioned as expected.

Support for the products will be provided by the supplying company or organization and not by

Agilent. Some of the products listed are Open Source GPL licensed and free. Other products can

be purchased from the companies listed.

Agilent does not have financial relationships with the companies listed and does not benefit from

users selecting the example products listed or other equivalent items.

Router Configuration Suggested Router configuration:

Login from web browser on PC (192.168.1.1) name admin, password admin

IP address 192.168.1.1

Disable DHCP v4

Enable dual stack IPv4/IPv6

IPv6 address 2001::1 (length 64)

Enable router advertisements

Disable DHCP v6

Cisco RVS4000 or equivalent IPv6 router has been used effectively

http://www.cisco.com/en/US/products/ps9928/index.html

Recommended PC Line in/out

interface

For audio performance measurements, the B&K ZE 0948 or an equivalent sound card is

recommended.

B&K ZE 0948 is currently supported only with Windows 7 installations

Brüel & Kjær Sound & Vibration Measurement A/S

http://www.bksv.com/products/roomacousticssoftware/7841dirac/ze0948.aspx

SIM selection

The Agilent PXT and IMS-SIP client and server can be configured to operate with VoLTE UEs with a

variety of SIMs loaded.

Band 13 VoLTE 2012 devices can be operated conveniently using Gemalto W004v2 SIMs. These

are available directly from Gemalto to users with a Non-Disclosure Agreement (NDA) in place with

Verizon Wireless.

Contact Agilent via the support links on the pages linked above to obtain Gemalto e-mail ordering

information

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Agilent Confidential

WireShark

WireShark is the industry standard protocol logging and analysis tool. Software is freely available

to download and install from http://www.wireshark.org/.

WireShark can be co-resident on Windows XP or Windows 7 PCs running the Agilent IMS-SIP

client and/or server. Key applications of WireShark include:

Reviewing more detailed SIP/SDP message contents

Analyzing RTP payload formatting and contents

Logging SIP call flows

Saving logs for offline analysis and de-bug

Traffic of interest can be selected by entering filters for example „RTP‟, „SIP‟, „RTP or SIP‟

Obtaining and Preparing Audio

Files

Sample speech files can be downloaded from the links below and many other online sources

ITU P.501 http://www.itu.int/net/itu-t/sigdb/genaudio/Pseries.htm

McGill University Telecommunications & Signal Processing Laboratory http://www-

mmsp.ece.mcgill.ca/documents/Data/index.html

Audacity is a free software tool than can be used to change sampling rates (to required 8 and

16kHz), normalize volume levels assemble test sequences. Audacity can also synthesize tones and

noise signals.

http://audacity.sourceforge.net/

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Agilent Confidential Rev 1.0 2012-03-05

Adding delay and Jitter Delay and jitter can be added into any of the cabled Ethernet connections in the system. This can

be used for subjective assessments of audio and video quality for VoLTE UE to UE calling and can

be used for subjective of parametric assessments of audio quality for calls between a VoLTE UE

and the Agilent IMS-SIP client.

ZTI‟s NetDisturb software application has been tested to work with the Agilent IMS-Server and

client. In principle, other delay and jitter tools could also be used with the system.

NetDisturb can be purchased on line from http://www.zti-telecom.com/EN/NetDisturb.html, at the

time of writing, limited time trial licenses are also available

NetDisturb can be loaded on the same Windows 7 PC as the Agilent IMS-SIP server, or can be run

on a separate PC connected into the LAN. The software requires two additional Ethernet ports to

be made available on the server (or another) PC. Additional Ethernet ports can be added

conveniently with USB to Ethernet adaptors, for example USB 2.0 Gigabit Ethernet Adaptors from

www.plugable.com

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client(s)

Windows XP or 7 PCEthernet

RF

ZTI NetDisturb

USB

USB-Ethernet adaptors can be used with ZTI

NetDisturb to add delay/Jitter to any Ethernet link in

the system. NetDisturb can run on a separate PC or

share a PC with Agilent IMS-SIP client and/or server

Delay/Jitter insertion

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Agilent Confidential

NetDisturb makes full use of two Ethernet ports, providing a bridged connection with

programmable impairments. The bridged connection appears transparent to other PCs and

instruments on the LAN. When in use by NetDisturb, two Ethernet ports are set to be invsible to

Windows and other applications running on the PC.

NetDisturb includes a feature set to apply impairments;

In one direction or both directions

On specific protocol and packet types or all packets

Statistical delays and jitter or constant delays and jitter

Information reproduced with permission of ZTI. NetDisturb is ZTI Copyright

Kapanga Softphone

The Kapanga softphone client http://www.kapanga.net/IP/home.cfm, available from Ecotronics

Ventures LLC has a feature set which complements the Agilent IMS-SIP client.

Kapanga shares no common IP with the Agilent IMS-SIP client, making it useful for expanded

interop test coverage. From version 1.00.2187, Kapanga has had good interoperability with the

Agilent IMS-SIP client, IMS-SIP server and has a feature set which makes it suitable for testing

with VoLTE UEs.

Kapanga can be purchased from Ecotronics with licensing options to enable AMR and AMR-WB

codecs.

Once configured correctly, Kapanga can be co-resident in the same PC as the Agilent IMS-SIP

client and/or server to provide complementary features.

A configuration file Kapanga-Agilent-VoLTE-config.cfg is provided in the zip file linked above.

Using the Kapanga menu, select File>Open then navigate to Kapanga-Agilent-VoLTE-config.cfg.

Kapanga will then be configured to operate with the Agilent IMS-SIP client, IMS-SIP server and

have defaults set for voice calls with VoLTE UE.

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Agilent Confidential Rev 1.0 2012-03-05

Once the Kapanga-Agilent-VoLTE-config.cfg has been loaded into the Kapanga client, it will be

necessary to also setup audio input and output devices. From Kapanga menus select

Configuration>Audio/Video/Network Wizard.

Select microphone and speaker devices and adjust levels. It may be necessary to repeat this step

after first call has been made to get stable selection of audio input and output devices

Once settings have been made, select Configuration>Restart Softphone to have settings take

effect and to update registration. Settings listed below should be checked manually after the

configuration file has been loaded. Depending on PC settings, some may not be set automatically

and may need to be manually configured.

Recommended Kapanga settings are as follows;

General Settings>Configuration> Network Interface; IPv6 checked and Use Interface set to

match PC IPv6 address e.g. 2001::210:1ff:fe23:4567

Call Control Settings>Media Related> Offered Codecs AMR-WB, AMR

Call Control Settings>SIP Parameters Username and Display name both set to

+18888888888, Send Hold SDP reINVITEs selected, Enable 100 Reliability selected.

Call Control Settings>Proxy Configuration Domain/Realm test.3gpp.com, Outbound Proxy set

to match IP address of PC hosting IMS-SIP server (e.g. 2001::210:1ff:fe23:4567), Register

checked, Register Automatically checked, Proxy Keep Alive un-checked

Call Control>Transport & Security Local UDP SIP Port 7000 (to avoid port clash with Agilent

server and client)

Codec settings>AMR Enable RFC 4867 checked, Reorder bits checked, Enable DTX

unchecked, Bandwidth efficient mode unchecked, Robust sorting unchecked, 20ms selected,

12200bps selected

Codec settings>AMR-WB Enable RFC 4867 checked, Reorder bits checked, Enable DTX

unchecked, Bandwidth efficient mode unchecked, Robust sorting unchecked, 20ms selected,

12650bps selected