Transcript
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 3
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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 2 nd 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
Loading complimentary applications
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
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/enus/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.
IMS-SIP client
RTP
Encoding
Audio in
Ethernet RTP
Decoding
Audio out
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
[email protected]), 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 IMS-SIP server & client
* 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.
.bat file, IPv6 router adv*
Windows 7 PC VoLTE UE Ethernet
Agilent PXT
RF
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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 IMS-SIP server & client(s)
Optional WAN connection for web browsing
Windows XP or 7 PC IPv6 Router
Ethernet
VoLTE UE
Agilent PXT
RF
VoLTE UE
Delay/Jitter tool, e.g. ZTI NetDisturb
Optional Delay/Jitter added to any link
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Optional 2nd Agilent PXT
RF
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)
Ethernet
Windows XP or 7 PC
Optional WAN connection for web browsing WiFi AP
WiFi
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 IMS-SIP server & client
* 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.
.bat file, IPv6 router adv*
B&K ZE 0948
Windows 7 PC USB
VoLTE UE Ethernet Audio Line in/out
Agilent PXT
RF
HATS
PESQ Audio Analyzer
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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
IMS-SIP client
RTP
Encoding
Audio in Sound card Line In/Out
Ethernet RTP
Decoding
Audio out
Normal call path
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 10 x Log(P x 1000) = 0 dBm
(0.775 Volts)2 / (600 ohms) = 0.001 Watts 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.
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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.
0dBm 600 ohm matched to 3.14dB down from codec full load
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Wave file
Vrms
Vpeak
70%
~0.775V
~1.095V
100%
1.112V
1.572V
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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
B&K ZE 0948
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 IMS-SIP server & client
Agilent IMS-SIP server & client USB
VoLTE UE
VoLTE UE Ethernet
Ethernet Audio Line in/out
Agilent PXT
Agilent PXT
RF
PESQ Audio Analyzer
PESQ Audio Analyzer
Headphone/mic jack
Headphone/mic jack
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RF
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 21
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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://wwwmmsp.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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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 NetDisturb
Agilent IMS-SIP server & client(s)
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
USB
Windows XP or 7 PC
Ethernet
VoLTE UE Delay/Jitter insertion
Agilent PXT
RF
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
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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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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
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